PJSIP one voip provider works and the other dosen’t

PatTheRat · March 5, 2025, 1:49pm

I am converting my small asterisk setup to PJSIP and have managed to create a working pjsip.conf and extensions.conf.

I have connected my phone to asterisk with PJSIP and am able to call out on one of my two voip providers. But for some reason the other provider does not work, even though the settings in PJSIP are identical.

Previously, when using sip.conf, the only difference in the two providers was one required CALLERID(num)=<my_number> which I have used in the current dialplan.

I ran ‘pjsip set logger on’ and got-

mymodem*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Received SIP request (1079 bytes) from UDP:111.111.11.1:5065 --->
INVITE sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=b405d6c8-74aa87d-13c9-83545D-571a4f6e-5fe9e490-46d60c
To: "815123456"<sip:815123456@192.168.0.1:5060>
Call-ID: b40916b0-74aa87d-13c9-83545D-76787d14-43ef54ca-46d60c
CSeq: 1 INVITE
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1222-46d60c-336097d9-b4023098
Max-Forwards: 70
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Type: application/sdp
Content-Length: 400

v=0
o=123456 1741093624 1741093624 IN IP4 111.111.11.1
s=-
c=IN IP4 111.111.11.1
t=0 0
m=audio 41872 RTP/AVP 107 8 0 18 96 108
a=rtpmap:107 AMR-WB/16000
a=fmtp:107 channels=1;mode-change-capability=2;max-red=220
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtpmap:108 telephone-event/16000
a=fmtp:108 0-15
a=maxptime:240
a=silenceSupp:off - - - -
a=ptime:20

<--- Transmitting SIP response (604 bytes) to UDP:111.111.11.1:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.11.1:5065;rport=5065;received=111.111.11.1;branch=z9hG4bK-1222-46d60c-336097d9-b4023098
Call-ID: b40916b0-74aa87d-13c9-83545D-76787d14-43ef54ca-46d60c
From: "home_KL" <sip:home_KL@192.168.0.1>;tag=b405d6c8-74aa87d-13c9-83545D-571a4f6e-5fe9e490-46d60c
To: "815123456" <sip:815123456@192.168.0.1>;tag=z9hG4bK-1222-46d60c-336097d9-b4023098
CSeq: 1 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1741093061/22126c4a969e0802181f360a2c3a2eb8",opaque="68404dc067f738ad",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (649 bytes) from UDP:111.111.11.1:5065 --->
ACK sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=b405d6c8-74aa87d-13c9-83545D-571a4f6e-5fe9e490-46d60c
To: "815123456"<sip:815123456@192.168.0.1>;tag=z9hG4bK-1222-46d60c-336097d9-b4023098
Call-ID: b40916b0-74aa87d-13c9-83545D-76787d14-43ef54ca-46d60c
CSeq: 1 ACK
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1222-46d60c-336097d9-b4023098
Max-Forwards: 70
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0


<--- Received SIP request (1346 bytes) from UDP:111.111.11.1:5065 --->
INVITE sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=b405d6c8-74aa87d-13c9-83545D-571a4f6e-5fe9e490-46d60c
To: "815123456"<sip:815123456@192.168.0.1:5060>
Call-ID: b40916b0-74aa87d-13c9-83545D-76787d14-43ef54ca-46d60c
CSeq: 2 INVITE
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1222-46d617-40a8a07-b4023258
Max-Forwards: 70
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741093061/22126c4a969e0802181f360a2c3a2eb8",uri="sip:815123456@192.168.0.1:5060",response="4a49ed6b397f2dd1255328040c880f27",algorithm=MD5,cnonce="46d617",opaque="68404dc067f738ad",qop=auth,nc=00000001
Content-Type: application/sdp
Content-Length: 400

v=0
o=123456 1741093624 1741093624 IN IP4 111.111.11.1
s=-
c=IN IP4 111.111.11.1
t=0 0
m=audio 41872 RTP/AVP 107 8 0 18 96 108
a=rtpmap:107 AMR-WB/16000
a=fmtp:107 channels=1;mode-change-capability=2;max-red=220
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtpmap:108 telephone-event/16000
a=fmtp:108 0-15
a=maxptime:240
a=silenceSupp:off - - - -
a=ptime:20

  == Setting global variable 'SIPDOMAIN' to '192.168.0.1'
<--- Transmitting SIP response (408 bytes) to UDP:111.111.11.1:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.11.1:5065;rport=5065;received=111.111.11.1;branch=z9hG4bK-1222-46d617-40a8a07-b4023258
Call-ID: b40916b0-74aa87d-13c9-83545D-76787d14-43ef54ca-46d60c
From: "home_KL" <sip:home_KL@192.168.0.1>;tag=b405d6c8-74aa87d-13c9-83545D-571a4f6e-5fe9e490-46d60c
To: "815123456" <sip:815123456@192.168.0.1>
CSeq: 2 INVITE
Server: Asterisk PBX 15.3.0
Content-Length:  0


    -- Executing [815123456@Phones:1] NoOp("PJSIP/home_KL-00000018", "home_KL") in new stack
    -- Executing [815123456@Phones:2] Set("PJSIP/home_KL-00000018", "CALLERID(num)=my_number") in new stack
    -- Executing [815123456@Phones:3] Goto("PJSIP/home_KL-00000018", "outgoing,815123456,2") in new stack
    -- Goto (outgoing,815123456,2)
    -- Executing [815123456@outgoing:2] Dial("PJSIP/home_KL-00000018", "PJSIP/815123456@Tan") in new stack
    -- Called PJSIP/815123456@Tan
<--- Transmitting SIP request (951 bytes) to UDP:222.222.222.22:5060 --->
INVITE sip:815123456@my.isp.com SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;rport;branch=z9hG4bKPje25641d5-b64e-4959-9a72-33a5792be927
From: "home_KL" <sip:my_number@192.168.0.1>;tag=03f5390a-3158-4b8e-abf3-ee7f20038e29
To: <sip:815123456@my.isp.com>
Contact: <sip:asterisk@111.111.11.1:5060>
Call-ID: 1679bbaa-adee-404e-8b0a-54df9a557455
CSeq: 17840 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 15.3.0
Content-Type: application/sdp
Content-Length:   259

v=0
o=- 803135795 803135795 IN IP4 111.111.11.1
s=Asterisk
c=IN IP4 111.111.11.1
t=0 0
m=audio 18224 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (343 bytes) from UDP:222.222.222.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bKPje25641d5-b64e-4959-9a72-33a5792be927;rport=5060
From: "home_KL" <sip:my_number@192.168.0.1>;tag=03f5390a-3158-4b8e-abf3-ee7f20038e29
To: <sip:815123456@my.isp.com>
Call-ID: 1679bbaa-adee-404e-8b0a-54df9a557455
CSeq: 17840 INVITE


<--- Received SIP response (374 bytes) from UDP:222.222.222.22:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bKPje25641d5-b64e-4959-9a72-33a5792be927;rport=5060
From: "home_KL" <sip:my_number@192.168.0.1>;tag=03f5390a-3158-4b8e-abf3-ee7f20038e29
To: <sip:815123456@my.isp.com>;tag=aprqngfrt-90nnps2008h0b
Call-ID: 1679bbaa-adee-404e-8b0a-54df9a557455
CSeq: 17840 INVITE


<--- Transmitting SIP request (443 bytes) to UDP:222.222.222.22:5060 --->
ACK sip:815123456@my.isp.com SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;rport;branch=z9hG4bKPje25641d5-b64e-4959-9a72-33a5792be927
From: "home_KL" <sip:my_number@192.168.0.1>;tag=03f5390a-3158-4b8e-abf3-ee7f20038e29
To: <sip:815123456@my.isp.com>;tag=aprqngfrt-90nnps2008h0b
Call-ID: 1679bbaa-adee-404e-8b0a-54df9a557455
CSeq: 17840 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 15.3.0
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/home_KL-00000018' status is 'CHANUNAVAIL'
<--- Transmitting SIP response (486 bytes) to UDP:111.111.11.1:5065 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 111.111.11.1:5065;rport=5065;received=111.111.11.1;branch=z9hG4bK-1222-46d617-40a8a07-b4023258
Call-ID: b40916b0-74aa87d-13c9-83545D-76787d14-43ef54ca-46d60c
From: "home_KL" <sip:home_KL@192.168.0.1>;tag=b405d6c8-74aa87d-13c9-83545D-571a4f6e-5fe9e490-46d60c
To: "815123456" <sip:815123456@192.168.0.1>;tag=56dae352-f661-4329-b021-ede58c76ea10
CSeq: 2 INVITE
Server: Asterisk PBX 15.3.0
Reason: Q.850;cause=34
Content-Length:  0


<--- Received SIP request (647 bytes) from UDP:111.111.11.1:5065 --->
ACK sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=b405d6c8-74aa87d-13c9-83545D-571a4f6e-5fe9e490-46d60c
To: "815123456"<sip:815123456@192.168.0.1>;tag=56dae352-f661-4329-b021-ede58c76ea10
Call-ID: b40916b0-74aa87d-13c9-83545D-76787d14-43ef54ca-46d60c
CSeq: 2 ACK
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1222-46d617-40a8a07-b4023258
Max-Forwards: 70
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0


<--- Received SIP request (809 bytes) from UDP:111.111.11.1:5065 --->
REGISTER sip:192.168.0.1 SIP/2.0
From: <sip:home_KL@192.168.0.1>;tag=b405cdb8-74aa87d-13c9-83545D-bb65967-385c001d-41a5e0
To: <sip:home_KL@192.168.0.1>
Call-ID: b40aab78-74aa87d-13c9-83545D-4e3d6305-7388e187-41a5e0
CSeq: 3 REGISTER
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1222-46d84c-17d8264d-b4022b58
Max-Forwards: 70
Supported: replaces,100rel
X-Serialnumber: 55 23 56 89 56
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
Expires: 0
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741092721/db88bf9d1ad846fc1360865ebf1ff88f",uri="sip:192.168.0.1:5060",response="4ac85a9ebb4c4772f011f81e2dc9d4b6",algorithm=MD5,cnonce="46d84d",opaque="7c6709f515df9e70",qop=auth,nc=00000002
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0


<--- Transmitting SIP response (594 bytes) to UDP:111.111.11.1:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.11.1:5065;rport=5065;received=111.111.11.1;branch=z9hG4bK-1222-46d84c-17d8264d-b4022b58
Call-ID: b40aab78-74aa87d-13c9-83545D-4e3d6305-7388e187-41a5e0
From: <sip:home_KL@192.168.0.1>;tag=b405cdb8-74aa87d-13c9-83545D-bb65967-385c001d-41a5e0
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-1222-46d84c-17d8264d-b4022b58
CSeq: 3 REGISTER
WWW-Authenticate: Digest  realm="asterisk",nonce="1741093062/1afc67560bf324322be07df2cb974a08",opaque="52cf85580f42f8e6",stale=true,algorithm=md5,qop="auth"
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (809 bytes) from UDP:111.111.11.1:5065 --->
REGISTER sip:192.168.0.1 SIP/2.0
From: <sip:home_KL@192.168.0.1>;tag=b405cdb8-74aa87d-13c9-83545D-bb65967-385c001d-41a5e0
To: <sip:home_KL@192.168.0.1>
Call-ID: b40aab78-74aa87d-13c9-83545D-4e3d6305-7388e187-41a5e0
CSeq: 4 REGISTER
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1222-46d856-2561cb43-b4023418
Max-Forwards: 70
Supported: replaces,100rel
X-Serialnumber: 55 23 56 89 56
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
Expires: 0
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741093062/1afc67560bf324322be07df2cb974a08",uri="sip:192.168.0.1:5060",response="0afa9608c372be310df11b71f2f33f51",algorithm=MD5,cnonce="46d856",opaque="52cf85580f42f8e6",qop=auth,nc=00000001
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0


    -- Removed contact 'sip:home_KL@111.111.11.1:5065' from AOR 'home_KL' due to request
  == Contact home_KL/sip:home_KL@111.111.11.1:5065 has been deleted
<--- Transmitting SIP response (503 bytes) to UDP:111.111.11.1:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.11.1:5065;rport=5065;received=111.111.11.1;branch=z9hG4bK-1222-46d856-2561cb43-b4023418
Call-ID: b40aab78-74aa87d-13c9-83545D-4e3d6305-7388e187-41a5e0
From: <sip:home_KL@192.168.0.1>;tag=b405cdb8-74aa87d-13c9-83545D-bb65967-385c001d-41a5e0
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-1222-46d856-2561cb43-b4023418
CSeq: 4 REGISTER
Date: Tue, 04 Mar 2025 12:57:42 GMT
Contact: <sip:192.168.0.1>
Expires: 0
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (552 bytes) from UDP:111.111.11.1:5065 --->
REGISTER sip:192.168.0.1 SIP/2.0
From: <sip:home_KL@192.168.0.1>;tag=b405dc38-74aa87d-13c9-83545D-40b343eb-33b25f68-46d878
To: <sip:home_KL@192.168.0.1>
Call-ID: b40aad20-74aa87d-13c9-83545D-37cebe7-3c595222-46d878
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1222-46d879-503ab5d1-b40235d8
Max-Forwards: 70
Supported: replaces,100rel
X-Serialnumber: 55 23 56 89 56
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
Expires: 600
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0


<--- Transmitting SIP response (583 bytes) to UDP:111.111.11.1:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.11.1:5065;rport=5065;received=111.111.11.1;branch=z9hG4bK-1222-46d879-503ab5d1-b40235d8
Call-ID: b40aad20-74aa87d-13c9-83545D-37cebe7-3c595222-46d878
From: <sip:home_KL@192.168.0.1>;tag=b405dc38-74aa87d-13c9-83545D-40b343eb-33b25f68-46d878
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-1222-46d879-503ab5d1-b40235d8
CSeq: 1 REGISTER
WWW-Authenticate: Digest  realm="asterisk",nonce="1741093062/1afc67560bf324322be07df2cb974a08",opaque="262113eb5ce48ed8",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (811 bytes) from UDP:111.111.11.1:5065 --->
REGISTER sip:192.168.0.1 SIP/2.0
From: <sip:home_KL@192.168.0.1>;tag=b405dc38-74aa87d-13c9-83545D-40b343eb-33b25f68-46d878
To: <sip:home_KL@192.168.0.1>
Call-ID: b40aad20-74aa87d-13c9-83545D-37cebe7-3c595222-46d878
CSeq: 2 REGISTER
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1222-46d895-67348f7f-b4023798
Max-Forwards: 70
Supported: replaces,100rel
X-Serialnumber: 55 23 56 89 56
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
Expires: 600
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741093062/1afc67560bf324322be07df2cb974a08",uri="sip:192.168.0.1:5060",response="1df7e32e27f5614629f2cf4d08ca0393",algorithm=MD5,cnonce="46d895",opaque="262113eb5ce48ed8",qop=auth,nc=00000001
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0


    -- Added contact 'sip:home_KL@111.111.11.1:5065' to AOR 'home_KL' with expiration of 600 seconds
  == Contact home_KL/sip:home_KL@111.111.11.1:5065 has been created
<--- Transmitting SIP response (559 bytes) to UDP:111.111.11.1:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.11.1:5065;rport=5065;received=111.111.11.1;branch=z9hG4bK-1222-46d895-67348f7f-b4023798
Call-ID: b40aad20-74aa87d-13c9-83545D-37cebe7-3c595222-46d878
From: <sip:home_KL@192.168.0.1>;tag=b405dc38-74aa87d-13c9-83545D-40b343eb-33b25f68-46d878
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-1222-46d895-67348f7f-b4023798
CSeq: 2 REGISTER
Date: Tue, 04 Mar 2025 12:57:42 GMT
Contact: <sip:home_KL@111.111.11.1:5065>;expires=599
Contact: <sip:192.168.0.1>
Expires: 600
Server: Asterisk PBX 15.3.0
Content-Length:  0


    -- Contact home_KL/sip:home_KL@111.111.11.1:5065 is now Unknown.  RTT: 0.000 msec
<--- Received SIP request (497 bytes) from UDP:111.111.11.1:5065 --->
SUBSCRIBE sip:home_KL@192.168.0.1:5060 SIP/2.0
From: <sip:home_KL@192.168.0.1:5060>;tag=b405e1a8-74aa87d-13c9-83545D-51e2e91e-52b0d7e6-46dc96
To: <sip:home_KL@192.168.0.1:5060>
Call-ID: b40918f8-74aa87d-13c9-83545D-7dcc865-80575c8-46dc96
CSeq: 1 SUBSCRIBE
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1223-46dc96-19db4040-b4023b18
Expires: 86400
Event: message-summary
Max-Forwards: 70
Supported: replaces,100rel
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0


<--- Transmitting SIP response (583 bytes) to UDP:111.111.11.1:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.11.1:5065;rport=5065;received=111.111.11.1;branch=z9hG4bK-1223-46dc96-19db4040-b4023b18
Call-ID: b40918f8-74aa87d-13c9-83545D-7dcc865-80575c8-46dc96
From: <sip:home_KL@192.168.0.1>;tag=b405e1a8-74aa87d-13c9-83545D-51e2e91e-52b0d7e6-46dc96
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-1223-46dc96-19db4040-b4023b18
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest  realm="asterisk",nonce="1741093063/b75d7523041e880a3137ca2ed1780e44",opaque="0b5880583cee0934",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (764 bytes) from UDP:111.111.11.1:5065 --->
SUBSCRIBE sip:home_KL@192.168.0.1:5060 SIP/2.0
From: <sip:home_KL@192.168.0.1:5060>;tag=b405e1a8-74aa87d-13c9-83545D-51e2e91e-52b0d7e6-46dc96
To: <sip:home_KL@192.168.0.1:5060>
Call-ID: b40918f8-74aa87d-13c9-83545D-7dcc865-80575c8-46dc96
CSeq: 2 SUBSCRIBE
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1223-46dc9d-57281166-b4023cd8
Expires: 86400
Event: message-summary
Max-Forwards: 70
Supported: replaces,100rel
Contact: <sip:home_KL@111.111.11.1:5065>
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741093063/b75d7523041e880a3137ca2ed1780e44",uri="sip:home_KL@192.168.0.1:5060",response="f703d9d8bd242b24d5dd571755fb56f5",algorithm=MD5,cnonce="46dc9d",opaque="0b5880583cee0934",qop=auth,nc=00000001
Content-Length: 0


<--- Transmitting SIP response (433 bytes) to UDP:111.111.11.1:5065 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 111.111.11.1:5065;rport=5065;received=111.111.11.1;branch=z9hG4bK-1223-46dc9d-57281166-b4023cd8
Call-ID: b40918f8-74aa87d-13c9-83545D-7dcc865-80575c8-46dc96
From: <sip:home_KL@192.168.0.1>;tag=b405e1a8-74aa87d-13c9-83545D-51e2e91e-52b0d7e6-46dc96
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-1223-46dc9d-57281166-b4023cd8
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (811 bytes) from UDP:111.111.11.1:5065 --->
REGISTER sip:192.168.0.1 SIP/2.0
From: <sip:home_KL@192.168.0.1>;tag=b405dc38-74aa87d-13c9-83545D-40b343eb-33b25f68-46d878
To: <sip:home_KL@192.168.0.1>
Call-ID: b40aad20-74aa87d-13c9-83545D-37cebe7-3c595222-46d878
CSeq: 3 REGISTER
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1451-4f6046-6e06e2f7-b4023958
Max-Forwards: 70
Supported: replaces,100rel
X-Serialnumber: 55 23 56 89 56
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
Expires: 600
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741093062/1afc67560bf324322be07df2cb974a08",uri="sip:192.168.0.1:5060",response="96584efbad2966ad9c4488c062a3441c",algorithm=MD5,cnonce="4f6047",opaque="262113eb5ce48ed8",qop=auth,nc=00000002
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0


<--- Transmitting SIP response (594 bytes) to UDP:111.111.11.1:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.11.1:5065;rport=5065;received=111.111.11.1;branch=z9hG4bK-1451-4f6046-6e06e2f7-b4023958
Call-ID: b40aad20-74aa87d-13c9-83545D-37cebe7-3c595222-46d878
From: <sip:home_KL@192.168.0.1>;tag=b405dc38-74aa87d-13c9-83545D-40b343eb-33b25f68-46d878
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-1451-4f6046-6e06e2f7-b4023958
CSeq: 3 REGISTER
WWW-Authenticate: Digest  realm="asterisk",nonce="1741093621/5c5aa8d9d19bf32f93b44aabf99e5795",opaque="47d493af60faee7e",stale=true,algorithm=md5,qop="auth"
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (810 bytes) from UDP:111.111.11.1:5065 --->
REGISTER sip:192.168.0.1 SIP/2.0
From: <sip:home_KL@192.168.0.1>;tag=b405dc38-74aa87d-13c9-83545D-40b343eb-33b25f68-46d878
To: <sip:home_KL@192.168.0.1>
Call-ID: b40aad20-74aa87d-13c9-83545D-37cebe7-3c595222-46d878
CSeq: 4 REGISTER
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-1452-4f6051-f0d6905-b4023e98
Max-Forwards: 70
Supported: replaces,100rel
X-Serialnumber: 55 23 56 89 56
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
Expires: 600
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741093621/5c5aa8d9d19bf32f93b44aabf99e5795",uri="sip:192.168.0.1:5060",response="b44324078bcc568f9b2be0d547e96524",algorithm=MD5,cnonce="4f6051",opaque="47d493af60faee7e",qop=auth,nc=00000001
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0


<--- Transmitting SIP response (557 bytes) to UDP:111.111.11.1:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.11.1:5065;rport=5065;received=111.111.11.1;branch=z9hG4bK-1452-4f6051-f0d6905-b4023e98
Call-ID: b40aad20-74aa87d-13c9-83545D-37cebe7-3c595222-46d878
From: <sip:home_KL@192.168.0.1>;tag=b405dc38-74aa87d-13c9-83545D-40b343eb-33b25f68-46d878
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-1452-4f6051-f0d6905-b4023e98
CSeq: 4 REGISTER
Date: Tue, 04 Mar 2025 13:07:01 GMT
Contact: <sip:home_KL@111.111.11.1:5065>;expires=599
Contact: <sip:192.168.0.1>
Expires: 600
Server: Asterisk PBX 15.3.0
Content-Length:  0

legend-
815123456 is the number I am dialling.
111.111.11.1 is my IP

I noticed the lines-

SIP/2.0 503 Service Unavailable
Reason: Q.850;cause=34
SIP/2.0 401 Unauthorized

My pjsip.conf part for the offending provider (the same for the working provider)-

[global]
type=global

; Define the transport
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

; #### Phones ####

; ## home_KL (Phone) ##
[home_KL]
type=endpoint
context=Phones
disallow=all
allow=ulaw,alaw
auth=home_KL
aors=home_KL
direct_media=yes

[home_KL]
type=auth
auth_type=userpass
username=home_KL
password=<my_password>

[home_KL]
type=aor
max_contacts=1
contact=sip:192.168.0.1

; ## Service-Provider1 ##
[Service-Provider1]
type=endpoint
context=Service-Provider1
disallow=all
allow=ulaw,alaw
outbound_auth=Service-Provider1
auth=Service-Provider1
aors=Service-Provider1
direct_media=yes

[Service-Provider1]
type=auth
auth_type=userpass
username=<my_number>
password=<my_password>

[Service-Provider1]
type=aor
max_contacts=1
contact=sip:my.provider.com

My extesnions.conf part for the offending provider.-

; # Dial Extension Tan #
exten => _X.,1,NoOp(${CALLERID(num)})
same => n,Set(CALLERID(num)=<my_number>)
same => n,Goto(outgoing,${EXTEN},2)

; ## Outgoing calls ##

; # Outgoing Providers #
[outgoing]
exten => _X.,1,Dial(PJSIP/${EXTEN}@Other)
exten => _X.,2,Dial(PJSIP/${EXTEN}@Tan)

Can anyone tell me some of the main reasons as to why voip providers can reject PJSIP as opposed to chan_sip, which in this case works fine, so I can troubleshoot this issue and one by one try solutions.
I have done some reading but am stuck, after reading the log file I do not believe CALLERID is the issue.
Any help is appreciated.

david551 · March 5, 2025, 2:13pm

Some providers try to read more into the Contact header than the SIP standards allow. You might have to set an option to use the caller ID, rather than “asterisk”.

PatTheRat · March 5, 2025, 11:23pm

Hi David, thanks for the reply.
Any idea how to do this?
I did find this in a search.-

So am going to try-
contact_user=<my_number>
and I am guessing it goes in the pjsip.conf like this-

[Service-Provider1]
type=endpoint
context=Service-Provider1
disallow=all
allow=ulaw,alaw
outbound_auth=Service-Provider1
auth=Service-Provider1
aors=Service-Provider1
direct_media=yes
contact_user=<my_number>

Would that be right?

PatTheRat · March 6, 2025, 12:08pm

Ok so I tried adding

contact_user=<my_number>

to the

[Service-Provider1]

like this-

[Service-Provider1]
type=endpoint
context=Service-Provider1
disallow=all
allow=ulaw,alaw
outbound_auth=Service-Provider1
auth=Service-Provider1
aors=Service-Provider1
direct_media=yes
contact_user=<my_number>

but it does not work.

Any other suggestions?

jcolp · March 6, 2025, 12:15pm

If you provide a working case from chan_sip and its configuration, then we may be able to spot the differences.

PatTheRat · March 6, 2025, 12:28pm

I will get one now.

PatTheRat · March 6, 2025, 1:06pm

I made a call using chan_sip with the same provider and the call goes through.
I ran-

sip set debug on

and got-


<--- SIP read from UDP:111.111.11.1:5065 --->
INVITE sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 1 INVITE
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-7aeb-1e0299e-3837288a-10f8cc0
Max-Forwards: 70
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Type: application/sdp
Content-Length: 376

v=0
o=123456 1741264859 1741264859 IN IP4 111.111.11.1
s=-
c=IN IP4 111.111.11.1
t=0 0
m=audio 64350 RTP/AVP 107 8 0 18 96 108
a=rtpmap:107 AMR-WB/16000
a=fmtp:107 channels=1;mode-change-capability=2;max-red=220
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtpmap:108 telephone-event/16000
a=fmtp:108 0-15
a=maxptime:240
a=ptime:20
<------------->
--- (14 headers 15 lines) ---
Sending to 111.111.11.1:5065 (no NAT)
Sending to 111.111.11.1:5065 (no NAT)
Using INVITE request as basis request - 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
Found peer 'home_KL' for 'home_KL' from 111.111.11.1:5065

<--- Reliably Transmitting (no NAT) to 111.111.11.1:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.11.1:5065;branch=z9hG4bK-7aeb-1e0299e-3837288a-10f8cc0;received=111.111.11.1;rport=5065
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as01b8088d
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 1 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a1073e9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:111.111.11.1:5065 --->
ACK sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as01b8088d
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 1 ACK
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-7aeb-1e0299e-3837288a-10f8cc0
Max-Forwards: 70
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- SIP read from UDP:111.111.11.1:5065 --->
INVITE sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 2 INVITE
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-7aeb-1e029a8-313b3450-10f8e80
Max-Forwards: 70
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Authorization: Digest username="home_KL",realm="asterisk",nonce="1a1073e9",uri="sip:815123456@192.168.0.1:5060",response="040c54c89e98eb6c295cb6cd43b81dd8",algorithm=MD5
Content-Type: application/sdp
Content-Length: 376

v=0
o=123456 1741264859 1741264859 IN IP4 111.111.11.1
s=-
c=IN IP4 111.111.11.1
t=0 0
m=audio 64350 RTP/AVP 107 8 0 18 96 108
a=rtpmap:107 AMR-WB/16000
a=fmtp:107 channels=1;mode-change-capability=2;max-red=220
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtpmap:108 telephone-event/16000
a=fmtp:108 0-15
a=maxptime:240
a=ptime:20
<------------->
--- (15 headers 15 lines) ---
Sending to 111.111.11.1:5065 (no NAT)
Using INVITE request as basis request - 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
Found peer 'home_KL' for 'home_KL' from 111.111.11.1:5065
  == Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 108
Found unknown media description format AMR-WB for ID 107
Found audio description format telephone-event for ID 96
Found unknown media description format telephone-event for ID 108
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 111.111.11.1:64350
Looking for 815123456 in phones (domain 192.168.0.1)
sip_route_dump: route/path hop: <sip:home_KL@111.111.11.1:5065>

<--- Transmitting (no NAT) to 111.111.11.1:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.11.1:5065;branch=z9hG4bK-7aeb-1e029a8-313b3450-10f8e80;received=111.111.11.1;rport=5065
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 2 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:815123456@111.111.11.1:5060>
Content-Length: 0


<------------>
    -- Executing [815123456@phones:1] NoOp("SIP/home_KL-0000000a", "home_KL") in new stack
    -- Executing [815123456@phones:2] Set("SIP/home_KL-0000000a", "CALLERID(num)=<my_number>") in new stack
    -- Executing [815123456@phones:3] Goto("SIP/home_KL-0000000a", "outgoing,815123456,2") in new stack
    -- Goto (outgoing,815123456,2)
    -- Executing [815123456@outgoing:2] Dial("SIP/home_KL-0000000a", "SIP/Tan/815123456") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 16470
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 222.222.222.22:5060:
INVITE sip:815123456@my.isp.com SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK0097d1f4;rport
Max-Forwards: 70
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>
Contact: <sip:<my_number>@111.111.11.1:5060>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.3.0
Date: Thu, 06 Mar 2025 12:32:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 431439887 431439887 IN IP4 111.111.11.1
s=Asterisk PBX 15.3.0
c=IN IP4 111.111.11.1
t=0 0
m=audio 16470 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/Tan/815123456

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK0097d1f4;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 102 INVITE

<------------->
--- (6 headers 0 lines) ---

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK0097d1f4;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-727597615-1741264351891
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 102 INVITE
WWW-Authenticate: DIGEST realm="my.isp.com",qop="auth",nonce="BroadWorksXm7xbu4r7Txqvrl4BW",algorithm=MD5
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 222.222.222.22:5060:
ACK sip:815123456@my.isp.com SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK0097d1f4;rport
Max-Forwards: 70
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-727597615-1741264351891
Contact: <sip:<my_number>@111.111.11.1:5060>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.3.0
Content-Length: 0


---
Audio is at 16470
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 222.222.222.22:5060:
INVITE sip:815123456@my.isp.com SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK12c56aba;rport
Max-Forwards: 70
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>
Contact: <sip:<my_number>@111.111.11.1:5060>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 15.3.0
Authorization: Digest username="<my_number>", realm="my.isp.com", algorithm=MD5, uri="sip:815123456@my.isp.com", nonce="BroadWorksXm7xbu4r7Txqvrl4BW", response="e194b1763d63d655c4c43e4df081a904", qop=auth, cnonce="4d0c0e4d", nc=00000001
Date: Thu, 06 Mar 2025 12:32:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 431439887 431439888 IN IP4 111.111.11.1
s=Asterisk PBX 15.3.0
c=IN IP4 111.111.11.1
t=0 0
m=audio 16470 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK12c56aba;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 103 INVITE

<------------->
--- (6 headers 0 lines) ---
Retransmitting #2 (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK2823eed7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as45eb32fc
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 6ebe468100228d4546986ab516b95910@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Thu, 06 Mar 2025 12:32:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK2823eed7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as45eb32fc
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 6ebe468100228d4546986ab516b95910@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Thu, 06 Mar 2025 12:32:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK12c56aba;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-1039748949-1741264352982
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 103 INVITE
Supported: timer
Contact: <sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp>
Session: Media
P-Asserted-Identity: <sip:815123456@my.isp.com;user=phone>
Privacy: none
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Type: application/sdp
Content-Length: 270

v=0
o=BroadWorks 1611506149 1 IN IP4 222.222.222.22
s=-
c=IN IP4 222.222.222.22
t=0 0
m=audio 20934 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
a=bsoft: 1 image udptl t38
<------------->
--- (14 headers 13 lines) ---
sip_route_dump: route/path hop: <sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp>
Found RTP audio format 0
Found RTP audio format 96
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 222.222.222.22:20934
    -- SIP/Tan-0000000b is making progress passing it to SIP/home_KL-0000000a
Audio is at 10236
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 111.111.11.1:5065 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 111.111.11.1:5065;branch=z9hG4bK-7aeb-1e029a8-313b3450-10f8e80;received=111.111.11.1;rport=5065
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as0cf7cef9
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 2 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:815123456@111.111.11.1:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 2103517998 2103517998 IN IP4 111.111.11.1
s=Asterisk PBX 15.3.0
c=IN IP4 111.111.11.1
t=0 0
m=audio 10236 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK12c56aba;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-1039748949-1741264352982
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 103 INVITE
Require: timer
Session-Expires: 1800;refresher=uac
Supported: timer
Contact: <sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp>
P-Asserted-Identity: <sip:815123456@my.isp.com;user=phone>
Privacy: none
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp
Content-Type: application/sdp
Content-Length: 270

v=0
o=BroadWorks 1611506149 1 IN IP4 222.222.222.22
s=-
c=IN IP4 222.222.222.22
t=0 0
m=audio 20934 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
a=bsoft: 1 image udptl t38
<------------->
--- (16 headers 13 lines) ---
sip_route_dump: route/path hop: <sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp>
Transmitting (NAT) to 222.222.222.22:5060:
ACK sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK28516ecb;rport
Max-Forwards: 70
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-1039748949-1741264352982
Contact: <sip:<my_number>@111.111.11.1:5060>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 103 ACK
User-Agent: Asterisk PBX 15.3.0
Content-Length: 0


---
    -- SIP/Tan-0000000b answered SIP/home_KL-0000000a
Audio is at 10236
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 111.111.11.1:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.11.1:5065;branch=z9hG4bK-7aeb-1e029a8-313b3450-10f8e80;received=111.111.11.1;rport=5065
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as0cf7cef9
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 2 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:815123456@111.111.11.1:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 2103517998 2103517998 IN IP4 111.111.11.1
s=Asterisk PBX 15.3.0
c=IN IP4 111.111.11.1
t=0 0
m=audio 10236 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:111.111.11.1:5065 --->
ACK sip:815123456@111.111.11.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as0cf7cef9
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 2 ACK
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-7aee-1e03225-7cc0c99-10f8e80
Max-Forwards: 70
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Authorization: Digest username="home_KL",realm="asterisk",nonce="1a1073e9",uri="sip:815123456@192.168.0.1:5060",response="040c54c89e98eb6c295cb6cd43b81dd8",algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
    -- Channel SIP/Tan-0000000b joined 'simple_bridge' basic-bridge <4842527a-6cf1-41e7-a3fa-f0232b4c91c9>
    -- Channel SIP/home_KL-0000000a joined 'simple_bridge' basic-bridge <4842527a-6cf1-41e7-a3fa-f0232b4c91c9>
Retransmitting #4 (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK2823eed7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as45eb32fc
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 6ebe468100228d4546986ab516b95910@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Thu, 06 Mar 2025 12:32:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '6ebe468100228d4546986ab516b95910@172.23.14.1:5060' Method: OPTIONS
[Mar  6 23:32:34] WARNING[1888][C-00000007]: chan_sip.c:7536 sip_write: Can't send 10 type frames with SIP write
[Mar  6 23:32:35] WARNING[1888][C-00000007]: chan_sip.c:7536 sip_write: Can't send 10 type frames with SIP write
[Mar  6 23:32:36] WARNING[1888][C-00000007]: chan_sip.c:7536 sip_write: Can't send 10 type frames with SIP write
[Mar  6 23:32:36] WARNING[1888][C-00000007]: chan_sip.c:7536 sip_write: Can't send 10 type frames with SIP write

<--- SIP read from UDP:111.111.11.1:5065 --->
BYE sip:815123456@111.111.11.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as0cf7cef9
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 3 BYE
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-7af1-1e03f54-2a988ac7-10f9040
Max-Forwards: 70
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Authorization: Digest username="home_KL",realm="asterisk",nonce="1a1073e9",uri="sip:815123456@111.111.11.1:5060",response="ce5de129e744a7cd8739ef645defa0ba",algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 111.111.11.1:5065 (no NAT)
Scheduling destruction of SIP dialog '116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 111.111.11.1:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.11.1:5065;branch=z9hG4bK-7af1-1e03f54-2a988ac7-10f9040;received=111.111.11.1;rport=5065
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as0cf7cef9
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 3 BYE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/home_KL-0000000a left 'simple_bridge' basic-bridge <4842527a-6cf1-41e7-a3fa-f0232b4c91c9>
  == Spawn extension (outgoing, 815123456, 2) exited non-zero on 'SIP/home_KL-0000000a'
    -- Channel SIP/Tan-0000000b left 'simple_bridge' basic-bridge <4842527a-6cf1-41e7-a3fa-f0232b4c91c9>
Scheduling destruction of SIP dialog '3db24116003041fc50f03d3f3327a5c8@my.isp.com' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 222.222.222.22:5060:
BYE sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK352de9d3;rport
Max-Forwards: 70
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-1039748949-1741264352982
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 104 BYE
User-Agent: Asterisk PBX 15.3.0
Authorization: Digest username="<my_number>", realm="my.isp.com", algorithm=MD5, uri="sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060", nonce="BroadWorksXm7xbu4r7Txqvrl4BW", response="6b00baf8ea1fbd69bef9379e0359f704", qop=auth, cnonce="360eb6d5", nc=00000002
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK352de9d3;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-1039748949-1741264352982
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 104 BYE
Content-Length: 0

Legend-

the number I am dialling= 815123456
my ip=111.111.11.1

Hope someone can figure it out for me.

jcolp · March 6, 2025, 1:10pm

Set from_user to your number on the PJSIP endpoint, and set from_domain to the hostname of the provider.

PatTheRat · March 6, 2025, 1:12pm

Will try it out

PatTheRat · March 6, 2025, 1:25pm

Well done, it works great! Thank you.

PatTheRat · March 11, 2025, 6:28am

Ah! worked until the next reboot of the device now it doesn’t work also incoming calls do not work. I have been trying to fix it for days but can not get it to work. It might dial out once and not work again. I thought chan_sip was annoying, PJsip is ten times worse.

jcolp · March 11, 2025, 8:33am

You’d need to show… anything.

PatTheRat · March 11, 2025, 8:38am

Yes I will post some files, but need a break from it as it’s driving me batty.

PatTheRat · March 12, 2025, 7:59am

Here are the files that come from my SIP setup which is working for both incoming and outgoing calls I will have to sperad it over several replys-

Log file for outgoing call (SIP)-


<--- SIP read from UDP:111.111.11.1:5065 --->
INVITE sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 1 INVITE
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-7aeb-1e0299e-3837288a-10f8cc0
Max-Forwards: 70
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Type: application/sdp
Content-Length: 376

v=0
o=123456 1741264859 1741264859 IN IP4 111.111.11.1
s=-
c=IN IP4 111.111.11.1
t=0 0
m=audio 64350 RTP/AVP 107 8 0 18 96 108
a=rtpmap:107 AMR-WB/16000
a=fmtp:107 channels=1;mode-change-capability=2;max-red=220
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtpmap:108 telephone-event/16000
a=fmtp:108 0-15
a=maxptime:240
a=ptime:20
<------------->
--- (14 headers 15 lines) ---
Sending to 111.111.11.1:5065 (no NAT)
Sending to 111.111.11.1:5065 (no NAT)
Using INVITE request as basis request - 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
Found peer 'home_KL' for 'home_KL' from 111.111.11.1:5065

<--- Reliably Transmitting (no NAT) to 111.111.11.1:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.11.1:5065;branch=z9hG4bK-7aeb-1e0299e-3837288a-10f8cc0;received=111.111.11.1;rport=5065
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as01b8088d
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 1 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a1073e9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:111.111.11.1:5065 --->
ACK sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as01b8088d
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 1 ACK
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-7aeb-1e0299e-3837288a-10f8cc0
Max-Forwards: 70
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- SIP read from UDP:111.111.11.1:5065 --->
INVITE sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 2 INVITE
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-7aeb-1e029a8-313b3450-10f8e80
Max-Forwards: 70
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Authorization: Digest username="home_KL",realm="asterisk",nonce="1a1073e9",uri="sip:815123456@192.168.0.1:5060",response="040c54c89e98eb6c295cb6cd43b81dd8",algorithm=MD5
Content-Type: application/sdp
Content-Length: 376

v=0
o=123456 1741264859 1741264859 IN IP4 111.111.11.1
s=-
c=IN IP4 111.111.11.1
t=0 0
m=audio 64350 RTP/AVP 107 8 0 18 96 108
a=rtpmap:107 AMR-WB/16000
a=fmtp:107 channels=1;mode-change-capability=2;max-red=220
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtpmap:108 telephone-event/16000
a=fmtp:108 0-15
a=maxptime:240
a=ptime:20
<------------->
--- (15 headers 15 lines) ---
Sending to 111.111.11.1:5065 (no NAT)
Using INVITE request as basis request - 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
Found peer 'home_KL' for 'home_KL' from 111.111.11.1:5065
  == Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 108
Found unknown media description format AMR-WB for ID 107
Found audio description format telephone-event for ID 96
Found unknown media description format telephone-event for ID 108
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 111.111.11.1:64350
Looking for 815123456 in phones (domain 192.168.0.1)
sip_route_dump: route/path hop: <sip:home_KL@111.111.11.1:5065>

<--- Transmitting (no NAT) to 111.111.11.1:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.11.1:5065;branch=z9hG4bK-7aeb-1e029a8-313b3450-10f8e80;received=111.111.11.1;rport=5065
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 2 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:815123456@111.111.11.1:5060>
Content-Length: 0


<------------>
    -- Executing [815123456@phones:1] NoOp("SIP/home_KL-0000000a", "home_KL") in new stack
    -- Executing [815123456@phones:2] Set("SIP/home_KL-0000000a", "CALLERID(num)=<my_number>") in new stack
    -- Executing [815123456@phones:3] Goto("SIP/home_KL-0000000a", "outgoing,815123456,2") in new stack
    -- Goto (outgoing,815123456,2)
    -- Executing [815123456@outgoing:2] Dial("SIP/home_KL-0000000a", "SIP/Tan/815123456") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 16470
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 222.222.222.22:5060:
INVITE sip:815123456@my.isp.com SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK0097d1f4;rport
Max-Forwards: 70
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>
Contact: <sip:<my_number>@111.111.11.1:5060>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.3.0
Date: Thu, 06 Mar 2025 12:32:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 431439887 431439887 IN IP4 111.111.11.1
s=Asterisk PBX 15.3.0
c=IN IP4 111.111.11.1
t=0 0
m=audio 16470 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/Tan/815123456

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK0097d1f4;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 102 INVITE

<------------->
--- (6 headers 0 lines) ---

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK0097d1f4;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-727597615-1741264351891
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 102 INVITE
WWW-Authenticate: DIGEST realm="my.isp.com",qop="auth",nonce="BroadWorksXm7xbu4r7Txqvrl4BW",algorithm=MD5
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 222.222.222.22:5060:
ACK sip:815123456@my.isp.com SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK0097d1f4;rport
Max-Forwards: 70
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-727597615-1741264351891
Contact: <sip:<my_number>@111.111.11.1:5060>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.3.0
Content-Length: 0


---
Audio is at 16470
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 222.222.222.22:5060:
INVITE sip:815123456@my.isp.com SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK12c56aba;rport
Max-Forwards: 70
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>
Contact: <sip:<my_number>@111.111.11.1:5060>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 15.3.0
Authorization: Digest username="<my_number>", realm="my.isp.com", algorithm=MD5, uri="sip:815123456@my.isp.com", nonce="BroadWorksXm7xbu4r7Txqvrl4BW", response="e194b1763d63d655c4c43e4df081a904", qop=auth, cnonce="4d0c0e4d", nc=00000001
Date: Thu, 06 Mar 2025 12:32:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 431439887 431439888 IN IP4 111.111.11.1
s=Asterisk PBX 15.3.0
c=IN IP4 111.111.11.1
t=0 0
m=audio 16470 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK12c56aba;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 103 INVITE

<------------->
--- (6 headers 0 lines) ---
Retransmitting #2 (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK2823eed7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as45eb32fc
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 6ebe468100228d4546986ab516b95910@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Thu, 06 Mar 2025 12:32:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK2823eed7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as45eb32fc
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 6ebe468100228d4546986ab516b95910@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Thu, 06 Mar 2025 12:32:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK12c56aba;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-1039748949-1741264352982
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 103 INVITE
Supported: timer
Contact: <sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp>
Session: Media
P-Asserted-Identity: <sip:815123456@my.isp.com;user=phone>
Privacy: none
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Type: application/sdp
Content-Length: 270

v=0
o=BroadWorks 1611506149 1 IN IP4 222.222.222.22
s=-
c=IN IP4 222.222.222.22
t=0 0
m=audio 20934 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
a=bsoft: 1 image udptl t38
<------------->
--- (14 headers 13 lines) ---
sip_route_dump: route/path hop: <sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp>
Found RTP audio format 0
Found RTP audio format 96
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 222.222.222.22:20934
    -- SIP/Tan-0000000b is making progress passing it to SIP/home_KL-0000000a
Audio is at 10236
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 111.111.11.1:5065 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 111.111.11.1:5065;branch=z9hG4bK-7aeb-1e029a8-313b3450-10f8e80;received=111.111.11.1;rport=5065
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as0cf7cef9
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 2 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:815123456@111.111.11.1:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 2103517998 2103517998 IN IP4 111.111.11.1
s=Asterisk PBX 15.3.0
c=IN IP4 111.111.11.1
t=0 0
m=audio 10236 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK12c56aba;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-1039748949-1741264352982
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 103 INVITE
Require: timer
Session-Expires: 1800;refresher=uac
Supported: timer
Contact: <sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp>
P-Asserted-Identity: <sip:815123456@my.isp.com;user=phone>
Privacy: none
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp
Content-Type: application/sdp
Content-Length: 270

v=0
o=BroadWorks 1611506149 1 IN IP4 222.222.222.22
s=-
c=IN IP4 222.222.222.22
t=0 0
m=audio 20934 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
a=bsoft: 1 image udptl t38
<------------->
--- (16 headers 13 lines) ---
sip_route_dump: route/path hop: <sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp>
Transmitting (NAT) to 222.222.222.22:5060:
ACK sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK28516ecb;rport
Max-Forwards: 70
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-1039748949-1741264352982
Contact: <sip:<my_number>@111.111.11.1:5060>
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 103 ACK
User-Agent: Asterisk PBX 15.3.0
Content-Length: 0


---
    -- SIP/Tan-0000000b answered SIP/home_KL-0000000a
Audio is at 10236
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 111.111.11.1:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.11.1:5065;branch=z9hG4bK-7aeb-1e029a8-313b3450-10f8e80;received=111.111.11.1;rport=5065
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as0cf7cef9
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 2 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:815123456@111.111.11.1:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 2103517998 2103517998 IN IP4 111.111.11.1
s=Asterisk PBX 15.3.0
c=IN IP4 111.111.11.1
t=0 0
m=audio 10236 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:111.111.11.1:5065 --->
ACK sip:815123456@111.111.11.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as0cf7cef9
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 2 ACK
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-7aee-1e03225-7cc0c99-10f8e80
Max-Forwards: 70
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Authorization: Digest username="home_KL",realm="asterisk",nonce="1a1073e9",uri="sip:815123456@192.168.0.1:5060",response="040c54c89e98eb6c295cb6cd43b81dd8",algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
    -- Channel SIP/Tan-0000000b joined 'simple_bridge' basic-bridge <4842527a-6cf1-41e7-a3fa-f0232b4c91c9>
    -- Channel SIP/home_KL-0000000a joined 'simple_bridge' basic-bridge <4842527a-6cf1-41e7-a3fa-f0232b4c91c9>
Retransmitting #4 (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK2823eed7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as45eb32fc
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 6ebe468100228d4546986ab516b95910@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Thu, 06 Mar 2025 12:32:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '6ebe468100228d4546986ab516b95910@172.23.14.1:5060' Method: OPTIONS
[Mar  6 23:32:34] WARNING[1888][C-00000007]: chan_sip.c:7536 sip_write: Can't send 10 type frames with SIP write
[Mar  6 23:32:35] WARNING[1888][C-00000007]: chan_sip.c:7536 sip_write: Can't send 10 type frames with SIP write
[Mar  6 23:32:36] WARNING[1888][C-00000007]: chan_sip.c:7536 sip_write: Can't send 10 type frames with SIP write
[Mar  6 23:32:36] WARNING[1888][C-00000007]: chan_sip.c:7536 sip_write: Can't send 10 type frames with SIP write

<--- SIP read from UDP:111.111.11.1:5065 --->
BYE sip:815123456@111.111.11.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as0cf7cef9
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 3 BYE
Via: SIP/2.0/UDP 111.111.11.1:5065;rport;branch=z9hG4bK-7af1-1e03f54-2a988ac7-10f9040
Max-Forwards: 70
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Authorization: Digest username="home_KL",realm="asterisk",nonce="1a1073e9",uri="sip:815123456@111.111.11.1:5060",response="ce5de129e744a7cd8739ef645defa0ba",algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 111.111.11.1:5065 (no NAT)
Scheduling destruction of SIP dialog '116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 111.111.11.1:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.11.1:5065;branch=z9hG4bK-7af1-1e03f54-2a988ac7-10f9040;received=111.111.11.1;rport=5065
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=1132d80-74aa87d-13c9-D4180E-1d929ff0-54a96b18-1e0299e
To: "815123456"<sip:815123456@192.168.0.1:5060>;tag=as0cf7cef9
Call-ID: 116cc50-74aa87d-13c9-D4180E-6cee0ffa-17d6d9da-1e0299d
CSeq: 3 BYE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/home_KL-0000000a left 'simple_bridge' basic-bridge <4842527a-6cf1-41e7-a3fa-f0232b4c91c9>
  == Spawn extension (outgoing, 815123456, 2) exited non-zero on 'SIP/home_KL-0000000a'
    -- Channel SIP/Tan-0000000b left 'simple_bridge' basic-bridge <4842527a-6cf1-41e7-a3fa-f0232b4c91c9>
Scheduling destruction of SIP dialog '3db24116003041fc50f03d3f3327a5c8@my.isp.com' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 222.222.222.22:5060:
BYE sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK352de9d3;rport
Max-Forwards: 70
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-1039748949-1741264352982
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 104 BYE
User-Agent: Asterisk PBX 15.3.0
Authorization: Digest username="<my_number>", realm="my.isp.com", algorithm=MD5, uri="sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060", nonce="BroadWorksXm7xbu4r7Txqvrl4BW", response="6b00baf8ea1fbd69bef9379e0359f704", qop=auth, cnonce="360eb6d5", nc=00000002
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:222.222.222.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.11.1:5060;received=111.111.11.1;branch=z9hG4bK352de9d3;rport=5060
From: "home_KL" <sip:<my_number>@my.isp.com>;tag=as549af0f3
To: <sip:815123456@my.isp.com>;tag=SDpknv099-1039748949-1741264352982
Call-ID: 3db24116003041fc50f03d3f3327a5c8@my.isp.com
CSeq: 104 BYE
Content-Length: 0

PatTheRat · March 12, 2025, 8:00am

Log file for incoming call (SIP)-

---
Really destroying SIP dialog '6370eec7312fde8279b0fc5e2e281c86@172.23.14.1:5060' Method: OPTIONS
[Mar 12 14:39:10] NOTICE[16501]: chan_sip.c:15776 sip_reregister:    -- Re-registration for  Me@sip.provider2.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to xx.xx.xxx.xx:5060:
REGISTER sip:sip.provider2.com SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK581abaa0
Max-Forwards: 70
From: <sip:Me@sip.provider2.com>;tag=as3cea05d4
To: <sip:Me@sip.provider2.com>
Call-ID: 44c569db6688bc235439b71b273777f7@[xxxx.xxxx.xxxx::x]
CSeq: 590 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 15.3.0
Authorization: Digest username="Me", realm="sip.provider2.com", algorithm=MD5, uri="sip:sip.provider2.com", nonce="442858845", response="9014e1d0f549fdce1617e11d95568d78"
Expires: 120
Contact: <sip:s@111.111.11.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:xx.xx.xxx.xx:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK581abaa0
From: <sip:Me@sip.provider2.com>;tag=as3cea05d4
To: <sip:Me@sip.provider2.com>
Contact: sip:xx.xx.xxx.xx:5060
Call-ID: 44c569db6688bc235439b71b273777f7@[xxxx.xxxx.xxxx::x]
CSeq: 590 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip.provider2.com",nonce="224466152",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name sip.provider2.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to xx.xx.xxx.xx:5060:
REGISTER sip:sip.provider2.com SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK67b97da4
Max-Forwards: 70
From: <sip:Me@sip.provider2.com>;tag=as3cea05d4
To: <sip:Me@sip.provider2.com>
Call-ID: 44c569db6688bc235439b71b273777f7@[xxxx.xxxx.xxxx::x]
CSeq: 591 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 15.3.0
Authorization: Digest username="Me", realm="sip.provider2.com", algorithm=MD5, uri="sip:sip.provider2.com", nonce="224466152", response="e252dbe9c605b3f694d6fb13a6afa489"
Expires: 120
Contact: <sip:s@111.111.11.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:xx.xx.xxx.xx:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK67b97da4
From: <sip:Me@sip.provider2.com>;tag=as3cea05d4
To: <sip:Me@sip.provider2.com>
Contact: <sip:s@111.111.11.1:5060>;expires=120
Call-ID: 44c569db6688bc235439b71b273777f7@[xxxx.xxxx.xxxx::x]
CSeq: 591 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
[Mar 12 14:39:10] NOTICE[16501]: chan_sip.c:24663 handle_response_register: Outbound Registration: Expiry for sip.provider2.com is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '44c569db6688bc235439b71b273777f7@[xxxx.xxxx.xxxx::x]' Method: REGISTER
Reliably Transmitting (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK7f9575b4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as09d8aaa1
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 58a705b269431a9463600273354fbfcb@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Wed, 12 Mar 2025 03:39:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK7f9575b4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as09d8aaa1
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 58a705b269431a9463600273354fbfcb@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Wed, 12 Mar 2025 03:39:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK7f9575b4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as09d8aaa1
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 58a705b269431a9463600273354fbfcb@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Wed, 12 Mar 2025 03:39:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK7f9575b4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as09d8aaa1
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 58a705b269431a9463600273354fbfcb@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Wed, 12 Mar 2025 03:39:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK7f9575b4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as09d8aaa1
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 58a705b269431a9463600273354fbfcb@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Wed, 12 Mar 2025 03:39:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Mar 12 14:39:21] WARNING[16501]: chan_sip.c:3785 __sip_xmit: sip_xmit of 0xb610a0f8 (len 524) to 172.23.14.2:5060 returned -1: Operation not permitted
Really destroying SIP dialog '58a705b269431a9463600273354fbfcb@172.23.14.1:5060' Method: OPTIONS

<--- SIP read from UDP:222.222.222.22:5060 --->
INVITE sip:my_phone_number@111.111.11.1:5060 SIP/2.0
Via: SIP/2.0/UDP 222.222.222.22:5060;branch=z9hG4bK9av2cs304gk50cu4h5j0.1
From: <sip:incoming_phone_number@voice.my.isp.com;user=phone>;tag=SDicvld01-981400494-1741750764629-
To: ". Me"<sip:my_phone_number@voice.my.isp.com>
Call-ID: SDicvld01-e14333bcdc9b27bd7d3687310e0d8717-jm6gpa0040
CSeq: 70763051 INVITE
Contact: <sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Recv-Info: x-broadworks-client-session-info
Accept: application/media_control+xml,application/sdp,multipart/mixed
Supported: 100rel,timer
Min-SE: 90
Session-Expires: 1800;refresher=uas
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 369

v=0
o=BroadWorks 1662407013 1 IN IP4 222.222.222.22
s=-
c=IN IP4 222.222.222.22
b=RR:2000
b=RS:600
b=AS:128
t=0 0
m=audio 22880 RTP/AVP 9 8 0 18 96
b=RR:2000
b=RS:600
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
a=ptime:20
a=bsoft: 1 image udptl t38
<------------->
--- (16 headers 20 lines) ---
Sending to 222.222.222.22:5060 (no NAT)
Sending to 222.222.222.22:5060 (no NAT)
Using INVITE request as basis request - SDicvld01-e14333bcdc9b27bd7d3687310e0d8717-jm6gpa0040
Found peer 'Tan' for 'incoming_phone_number' from 222.222.222.22:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 96
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 222.222.222.22:22880
Looking for my_phone_number in Tan (domain 111.111.11.1)
sip_route_dump: route/path hop: <sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp>

<--- Transmitting (NAT) to 222.222.222.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 222.222.222.22:5060;branch=z9hG4bK9av2cs304gk50cu4h5j0.1;received=222.222.222.22;rport=5060
From: <sip:incoming_phone_number@voice.my.isp.com;user=phone>;tag=SDicvld01-981400494-1741750764629-
To: ". Me"<sip:my_phone_number@voice.my.isp.com>
Call-ID: SDicvld01-e14333bcdc9b27bd7d3687310e0d8717-jm6gpa0040
CSeq: 70763051 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:my_phone_number@111.111.11.1:5060>
Content-Length: 0


<------------>
    -- Executing [my_phone_number@Tan:1] Goto("SIP/Tan-0000000c", "phones,333,1") in new stack
    -- Goto (phones,333,1)
    -- Executing [333@phones:1] NoOp("SIP/Tan-0000000c", "incoming_phone_number") in new stack
    -- Executing [333@phones:2] GotoIf("SIP/Tan-0000000c", "1?allowed") in new stack
    -- Goto (phones,333,5)
    -- Executing [333@phones:5] Ringing("SIP/Tan-0000000c", "") in new stack

<--- Transmitting (NAT) to 222.222.222.22:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 222.222.222.22:5060;branch=z9hG4bK9av2cs304gk50cu4h5j0.1;received=222.222.222.22;rport=5060
From: <sip:incoming_phone_number@voice.my.isp.com;user=phone>;tag=SDicvld01-981400494-1741750764629-
To: ". Me"<sip:my_phone_number@voice.my.isp.com>;tag=as2e5e5c7f
Call-ID: SDicvld01-e14333bcdc9b27bd7d3687310e0d8717-jm6gpa0040
CSeq: 70763051 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:my_phone_number@111.111.11.1:5060>
Content-Length: 0


<------------>
    -- Executing [333@phones:6] Dial("SIP/Tan-0000000c", "SIP/home_KL") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 12578
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 111.111.11.1:5065:
INVITE sip:home_KL@111.111.11.1:5065 SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK173f3475
Max-Forwards: 70
From: <sip:incoming_phone_number@111.111.11.1>;tag=as0c25fbed
To: <sip:home_KL@111.111.11.1:5065>
Contact: <sip:incoming_phone_number@111.111.11.1:5060>
Call-ID: 736141401a0533bc1c072a3c48dd77c5@111.111.11.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.3.0
Date: Wed, 12 Mar 2025 03:39:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 1364597696 1364597696 IN IP4 111.111.11.1
s=Asterisk PBX 15.3.0
c=IN IP4 111.111.11.1
t=0 0
m=audio 12578 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/home_KL

<--- SIP read from UDP:111.111.11.1:5065 --->
SIP/2.0 180 Ringing
From: <sip:incoming_phone_number@111.111.11.1>;tag=as0c25fbed
To: <sip:home_KL@111.111.11.1:5065>;tag=b437c198-74aa87d-13c9-D4180E-1634cba5-19666c0e-18a105c
Call-ID: 736141401a0533bc1c072a3c48dd77c5@111.111.11.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK173f3475
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:home_KL@111.111.11.1:5065>
    -- SIP/home_KL-0000000d is ringing

<--- Transmitting (NAT) to 222.222.222.22:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 222.222.222.22:5060;branch=z9hG4bK9av2cs304gk50cu4h5j0.1;received=222.222.222.22;rport=5060
From: <sip:incoming_phone_number@voice.my.isp.com;user=phone>;tag=SDicvld01-981400494-1741750764629-
To: ". Me"<sip:my_phone_number@voice.my.isp.com>;tag=as2e5e5c7f
Call-ID: SDicvld01-e14333bcdc9b27bd7d3687310e0d8717-jm6gpa0040
CSeq: 70763051 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:my_phone_number@111.111.11.1:5060>
Content-Length: 0


<------------>
Reliably Transmitting (no NAT) to 111.111.11.1:5065:
OPTIONS sip:home_KL@111.111.11.1:5065 SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK03142291
Max-Forwards: 70
From: "asterisk" <sip:asterisk@111.111.11.1>;tag=as603af782
To: <sip:home_KL@111.111.11.1:5065>
Contact: <sip:asterisk@111.111.11.1:5060>
Call-ID: 199dede106ff0750478522592fa139bb@111.111.11.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Wed, 12 Mar 2025 03:39:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:111.111.11.1:5065 --->
SIP/2.0 486 Busy Here
From: "asterisk"<sip:asterisk@111.111.11.1>;tag=as603af782
To: <sip:home_KL@111.111.11.1:5065>;tag=b437c368-74aa87d-13c9-D4180E-4f81e975-64f52c14-18a1b9d
Call-ID: 199dede106ff0750478522592fa139bb@111.111.11.1:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK03142291
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS
X-Serialnumber: 55 23 56 89 56
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
Accept: application/sdp
Accept-Language: en
Accept-Encoding: 
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '199dede106ff0750478522592fa139bb@111.111.11.1:5060' Method: OPTIONS

<--- SIP read from UDP:111.111.11.1:5065 --->
SIP/2.0 200 OK
From: <sip:incoming_phone_number@111.111.11.1>;tag=as0c25fbed
To: <sip:home_KL@111.111.11.1:5065>;tag=b437c198-74aa87d-13c9-D4180E-1634cba5-19666c0e-18a105c
Call-ID: 736141401a0533bc1c072a3c48dd77c5@111.111.11.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK173f3475
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.11.1:5065>
Content-Type: application/sdp
Content-Length: 202

v=0
o=Box 256236562 256236562 IN IP4 111.111.11.1
s=-
c=IN IP4 111.111.11.1
t=0 0
m=audio 49916 RTP/AVP 8 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=maxptime:240
a=sendrecv
<------------->
--- (13 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 96
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 111.111.11.1:49916
sip_route_dump: route/path hop: <sip:home_KL@111.111.11.1:5065>
set_destination: Parsing <sip:home_KL@111.111.11.1:5065> for address/port to send to
set_destination: set destination to 111.111.11.1:5065
Transmitting (no NAT) to 111.111.11.1:5065:
ACK sip:home_KL@111.111.11.1:5065 SIP/2.0
Via: SIP/2.0/UDP 111.111.11.1:5060;branch=z9hG4bK6f8290ad
Max-Forwards: 70
From: <sip:incoming_phone_number@111.111.11.1>;tag=as0c25fbed
To: <sip:home_KL@111.111.11.1:5065>;tag=b437c198-74aa87d-13c9-D4180E-1634cba5-19666c0e-18a105c
Contact: <sip:incoming_phone_number@111.111.11.1:5060>
Call-ID: 736141401a0533bc1c072a3c48dd77c5@111.111.11.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.3.0
Content-Length: 0


---
    -- SIP/home_KL-0000000d answered SIP/Tan-0000000c
Audio is at 15142
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 222.222.222.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 222.222.222.22:5060;branch=z9hG4bK9av2cs304gk50cu4h5j0.1;received=222.222.222.22;rport=5060
From: <sip:incoming_phone_number@voice.my.isp.com;user=phone>;tag=SDicvld01-981400494-1741750764629-
To: ". Me"<sip:my_phone_number@voice.my.isp.com>;tag=as2e5e5c7f
Call-ID: SDicvld01-e14333bcdc9b27bd7d3687310e0d8717-jm6gpa0040
CSeq: 70763051 INVITE
Server: Asterisk PBX 15.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:my_phone_number@111.111.11.1:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 260

v=0
o=root 1226887283 1226887283 IN IP4 111.111.11.1
s=Asterisk PBX 15.3.0
c=IN IP4 111.111.11.1
t=0 0
m=audio 15142 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:222.222.222.22:5060 --->
ACK sip:my_phone_number@111.111.11.1:5060 SIP/2.0
Via: SIP/2.0/UDP 222.222.222.22:5060;branch=z9hG4bKn88jmv00804ufg1ceid0.1
From: <sip:incoming_phone_number@voice.my.isp.com;user=phone>;tag=SDicvld01-981400494-1741750764629-
To: ". Me"<sip:my_phone_number@voice.my.isp.com>;tag=as2e5e5c7f
Call-ID: SDicvld01-e14333bcdc9b27bd7d3687310e0d8717-jm6gpa0040
CSeq: 70763051 ACK
Contact: <sip:SDgt55b-vp9pm6n1vn1nrhs6gngof0drovpsfkkt000e420@222.222.222.22:5060;transport=udp>
Max-Forwards: 29
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- Channel SIP/home_KL-0000000d joined 'simple_bridge' basic-bridge <cf0a2ffc-6d85-4ee3-ad97-2d5078bc6298>
    -- Channel SIP/Tan-0000000c joined 'simple_bridge' basic-bridge <cf0a2ffc-6d85-4ee3-ad97-2d5078bc6298>
Reliably Transmitting (no NAT) to 172.23.14.2:5060:
OPTIONS sip:172.23.14.2 SIP/2.0
Via: SIP/2.0/UDP 172.23.14.1:5060;branch=z9hG4bK2d8168f1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.23.14.1>;tag=as2489d601
To: <sip:172.23.14.2>
Contact: <sip:asterisk@172.23.14.1:5060>
Call-ID: 2873ad2d499948b115523a5e06de6492@172.23.14.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.3.0
Date: Wed, 12 Mar 2025 03:39:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

PatTheRat · March 12, 2025, 8:01am

sip.conf file used -

[general]
context=public                  ; Default context for incoming calls. Defaults >
allowoverlap=no                 ; Disable overlap dialing support. (Default is >
udpbindaddr=0.0.0.0             ; IP address to bind UDP listen socket to (0.0.>
                                ; Optionally add a port number, 192.168.1.1:506>
tcpenable=no                    ; Enable server for incoming TCP connections (d>
tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0>
                                ; Optionally add a port number, 192.168.1.1:506>
transport=udp                   ; Set the default transports.  The order determ>
                                ; If tcpenable=no and the transport set is tcp,>
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                ; Note: Asterisk only uses the first host
                                ; in SRV records
                                ; Disabling DNS SRV lookups disables the
                                ; ability to place SIP calls based on domain
                                ; names to some other SIP users on the Internet
                                ; Specifying a port in a SIP peer definition or
                                ; when dialing outbound calls will supress SRV
                                ; lookups for that peer or call.
qualify=yes
;canreinvite=no

; # Sip Providers registration #
register => userid2:password2@sip.provider2.com
register => my_phone_number:password1@voice.my.isp.com


[authentication]
[basic-options](!)                ; a template
        dtmfmode=rfc2833
        context=from-office
        type=friend
[natted-phone](!,basic-options)   ; another template inheriting basic-options
        directmedia=no
        host=dynamic
[public-phone](!,basic-options)   ; another template inheriting basic-options
        directmedia=yes
[my-codecs](!)                    ; a template for my preferred codecs
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw
[ulaw-phone](!)                   ; and another one for ulaw-only
        disallow=all
        allow=ulaw

; # Phones #
[home_KL]                               ;your home telephone connection to aste>
        type=friend
        context=phones
        allow=ulaw,alaw
        secret=password_basic
        host=dynamic
[mobile-home]
        type=friend
        context=phones
        allow=ulaw,alaw
        secret=password_basic
        host=dynamic
; # Remote asterisk users #
[John]
        type=friend
        context=John_incoming           ;where to look in the extensions.conf f>
        allow=ulaw,alaw
        secret=
        host=172.23.14.2                ;wireguard tunnel, opposite end (johns end)

; # Sip Providers details #
[Tan]                                   ;inbound and outbound calls
        type=friend
        context=Tan
        allow=ulaw,alaw
        secret=password1
        host=voice.my.isp.com
        nat=force_rport,comedia
        insecure=invite
        fromdomain=voice.my.isp.com
        defaultuser=my_phone_number
[provider2]                              ;outbound calls only
        type=friend
        context=provider2
        allow=ulaw,alaw
        secret=password2
        host=sip.provider2.com
        nat=force_rport,comedia
        insecure=invite
        fromdomain=sip.provider2.com
        defaultuser=userid2

extensions.conf file used-

; # Incoming #
[Tan]
exten => my_phone_number,1,Goto(phones,333,1)

; # Internal Devices #
[phones]
include => internal
include => remote

[internal]                                  ;incoming calls
exten => _2,1,NoOp()                        ;if 2 comes in from outside
exten => _2,n,Dial(SIP/home_KL,30)          ;dial home phone
exten => _2,n,Hangup()                      ;hang up

exten => 111,1,Dial(SIP/home_KL)
same => n,Hangup

exten => 222,1,Dial(SIP/mobile-home)
same => n,Hangup

; # Block caller list #
exten => 333,1,NoOp(${CALLERID(num)})
same => n,GotoIf($["${DB(blockcaller/${CALLERID(num)})}" != "1"]?allowed)
same => n,Playback(Number_not_avaliable)
same => n,Hangup()
same => n(allowed),Ringing                          ;caller hears ringing tone
same => n,Dial(SIP/home_KL)
same => n,Hangup()

[John_incoming]
include => internal

; # Outgoing Dials #

[remote]                                    ;outgoing calls
exten => _555,1,NoOp()                      ;if 555 is dialled
exten => _555,n,Dial(SIP/John/${EXTEN})     ;go to sip.conf [John] where it con>
exten => _555,n,Hangup()                    ;hangup

; # Dial Extension provider2 #
exten => _2X.,1,Goto(outgoing,${EXTEN:1},1)          ;press 2 for provider2

; # Dial Extension Tan #
exten => _X.,1,NoOp(${CALLERID(num)})
same => n,Set(CALLERID(num)=my_phone_number)
same => n,Goto(outgoing,${EXTEN},2)

; # Outgoing Providers #
[outgoing]
exten => _X.,1,Dial(SIP/provider2/${EXTEN})
exten => _X.,2,Dial(SIP/TaTan/${EXTEN})

PatTheRat · March 12, 2025, 8:02am

Here are the files that come from my PJSIP setup which does not work for incoming or outgoing calls-

Log file for outgoing call (PJSIP) NOT working-

PJSIP Logging enabled
<--- Received SIP request (1076 bytes) from UDP:111.111.111.11:5065 --->
INVITE sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=b4369bf8-85bb56-13c9-83545D-36b15fee-13f3d1b-100bcd
To: "815123456"<sip:815123456@192.168.0.1:5060>
Call-ID: b5185a30-85bb56-13c9-83545D-b9270b5-1025aaa5-100bcd
CSeq: 1 INVITE
Via: SIP/2.0/UDP 111.111.111.11:5065;rport;branch=z9hG4bK-41b-100bcd-2c33a508-b4397b18
Max-Forwards: 70
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.111.11:5065>
Content-Type: application/sdp
Content-Length: 400

v=0
o=box 125485 45215 1256325 IN IP4 111.111.111.11
s=-
c=IN IP4 111.111.111.11
t=0 0
m=audio 40962 RTP/AVP 107 8 0 18 96 108
a=rtpmap:107 AMR-WB/16000
a=fmtp:107 channels=1;mode-change-capability=2;max-red=220
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtpmap:108 telephone-event/16000
a=fmtp:108 0-15
a=maxptime:240
a=silenceSupp:off - - - -
a=ptime:20

<--- Transmitting SIP response (600 bytes) to UDP:111.111.111.11:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.111.11:5065;rport=5065;received=111.111.111.11;branch=z9hG4bK-41b-100bcd-2c33a508-b4397b18
Call-ID: b5185a30-85bb56-13c9-83545D-b9270b5-1025aaa5-100bcd
From: "home_KL" <sip:home_KL@192.168.0.1>;tag=b4369bf8-85bb56-13c9-83545D-36b15fee-13f3d1b-100bcd
To: "815123456" <sip:815123456@192.168.0.1>;tag=z9hG4bK-41b-100bcd-2c33a508-b4397b18
CSeq: 1 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1741757659/78d9af875b944a149aceb9e6307704c1",opaque="2c2d418120f9aa12",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (645 bytes) from UDP:111.111.111.11:5065 --->
ACK sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=b4369bf8-85bb56-13c9-83545D-36b15fee-13f3d1b-100bcd
To: "815123456"<sip:815123456@192.168.0.1>;tag=z9hG4bK-41b-100bcd-2c33a508-b4397b18
Call-ID: b5185a30-85bb56-13c9-83545D-b9270b5-1025aaa5-100bcd
CSeq: 1 ACK
Via: SIP/2.0/UDP 111.111.111.11:5065;rport;branch=z9hG4bK-41b-100bcd-2c33a508-b4397b18
Max-Forwards: 70
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.111.11:5065>
Content-Length: 0


<--- Received SIP request (1344 bytes) from UDP:111.111.111.11:5065 --->
INVITE sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=b4369bf8-85bb56-13c9-83545D-36b15fee-13f3d1b-100bcd
To: "815123456"<sip:815123456@192.168.0.1:5060>
Call-ID: b5185a30-85bb56-13c9-83545D-b9270b5-1025aaa5-100bcd
CSeq: 2 INVITE
Via: SIP/2.0/UDP 111.111.111.11:5065;rport;branch=z9hG4bK-41b-100bd7-3eb2bb26-b4397cd8
Max-Forwards: 70
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.111.11:5065>
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741757659/78d9af875b944a149aceb9e6307704c1",uri="sip:815123456@192.168.0.1:5060",response="359d230e8b2d57d8f0fc411501a36ca4",algorithm=MD5,cnonce="100bd7",opaque="2c2d418120f9aa12",qop=auth,nc=00000001
Content-Type: application/sdp
Content-Length: 400

v=0
o=box 125485 45215 1256325 IN IP4 111.111.111.11
s=-
c=IN IP4 111.111.111.11
t=0 0
m=audio 40962 RTP/AVP 107 8 0 18 96 108
a=rtpmap:107 AMR-WB/16000
a=fmtp:107 channels=1;mode-change-capability=2;max-red=220
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtpmap:108 telephone-event/16000
a=fmtp:108 0-15
a=maxptime:240
a=silenceSupp:off - - - -
a=ptime:20

  == Setting global variable 'SIPDOMAIN' to '192.168.0.1'
<--- Transmitting SIP response (406 bytes) to UDP:111.111.111.11:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.111.11:5065;rport=5065;received=111.111.111.11;branch=z9hG4bK-41b-100bd7-3eb2bb26-b4397cd8
Call-ID: b5185a30-85bb56-13c9-83545D-b9270b5-1025aaa5-100bcd
From: "home_KL" <sip:home_KL@192.168.0.1>;tag=b4369bf8-85bb56-13c9-83545D-36b15fee-13f3d1b-100bcd
To: "815123456" <sip:815123456@192.168.0.1>
CSeq: 2 INVITE
Server: Asterisk PBX 15.3.0
Content-Length:  0


    -- Executing [815123456@Phones:1] NoOp("PJSIP/home_KL-00000002", "home_KL") in new stack
    -- Executing [815123456@Phones:2] Set("PJSIP/home_KL-00000002", "CALLERID(num)=my_phone_number") in new stack
    -- Executing [815123456@Phones:3] Goto("PJSIP/home_KL-00000002", "outgoing,815123456,2") in new stack
    -- Goto (outgoing,815123456,2)
    -- Executing [815123456@outgoing:2] Dial("PJSIP/home_KL-00000002", "PJSIP/815123456@Tan") in new stack
    -- Called PJSIP/815123456@Tan
<--- Transmitting SIP request (958 bytes) to UDP:222.222.222.22:5060 --->
INVITE sip:815123456@voice.my.isp.com SIP/2.0
Via: SIP/2.0/UDP 111.111.111.11:5060;rport;branch=z9hG4bKPj6d342c27-6acf-47b6-91f0-44308194688a
From: <sip:my_phone_number@voice.my.isp.com>;tag=3253c577-1528-49a1-8caa-1a7c5eb31aae
To: <sip:815123456@voice.my.isp.com>
Contact: <sip:my_phone_number@111.111.111.11:5060>
Call-ID: 0e055441-ba0d-4d2d-9be4-3dabfafb0652
CSeq: 20288 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 15.3.0
Content-Type: application/sdp
Content-Length:   261

v=0
o=- 1594399449 1594399449 IN IP4 111.111.111.11
s=Asterisk
c=IN IP4 111.111.111.11
t=0 0
m=audio 19232 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (346 bytes) from UDP:222.222.222.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.111.11:5060;received=111.111.111.11;branch=z9hG4bKPj6d342c27-6acf-47b6-91f0-44308194688a;rport=5060
From: <sip:my_phone_number@voice.my.isp.com>;tag=3253c577-1528-49a1-8caa-1a7c5eb31aae
To: <sip:815123456@voice.my.isp.com>
Call-ID: 0e055441-ba0d-4d2d-9be4-3dabfafb0652
CSeq: 20288 INVITE


<--- Received SIP response (377 bytes) from UDP:222.222.222.22:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 111.111.111.11:5060;received=111.111.111.11;branch=z9hG4bKPj6d342c27-6acf-47b6-91f0-44308194688a;rport=5060
From: <sip:my_phone_number@voice.my.isp.com>;tag=3253c577-1528-49a1-8caa-1a7c5eb31aae
To: <sip:815123456@voice.my.isp.com>;tag=aprqngfrt-oprhcs100oj04
Call-ID: 0e055441-ba0d-4d2d-9be4-3dabfafb0652
CSeq: 20288 INVITE


<--- Transmitting SIP request (446 bytes) to UDP:222.222.222.22:5060 --->
ACK sip:815123456@voice.my.isp.com SIP/2.0
Via: SIP/2.0/UDP 111.111.111.11:5060;rport;branch=z9hG4bKPj6d342c27-6acf-47b6-91f0-44308194688a
From: <sip:my_phone_number@voice.my.isp.com>;tag=3253c577-1528-49a1-8caa-1a7c5eb31aae
To: <sip:815123456@voice.my.isp.com>;tag=aprqngfrt-oprhcs100oj04
Call-ID: 0e055441-ba0d-4d2d-9be4-3dabfafb0652
CSeq: 20288 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 15.3.0
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/home_KL-00000002' status is 'CHANUNAVAIL'
<--- Transmitting SIP response (484 bytes) to UDP:111.111.111.11:5065 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 111.111.111.11:5065;rport=5065;received=111.111.111.11;branch=z9hG4bK-41b-100bd7-3eb2bb26-b4397cd8
Call-ID: b5185a30-85bb56-13c9-83545D-b9270b5-1025aaa5-100bcd
From: "home_KL" <sip:home_KL@192.168.0.1>;tag=b4369bf8-85bb56-13c9-83545D-36b15fee-13f3d1b-100bcd
To: "815123456" <sip:815123456@192.168.0.1>;tag=a2729bae-7969-47b9-aeae-81a8991361e3
CSeq: 2 INVITE
Server: Asterisk PBX 15.3.0
Reason: Q.850;cause=34
Content-Length:  0


<--- Received SIP request (645 bytes) from UDP:111.111.111.11:5065 --->
ACK sip:815123456@192.168.0.1:5060 SIP/2.0
From: "home_KL"<sip:home_KL@192.168.0.1>;tag=b4369bf8-85bb56-13c9-83545D-36b15fee-13f3d1b-100bcd
To: "815123456"<sip:815123456@192.168.0.1>;tag=a2729bae-7969-47b9-aeae-81a8991361e3
Call-ID: b5185a30-85bb56-13c9-83545D-b9270b5-1025aaa5-100bcd
CSeq: 2 ACK
Via: SIP/2.0/UDP 111.111.111.11:5065;rport;branch=z9hG4bK-41b-100bd7-3eb2bb26-b4397cd8
Max-Forwards: 70
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, UPDATE, OPTIONS, PRACK
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
X-Serialnumber: 55 23 56 89 56
Contact: <sip:home_KL@111.111.111.11:5065>
Content-Length: 0


<--- Received SIP request (807 bytes) from UDP:111.111.111.11:5065 --->
REGISTER sip:192.168.0.1 SIP/2.0
From: <sip:home_KL@192.168.0.1>;tag=b43692e8-85bb56-13c9-83545D-252388eb-1eb706d0-f7885
To: <sip:home_KL@192.168.0.1>
Call-ID: b51a1448-85bb56-13c9-83545D-5ae4dc95-5a62d30a-f7885
CSeq: 3 REGISTER
Via: SIP/2.0/UDP 111.111.111.11:5065;rport;branch=z9hG4bK-41c-100dda-2d668280-b43975d8
Max-Forwards: 70
Supported: replaces,100rel
X-Serialnumber: 55 23 56 89 56
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
Expires: 0
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741757622/acf71f7b17730780c9fd40742e634ce4",uri="sip:192.168.0.1:5060",response="7749721f0b13479ccf40735274e9e19c",algorithm=MD5,cnonce="100dda",opaque="56aaa11327369e3d",qop=auth,nc=00000002
Contact: <sip:home_KL@111.111.111.11:5065>
Content-Length: 0


<--- Transmitting SIP response (591 bytes) to UDP:111.111.111.11:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.111.11:5065;rport=5065;received=111.111.111.11;branch=z9hG4bK-41c-100dda-2d668280-b43975d8
Call-ID: b51a1448-85bb56-13c9-83545D-5ae4dc95-5a62d30a-f7885
From: <sip:home_KL@192.168.0.1>;tag=b43692e8-85bb56-13c9-83545D-252388eb-1eb706d0-f7885
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-41c-100dda-2d668280-b43975d8
CSeq: 3 REGISTER
WWW-Authenticate: Digest  realm="asterisk",nonce="1741757660/67e253d43d0269f0fe0fa76d21f077ea",opaque="297985e716ba2075",stale=true,algorithm=md5,qop="auth"
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (807 bytes) from UDP:111.111.111.11:5065 --->
REGISTER sip:192.168.0.1 SIP/2.0
From: <sip:home_KL@192.168.0.1>;tag=b43692e8-85bb56-13c9-83545D-252388eb-1eb706d0-f7885
To: <sip:home_KL@192.168.0.1>
Call-ID: b51a1448-85bb56-13c9-83545D-5ae4dc95-5a62d30a-f7885
CSeq: 4 REGISTER
Via: SIP/2.0/UDP 111.111.111.11:5065;rport;branch=z9hG4bK-41c-100de9-46c537a6-b4397e98
Max-Forwards: 70
Supported: replaces,100rel
X-Serialnumber: 55 23 56 89 56
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
Expires: 0
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741757660/67e253d43d0269f0fe0fa76d21f077ea",uri="sip:192.168.0.1:5060",response="5563eeff3c36bbf82692c9fed419bbb9",algorithm=MD5,cnonce="100de9",opaque="297985e716ba2075",qop=auth,nc=00000001
Contact: <sip:home_KL@111.111.111.11:5065>
Content-Length: 0


  == Contact home_KL/sip:home_KL@111.111.111.11:5065 has been deleted
    -- Removed contact 'sip:home_KL@111.111.111.11:5065' from AOR 'home_KL' due to request
<--- Transmitting SIP response (500 bytes) to UDP:111.111.111.11:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.111.11:5065;rport=5065;received=111.111.111.11;branch=z9hG4bK-41c-100de9-46c537a6-b4397e98
Call-ID: b51a1448-85bb56-13c9-83545D-5ae4dc95-5a62d30a-f7885
From: <sip:home_KL@192.168.0.1>;tag=b43692e8-85bb56-13c9-83545D-252388eb-1eb706d0-f7885
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-41c-100de9-46c537a6-b4397e98
CSeq: 4 REGISTER
Date: Wed, 12 Mar 2025 05:34:20 GMT
Contact: <sip:192.168.0.1>
Expires: 0
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (552 bytes) from UDP:111.111.111.11:5065 --->
REGISTER sip:192.168.0.1 SIP/2.0
From: <sip:home_KL@192.168.0.1>;tag=b436a168-85bb56-13c9-83545D-7cd8d04a-355070b3-100e0e
To: <sip:home_KL@192.168.0.1>
Call-ID: b51a15f0-85bb56-13c9-83545D-357d13f1-59fa54bd-100e0d
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 111.111.111.11:5065;rport;branch=z9hG4bK-41c-100e0e-14ceecc0-b4398058
Max-Forwards: 70
Supported: replaces,100rel
X-Serialnumber: 55 23 56 89 56
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
Expires: 600
Contact: <sip:home_KL@111.111.111.11:5065>
Content-Length: 0


<--- Transmitting SIP response (582 bytes) to UDP:111.111.111.11:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.111.11:5065;rport=5065;received=111.111.111.11;branch=z9hG4bK-41c-100e0e-14ceecc0-b4398058
Call-ID: b51a15f0-85bb56-13c9-83545D-357d13f1-59fa54bd-100e0d
From: <sip:home_KL@192.168.0.1>;tag=b436a168-85bb56-13c9-83545D-7cd8d04a-355070b3-100e0e
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-41c-100e0e-14ceecc0-b4398058
CSeq: 1 REGISTER
WWW-Authenticate: Digest  realm="asterisk",nonce="1741757660/67e253d43d0269f0fe0fa76d21f077ea",opaque="2d4f0d31203a9566",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (811 bytes) from UDP:111.111.111.11:5065 --->
REGISTER sip:192.168.0.1 SIP/2.0
From: <sip:home_KL@192.168.0.1>;tag=b436a168-85bb56-13c9-83545D-7cd8d04a-355070b3-100e0e
To: <sip:home_KL@192.168.0.1>
Call-ID: b51a15f0-85bb56-13c9-83545D-357d13f1-59fa54bd-100e0d
CSeq: 2 REGISTER
Via: SIP/2.0/UDP 111.111.111.11:5065;rport;branch=z9hG4bK-41c-100e26-77e6c65e-b4398218
Max-Forwards: 70
Supported: replaces,100rel
X-Serialnumber: 55 23 56 89 56
User-Agent: MyRouter 123456 Build 555555 5555555 55555 -- 55 23 56 89 56
Expires: 600
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741757660/67e253d43d0269f0fe0fa76d21f077ea",uri="sip:192.168.0.1:5060",response="a8ded1840c8adb1d49a4d8dec9d65b50",algorithm=MD5,cnonce="100e26",opaque="2d4f0d31203a9566",qop=auth,nc=00000001
Contact: <sip:home_KL@111.111.111.11:5065>
Content-Length: 0


    -- Added contact 'sip:home_KL@111.111.111.11:5065' to AOR 'home_KL' with expiration of 600 seconds
<--- Transmitting SIP response (558 bytes) to UDP:111.111.111.11:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.111.11:5065;rport=5065;received=111.111.111.11;branch=z9hG4bK-41c-100e26-77e6c65e-b4398218
Call-ID: b51a15f0-85bb56-13c9-83545D-357d13f1-59fa54bd-100e0d
From: <sip:home_KL@192.168.0.1>;tag=b436a168-85bb56-13c9-83545D-7cd8d04a-355070b3-100e0e
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-41c-100e26-77e6c65e-b4398218
CSeq: 2 REGISTER
Date: Wed, 12 Mar 2025 05:34:20 GMT
Contact: <sip:home_KL@111.111.111.11:5065>;expires=599
Contact: <sip:192.168.0.1>
Expires: 600
Server: Asterisk PBX 15.3.0
Content-Length:  0


  == Contact home_KL/sip:home_KL@111.111.111.11:5065 has been created
    -- Contact home_KL/sip:home_KL@111.111.111.11:5065 is now Unknown.  RTT: 0.000 msec
<--- Received SIP request (496 bytes) from UDP:111.111.111.11:5065 --->
SUBSCRIBE sip:home_KL@192.168.0.1:5060 SIP/2.0
From: <sip:home_KL@192.168.0.1:5060>;tag=b436a6d8-85bb56-13c9-83545D-81c6ff3-639a173d-10124f
To: <sip:home_KL@192.168.0.1:5060>
Call-ID: b5185c78-85bb56-13c9-83545D-2638251-4aa69daf-10124f
CSeq: 1 SUBSCRIBE
Via: SIP/2.0/UDP 111.111.111.11:5065;rport;branch=z9hG4bK-41d-10124f-72c1ba4b-b4398598
Expires: 86400
Event: message-summary
Max-Forwards: 70
Supported: replaces,100rel
Contact: <sip:home_KL@111.111.111.11:5065>
Content-Length: 0


<--- Transmitting SIP response (581 bytes) to UDP:111.111.111.11:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.111.11:5065;rport=5065;received=111.111.111.11;branch=z9hG4bK-41d-10124f-72c1ba4b-b4398598
Call-ID: b5185c78-85bb56-13c9-83545D-2638251-4aa69daf-10124f
From: <sip:home_KL@192.168.0.1>;tag=b436a6d8-85bb56-13c9-83545D-81c6ff3-639a173d-10124f
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-41d-10124f-72c1ba4b-b4398598
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest  realm="asterisk",nonce="1741757661/e8e3c840ebfb030be4e58916f85ca94b",opaque="787990e40f8ae5e7",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (763 bytes) from UDP:111.111.111.11:5065 --->
SUBSCRIBE sip:home_KL@192.168.0.1:5060 SIP/2.0
From: <sip:home_KL@192.168.0.1:5060>;tag=b436a6d8-85bb56-13c9-83545D-81c6ff3-639a173d-10124f
To: <sip:home_KL@192.168.0.1:5060>
Call-ID: b5185c78-85bb56-13c9-83545D-2638251-4aa69daf-10124f
CSeq: 2 SUBSCRIBE
Via: SIP/2.0/UDP 111.111.111.11:5065;rport;branch=z9hG4bK-41d-101255-2b43a121-b4398758
Expires: 86400
Event: message-summary
Max-Forwards: 70
Supported: replaces,100rel
Contact: <sip:home_KL@111.111.111.11:5065>
Authorization: Digest username="home_KL",realm="asterisk",nonce="1741757661/e8e3c840ebfb030be4e58916f85ca94b",uri="sip:home_KL@192.168.0.1:5060",response="8d1b2e8496156dad8c9d1e9aa004e7c2",algorithm=MD5,cnonce="101255",opaque="787990e40f8ae5e7",qop=auth,nc=00000001
Content-Length: 0


<--- Transmitting SIP response (431 bytes) to UDP:111.111.111.11:5065 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 111.111.111.11:5065;rport=5065;received=111.111.111.11;branch=z9hG4bK-41d-101255-2b43a121-b4398758
Call-ID: b5185c78-85bb56-13c9-83545D-2638251-4aa69daf-10124f
From: <sip:home_KL@192.168.0.1>;tag=b436a6d8-85bb56-13c9-83545D-81c6ff3-639a173d-10124f
To: <sip:home_KL@192.168.0.1>;tag=z9hG4bK-41d-101255-2b43a121-b4398758
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 15.3.0
Content-Length:  0

When an incoming call is made the CLI shows nothing and the caller hears nothing or a busy signal.

Here is what ‘pjsip show endpoints’ shows-

==========================================================================================

 Endpoint:  provider2                                                Not in use    0 of inf
    OutAuth:  provider2/userid2
        Aor:  provider2                                              1
      Contact:  provider2/sip:sip.provider2.com                    79a14c71d3 Unknown         nan

 Endpoint:  Tan                                            Not in use    0 of inf
    OutAuth:  Tan/my_phone_number
     InAuth:  Tan/my_phone_number
        Aor:  Tan                                          1
      Contact:  Tan/sip:voice.my.isp.com     671c5c1a67 Unknown         nan

 Endpoint:  home_KL                                              Not in use    0 of inf
     InAuth:  home_KL/home_KL
        Aor:  home_KL                                            1
      Contact:  home_KL/sip:home_KL@111.111.11.1:5065      c1ef3f0c17 Unknown         nan
      Contact:  home_KL/sip:192.168.0.1                    ebb1defd1b Unknown         nan

PatTheRat · March 12, 2025, 8:02am

pjsip.conf file used-

[global]
type=global

; Define the transport
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

; #### Phones ####

; ## home_KL (Phone) ##
[home_KL]
type=endpoint
context=Phones
disallow=all
allow=ulaw,alaw
auth=home_KL
aors=home_KL
direct_media=yes

[home_KL]
type=auth
auth_type=userpass
username=home_KL
password=password_basic

[home_KL]
type=aor
max_contacts=1
contact=sip:192.168.0.1


; ### VOIP Providers ###

; ## Tan ##
[Tan]                                  ;inbound and outbound calls
type=endpoint
context=Tan
disallow=all
allow=ulaw,alaw
outbound_auth=Tan
auth=Tan
aors=Tan
direct_media=yes
from_user=my_phone_number
from_domain=voice.my.isp.com

[Tan]
type=auth
auth_type=userpass
username=my_phone_number
password=password1

[Tan]
type=aor
max_contacts=1
contact=sip:voice.my.isp.com

; ## provider2 ##
[provider2]                          ;outbound calls only
type=endpoint
context=provider2
allow=ulaw,alaw
outbound_auth=provider2
aors=provider2
direct_media=yes

[provider2]
type=auth
auth_type=userpass
username=userid2
password=password2

[provider2]
type=aor
max_contacts=1
contact=sip:sip.provider2.com

extension.conf file used-

; ## Incoming calls ##
[Tan]
exten => my_phone_number,1,Goto(phones,111,1)

; ## Outgoing dials ##

[Phones]

exten => 111,1,Dial(PJSIP/home_KL)
same => n,Hangup

exten => 222,1,Dial(PJSIP/mobile-home)
same => n,Hangup

; # Dial Extension provider2 #
exten => _2X.,1,Goto(outgoing,${EXTEN:1},1)          ;press 2 for provider2

; # Dial Extension Tan #
exten => _X.,1,NoOp(${CALLERID(num)})
same => n,Set(CALLERID(num)=my_phone_number)
same => n,Goto(outgoing,${EXTEN},2)

; ## Outgoing calls ##

; # Outgoing Providers #
[outgoing]
exten => _X.,1,Dial(PJSIP/${EXTEN}@provider2)
exten => _X.,2,Dial(PJSIP/${EXTEN}@Tan)

I believe the pjsip is not registering with the provider properly.
Any help would be appreciated.

jcolp · March 12, 2025, 8:52am

You didn’t show the outbound registration. One thing that can be different (I have no idea about your provider’s deployment) is that chan_pjsip obeys load balancing SRV/NAPTR, meaning an outbound registration can go to one IP address and calls to another. Maybe they don’t like that.

PatTheRat · March 12, 2025, 12:12pm

So I guess I will try adding something like this-

[my_provider]
type = registration
server_uri = sip:registrar@example.com
client_uri = sip:client@example.com
contact_user = inbound-calls

so mine would look like-

[Tan]
type = registration
server_uri = sip:my_phone_number@voice.my.isp.com
client_uri = sip:my_phone_number@voice.my.isp.com
contact_user = Tan

Am not sure if the server_uri and client_uri are correct or the contact_user for that matter.

Also should I add, is this necessary?-

[my_provider_identify]
type = identify
match = <ip address of provider>
endpoint = my_provider

and would mine look like-

[Tan]
type = identify
match = 222.222.222.22
endpoint = Tan

Thanks.

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