Asterisk outgoing calls convert from VoIP to GSM

I’m using Asterisk to make multiple outgoing calls to clients, but I believe some of them have blocked VoIP codecs, so my calls don’t go through. is that even possible? Is there a way to convert an outgoing call codec from VoIP to GSM, without the need to buy one of these VoIP getaways?

I rather suspect they are blocking on the number range.

1 Like

i thought that too at first, and that was correct after few tests and changing the caller id, but there are two companies that I’m trying to call using a wide of different kinds of caller id and always getting a busy response in asterisk CLI, but when a call is made from an ordinary phone with a single sim card the call goes normal !! I’m really so curious how they achieved this kind of solution

just so you know, i have used 5 diffrent sip providers, and more then 50 caller id. and always getting the busy response. The call doesnt even start at all just beep in my headset determine that the call cuts instantly and displaying busy response in my CLI

Network operators will get get unforgeable accounting number indications and/or an indication as to whether the first network operator in the chain has verified the number.

What is in the actual SIP final response packet?

1 Like

I will make a test and provide a screenshot in few minutes

The actual text, from the log file, is generally better than a screen shot.

please can you be more brief about that answer

The decision might be based on ANI or a not screened/failed screen status.

this is the log text:

Running as user ‘asterisk’
Running under group ‘asterisk’
Connected to Asterisk 16.12.0 currently running on li1481-56 (pid = 1010)
== Using SIP RTP CoS mark 5
– Executing [4915735852785@External-calls:1] NoOp(“SIP/medeva222990-00000000”, “### German Destination ## EXTERNAL CALL ###”) in new stack
– Executing [4915735852785@External-calls:2] Set(“SIP/medeva222990-00000000”, “CALLERID(num)=491743320982”) in new stack
– Executing [4915735852785@External-calls:3] Dial(“SIP/medeva222990-00000000”, “SIP/4915735852785@World”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/4915735852785@World
– Got SIP response 486 “Busy Here” back from 87.238.100.40:5060
– SIP/World-00000001 is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Executing [4915735852785@External-calls:4] Answer(“SIP/medeva222990-00000000”, “”) in new stack
– Executing [4915735852785@External-calls:5] Hangup(“SIP/medeva222990-00000000”, “”) in new stack
== Spawn extension (External-calls, 4915735852785, 5) exited non-zero on ‘SIP/medeva222990-00000000’
li1481-56*CLI>

That’s a simple busy, so doesn’t really add more information.

For future information, what is really wanted here is the protocol debugging (“sip set debug on”, in this case.

Also, if this is a new system, chan_sip is no longer fully supported.

1 Like

You can always use Allow and Disallow, to limit the codec to gsm, but I don’t think that will help, and your service provider might no even support it.

– Executing [4915735852785@External-calls:1] NoOp(“SIP/medeva222990-00000004”, “### German Destination ## EXTERNAL CALL ###”) in new stack
Really destroying SIP dialog ‘1167415691’ Method: REGISTER
Really destroying SIP dialog ‘2582074489’ Method: REGISTER
Really destroying SIP dialog ‘110065892’ Method: REGISTER
Really destroying SIP dialog ‘2251560955’ Method: REGISTER
Really destroying SIP dialog ‘618724328’ Method: REGISTER
Really destroying SIP dialog ‘2615571913’ Method: REGISTER
Really destroying SIP dialog ‘2844084401’ Method: REGISTER
Really destroying SIP dialog ‘561179678’ Method: REGISTER
Really destroying SIP dialog ‘1740159921’ Method: REGISTER
Really destroying SIP dialog ‘873254018’ Method: REGISTER
– Executing [4915735852785@External-calls:2] Set(“SIP/medeva222990-00000004”, “CALLERID(num)=491768179260”) in new stack
Really destroying SIP dialog ‘1072580459’ Method: REGISTER
Really destroying SIP dialog ‘1916530023’ Method: REGISTER
Really destroying SIP dialog ‘2019744864’ Method: REGISTER
– Executing [4915735852785@External-calls:3] Dial(“SIP/medeva222990-00000004”, “SIP/4915735852785@World”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 15068
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec gsm to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 82.215.255.13:5060:
INVITE sip:4915735852785@sip.worldcall.be SIP/2.0
Via: SIP/2.0/UDP 139.162.223.88:5060;branch=z9hG4bK2955d51f;rport
Max-Forwards: 70
From: sip:491768179260@139.162.223.88;tag=as5d92ade0
To: sip:4915735852785@sip.worldcall.be
Contact: sip:491768179260@139.162.223.88:5060
Call-ID: 73b08e1b34d79604278ce3a1302d646c@139.162.223.88:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.12.0
Date: Thu, 04 Feb 2021 11:22:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 441

v=0
o=root 1112656055 1112656055 IN IP4 139.162.223.88
s=Asterisk PBX 16.12.0
c=IN IP4 139.162.223.88
t=0 0
m=audio 15068 RTP/AVP 8 0 3 4 111 18 110 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:60
a=sendrecv


Really destroying SIP dialog ‘3073140332’ Method: REGISTER
– Called SIP/4915735852785@World

<— SIP read from UDP:82.215.255.13:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 139.162.223.88:5060;branch=z9hG4bK2955d51f;rport=5060
Record-Route: sip:82.215.255.13;lr;ep
To: sip:4915735852785@sip.worldcall.be;tag=CCU3KCRYXE75ZPI3FGXA____.i
From: sip:491768179260@139.162.223.88;tag=as5d92ade0
Call-ID: 73b08e1b34d79604278ce3a1302d646c@139.162.223.88:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm=“sip.worldcall.be”,nonce=“1612437763:e58f66d59a83a6b13730102bbc2bc6a4cdd7c697”
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Transmitting (NAT) to 82.215.255.13:5060:
ACK sip:4915735852785@sip.worldcall.be SIP/2.0
Via: SIP/2.0/UDP 139.162.223.88:5060;branch=z9hG4bK2955d51f;rport
Max-Forwards: 70
From: sip:491768179260@139.162.223.88;tag=as5d92ade0
To: sip:4915735852785@sip.worldcall.be;tag=CCU3KCRYXE75ZPI3FGXA____.i
Contact: sip:491768179260@139.162.223.88:5060
Call-ID: 73b08e1b34d79604278ce3a1302d646c@139.162.223.88:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.12.0
Content-Length: 0


Audio is at 15068
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec gsm to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 82.215.255.13:5060:
INVITE sip:4915735852785@sip.worldcall.be SIP/2.0
Via: SIP/2.0/UDP 139.162.223.88:5060;branch=z9hG4bK0eb2f611;rport
Max-Forwards: 70
From: sip:491768179260@139.162.223.88;tag=as5d92ade0
To: sip:4915735852785@sip.worldcall.be
Contact: sip:491768179260@139.162.223.88:5060
Call-ID: 73b08e1b34d79604278ce3a1302d646c@139.162.223.88:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 16.12.0
Authorization: Digest username=“medevamark”, realm=“sip.worldcall.be”, algorithm=MD5, uri=“sip:4915735852785@sip.worldcall.be”, nonce=“1612437763:e58f66d59a83a6b13730102bbc2bc6a4cdd7c697”, response=“c8c817007bd00bbfe9e72b0a70abaa62”
Date: Thu, 04 Feb 2021 11:22:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 441

v=0
o=root 1112656055 1112656056 IN IP4 139.162.223.88
s=Asterisk PBX 16.12.0
c=IN IP4 139.162.223.88
t=0 0
m=audio 15068 RTP/AVP 8 0 3 4 111 18 110 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:60
a=sendrecv


Retransmitting #1 (NAT) to 82.215.255.13:5060:
INVITE sip:4915735852785@sip.worldcall.be SIP/2.0
Via: SIP/2.0/UDP 139.162.223.88:5060;branch=z9hG4bK0eb2f611;rport
Max-Forwards: 70
From: sip:491768179260@139.162.223.88;tag=as5d92ade0
To: sip:4915735852785@sip.worldcall.be
Contact: sip:491768179260@139.162.223.88:5060
Call-ID: 73b08e1b34d79604278ce3a1302d646c@139.162.223.88:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 16.12.0
Authorization: Digest username=“medevamark”, realm=“sip.worldcall.be”, algorithm=MD5, uri=“sip:4915735852785@sip.worldcall.be”, nonce=“1612437763:e58f66d59a83a6b13730102bbc2bc6a4cdd7c697”, response=“c8c817007bd00bbfe9e72b0a70abaa62”
Date: Thu, 04 Feb 2021 11:22:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 441

v=0
o=root 1112656055 1112656056 IN IP4 139.162.223.88
s=Asterisk PBX 16.12.0
c=IN IP4 139.162.223.88
t=0 0
m=audio 15068 RTP/AVP 8 0 3 4 111 18 110 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:60
a=sendrecv


<— SIP read from UDP:82.215.255.13:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 139.162.223.88:5060;branch=z9hG4bK0eb2f611;rport=5060
To: sip:4915735852785@sip.worldcall.be
From: sip:491768179260@139.162.223.88;tag=as5d92ade0
Call-ID: 73b08e1b34d79604278ce3a1302d646c@139.162.223.88:5060
CSeq: 103 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘808974254’ Method: REGISTER

<— SIP read from UDP:82.215.255.13:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 139.162.223.88:5060;branch=z9hG4bK0eb2f611;rport=5060
To: sip:4915735852785@sip.worldcall.be
From: sip:491768179260@139.162.223.88;tag=as5d92ade0
Call-ID: 73b08e1b34d79604278ce3a1302d646c@139.162.223.88:5060
CSeq: 103 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘1897526124’ Method: REGISTER
Really destroying SIP dialog ‘3553755349’ Method: REGISTER
Really destroying SIP dialog ‘295538090’ Method: REGISTER
Really destroying SIP dialog ‘2296165687’ Method: REGISTER
Really destroying SIP dialog ‘3869879304’ Method: REGISTER
Really destroying SIP dialog ‘98723703’ Method: REGISTER
Really destroying SIP dialog ‘3336342408’ Method: REGISTER
Really destroying SIP dialog ‘332462016’ Method: REGISTER
Really destroying SIP dialog ‘2903042514’ Method: REGISTER
Really destroying SIP dialog ‘1271735885’ Method: REGISTER
Really destroying SIP dialog ‘4263156869’ Method: REGISTER
Really destroying SIP dialog ‘862011826’ Method: REGISTER
Really destroying SIP dialog ‘1241212981’ Method: REGISTER
Really destroying SIP dialog ‘3447381492’ Method: REGISTER
Retransmitting #4 (NAT) to 185.16.38.109:63911:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.16.38.109:63911;branch=z9hG4bK1970516693;received=185.16.38.109;rport=63911
From: sip:87222@139.162.223.88;tag=907230795
To: sip:7006436685423@139.162.223.88;tag=as4c779955
Call-ID: 2010186050-831448554-619433317
CSeq: 1 INVITE
Server: Asterisk PBX 16.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“46a4692a”
Content-Length: 0


Really destroying SIP dialog ‘1339370671’ Method: REGISTER
Really destroying SIP dialog ‘1799033934’ Method: REGISTER
Really destroying SIP dialog ‘329206218’ Method: REGISTER
Really destroying SIP dialog ‘1472114825’ Method: REGISTER
Really destroying SIP dialog ‘3609602057’ Method: REGISTER
Really destroying SIP dialog ‘2852314279’ Method: REGISTER
Really destroying SIP dialog ‘1215897344’ Method: REGISTER
Really destroying SIP dialog ‘969166037’ Method: REGISTER
Retransmitting #3 (NAT) to 185.16.38.107:55241:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.16.38.107:55241;branch=z9hG4bK1019709973;received=185.16.38.107;rport=55241
From: sip:12284@139.162.223.88;tag=1043430879
To: sip:00048221530543@139.162.223.88;tag=as6bf43c63
Call-ID: 1759806509-401364634-491428581
CSeq: 1 INVITE
Server: Asterisk PBX 16.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“2afafab9”
Content-Length: 0


Really destroying SIP dialog ‘1993692684’ Method: REGISTER
Really destroying SIP dialog ‘347231999’ Method: REGISTER
Really destroying SIP dialog ‘3797850834’ Method: REGISTER
Really destroying SIP dialog ‘368680959’ Method: REGISTER
Really destroying SIP dialog ‘2632288365’ Method: REGISTER
Really destroying SIP dialog ‘2143498068’ Method: REGISTER
Really destroying SIP dialog ‘4013944795’ Method: REGISTER
Really destroying SIP dialog ‘3488289165’ Method: REGISTER
Really destroying SIP dialog ‘4125147088’ Method: REGISTER
Retransmitting #5 (NAT) to 193.29.14.123:54536:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 193.29.14.123:54536;branch=z9hG4bK305179158;received=193.29.14.123;rport=54536
From: sip:2001@139.162.223.88;tag=758047494
To: sip:95237441536737166@139.162.223.88;tag=as58ededec
Call-ID: 1512820313-944803134-1089283473
CSeq: 1 INVITE
Server: Asterisk PBX 16.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“1641a165”
Content-Length: 0


Really destroying SIP dialog ‘3262581129’ Method: REGISTER
Really destroying SIP dialog ‘3704450970’ Method: REGISTER
Really destroying SIP dialog ‘3207470467’ Method: REGISTER
Really destroying SIP dialog ‘2021972172’ Method: REGISTER
Really destroying SIP dialog ‘3841105290’ Method: REGISTER
Really destroying SIP dialog ‘2865936831’ Method: REGISTER
Really destroying SIP dialog ‘3177077366’ Method: REGISTER
Really destroying SIP dialog ‘2864399545’ Method: REGISTER

<— SIP read from UDP:82.215.255.13:5060 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 139.162.223.88:5060;branch=z9hG4bK0eb2f611;rport=5060
Record-Route: sip:82.215.255.13;lr;ep
To: sip:4915735852785@sip.worldcall.be;tag=CCU3KCRYXE75ZPI3T2JQ____.i
From: sip:491768179260@139.162.223.88;tag=as5d92ade0
Call-ID: 73b08e1b34d79604278ce3a1302d646c@139.162.223.88:5060
CSeq: 103 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Server: PortaSIP
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– Got SIP response 486 “Busy Here” back from 82.215.255.13:5060
Transmitting (NAT) to 82.215.255.13:5060:
ACK sip:4915735852785@sip.worldcall.be SIP/2.0
Via: SIP/2.0/UDP 139.162.223.88:5060;branch=z9hG4bK0eb2f611;rport
Max-Forwards: 70
From: sip:491768179260@139.162.223.88;tag=as5d92ade0
To: sip:4915735852785@sip.worldcall.be;tag=CCU3KCRYXE75ZPI3T2JQ____.i
Contact: sip:491768179260@139.162.223.88:5060
Call-ID: 73b08e1b34d79604278ce3a1302d646c@139.162.223.88:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 16.12.0
Content-Length: 0


-- SIP/World-00000005 is busy

== Everyone is busy/congested at this time (1:1/0/0)
– Executing [4915735852785@External-calls:4] Answer(“SIP/medeva222990-00000004”, “”) in new stack
Audio is at 15344
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 196.206.121.13:53468 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 196.206.121.13:53468;branch=z9hG4bKPjdc361ca92602419086dd8a1348208674;received=196.206.121.13;rport=53468
From: sip:medeva222990@139.162.223.88;tag=86c462c0a69f4de2bee71cefe998f67e
To: sip:4915735852785@139.162.223.88;tag=as24714a73
Call-ID: 075b4b68decc40b38308017c930e4686
CSeq: 1746 INVITE
Server: Asterisk PBX 16.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:4915735852785@139.162.223.88:5060
Content-Type: application/sdp
Require: timer
Content-Length: 291

v=0
o=root 1113571421 1113571421 IN IP4 139.162.223.88
s=Asterisk PBX 16.12.0
c=IN IP4 139.162.223.88
t=0 0
m=audio 15344 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<------------>
– Executing [4915735852785@External-calls:5] Hangup(“SIP/medeva222990-00000004”, “”) in new stack
== Spawn extension (External-calls, 4915735852785, 5) exited non-zero on ‘SIP/medeva222990-00000004’
Scheduling destruction of SIP dialog ‘075b4b68decc40b38308017c930e4686’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:196.206.121.13:53468 —>
ACK sip:4915735852785@139.162.223.88:5060 SIP/2.0
Via: SIP/2.0/UDP 196.206.121.13:53468;rport;branch=z9hG4bKPj0a43faf1cf994580b90be8d7caf4db15
Max-Forwards: 70
From: sip:medeva222990@139.162.223.88;tag=86c462c0a69f4de2bee71cefe998f67e
To: sip:4915735852785@139.162.223.88;tag=as24714a73
Call-ID: 075b4b68decc40b38308017c930e4686
CSeq: 1746 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Reliably Transmitting (NAT) to 196.206.121.13:53468:
BYE sip:medeva222990@196.206.121.13:53468;ob SIP/2.0
Via: SIP/2.0/UDP 139.162.223.88:5060;branch=z9hG4bK6171f6d3;rport
Max-Forwards: 70
From: sip:4915735852785@139.162.223.88;tag=as24714a73
To: sip:medeva222990@139.162.223.88;tag=86c462c0a69f4de2bee71cefe998f67e
Call-ID: 075b4b68decc40b38308017c930e4686
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.12.0
X-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
Content-Length: 0


Scheduling destruction of SIP dialog ‘075b4b68decc40b38308017c930e4686’ in 6400 ms (Method: ACK)
Really destroying SIP dialog ‘73b08e1b34d79604278ce3a1302d646c@139.162.223.88:5060’ Method: INVITE
Really destroying SIP dialog ‘92602479’ Method: REGISTER

<— SIP read from UDP:196.206.121.13:53468 —>
INVITE sip:4915735852785@139.162.223.88:5060 SIP/2.0
Via: SIP/2.0/UDP 196.206.121.13:53468;rport;branch=z9hG4bKPj391d837edbfd4e25ba0d419eb483f166
Max-Forwards: 70
From: sip:medeva222990@139.162.223.88;tag=86c462c0a69f4de2bee71cefe998f67e
To: sip:4915735852785@139.162.223.88;tag=as24714a73
Contact: sip:medeva222990@196.206.121.13:53468;ob
Call-ID: 075b4b68decc40b38308017c930e4686
CSeq: 1747 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 319

v=0
o=- 3821430163 3821430164 IN IP4 196.206.121.13
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4014 RTP/AVP 8 101
c=IN IP4 196.206.121.13
b=TIAS:64000
a=rtcp:4015 IN IP4 192.168.1.9
a=ssrc:1709190253 cname:594a12183f4059c0
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
— (14 headers 15 lines) —
Sending to 196.206.121.13:53468 (NAT)
Using INVITE request as basis request - 075b4b68decc40b38308017c930e4686
Really destroying SIP dialog ‘669263526’ Method: REGISTER
Really destroying SIP dialog ‘2505109900’ Method: REGISTER

<— SIP read from UDP:196.206.121.13:53468 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 139.162.223.88:5060;rport=5060;received=139.162.223.88;branch=z9hG4bK6171f6d3
Call-ID: 075b4b68decc40b38308017c930e4686
From: sip:4915735852785@139.162.223.88;tag=as24714a73
To: sip:medeva222990@139.162.223.88;tag=86c462c0a69f4de2bee71cefe998f67e
CSeq: 102 BYE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘075b4b68decc40b38308017c930e4686’ Method: INVITE
Really destroying SIP dialog ‘3139021594’ Method: REGISTER
Really destroying SIP dialog ‘3037468440’ Method: REGISTER
Really destroying SIP dialog ‘3808302571’ Method: REGISTER
Really destroying SIP dialog ‘1153929288’ Method: REGISTER
Really destroying SIP dialog ‘2589002417’ Method: REGISTER
Retransmitting #4 (NAT) to 193.29.14.124:50353:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 193.29.14.124:50353;branch=z9hG4bK1637576458;received=193.29.14.124;rport=50353
From: sip:83314@139.162.223.88;tag=742218712
To: sip:700441964598137@139.162.223.88;tag=as38bc0bd4
Call-ID: 401893095-827736605-1513836897
CSeq: 1 INVITE
Server: Asterisk PBX 16.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“21e186a3”
Content-Length: 0


Really destroying SIP dialog ‘3272327430’ Method: REGISTER
Really destroying SIP dialog ‘54793420’ Method: REGISTER
Reliably Transmitting (NAT) to 196.206.121.13:53468:
OPTIONS sip:medeva222990@196.206.121.13:53468;ob SIP/2.0
Via: SIP/2.0/UDP 139.162.223.88:5060;branch=z9hG4bK5d58a9cb;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@139.162.223.88;tag=as783f3c8c
To: sip:medeva222990@196.206.121.13:53468;ob
Contact: sip:asterisk@139.162.223.88:5060
Call-ID: 445483b5345b080a672e3de976d2430a@139.162.223.88:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.12.0
Date: Thu, 04 Feb 2021 11:22:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


Really destroying SIP dialog ‘1245733887’ Method: REGISTER

<— SIP read from UDP:196.206.121.13:53468 —>

<------------->
Really destroying SIP dialog ‘521789833’ Method: REGISTER
Really destroying SIP dialog ‘3791014056’ Method: REGISTER

<— SIP read from UDP:196.206.121.13:53468 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 139.162.223.88:5060;rport=5060;received=139.162.223.88;branch=z9hG4bK5d58a9cb
Call-ID: 445483b5345b080a672e3de976d2430a@139.162.223.88:5060
From: “asterisk” sip:asterisk@139.162.223.88;tag=as783f3c8c
To: sip:medeva222990@196.206.121.13;ob;tag=z9hG4bK5d58a9cb
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.20.3
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘445483b5345b080a672e3de976d2430a@139.162.223.88:5060’ Method: OPTIONS
Really destroying SIP dialog ‘2803966815’ Method: REGISTER
Really destroying SIP dialog ‘3513437915’ Method: REGISTER
Really destroying SIP dialog ‘264875975’ Method: REGISTER

<— SIP read from UDP:196.206.121.13:53468 —>
INVITE sip:4915735852785@139.162.223.88:5060 SIP/2.0
Via: SIP/2.0/UDP 196.206.121.13:53468;rport;branch=z9hG4bKPj391d837edbfd4e25ba0d419eb483f166
Max-Forwards: 70
From: sip:medeva222990@139.162.223.88;tag=86c462c0a69f4de2bee71cefe998f67e
To: sip:4915735852785@139.162.223.88;tag=as24714a73
Contact: sip:medeva222990@196.206.121.13:53468;ob
Call-ID: 075b4b68decc40b38308017c930e4686
CSeq: 1747 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 319

v=0
o=- 3821430163 3821430164 IN IP4 196.206.121.13
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4014 RTP/AVP 8 101
c=IN IP4 196.206.121.13
b=TIAS:64000
a=rtcp:4015 IN IP4 192.168.1.9
a=ssrc:1709190253 cname:594a12183f4059c0
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
— (14 headers 15 lines) —
Sending to 196.206.121.13:53468 (NAT)

<— Reliably Transmitting (NAT) to 196.206.121.13:53468 —>
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 196.206.121.13:53468;branch=z9hG4bKPj391d837edbfd4e25ba0d419eb483f166;received=196.206.121.13;rport=53468
From: sip:medeva222990@139.162.223.88;tag=86c462c0a69f4de2bee71cefe998f67e
To: sip:4915735852785@139.162.223.88;tag=as24714a73
Call-ID: 075b4b68decc40b38308017c930e4686
CSeq: 1747 INVITE
Server: Asterisk PBX 16.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘655216389’ Method: REGISTER

<— SIP read from UDP:196.206.121.13:53468 —>
ACK sip:4915735852785@139.162.223.88:5060 SIP/2.0
Via: SIP/2.0/UDP 196.206.121.13:53468;rport;branch=z9hG4bKPj391d837edbfd4e25ba0d419eb483f166
Max-Forwards: 70
From: sip:medeva222990@139.162.223.88;tag=86c462c0a69f4de2bee71cefe998f67e
To: sip:4915735852785@139.162.223.88;tag=as24714a73
Call-ID: 075b4b68decc40b38308017c930e4686
CSeq: 1747 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘075b4b68decc40b38308017c930e4686’ Method: ACK
Really destroying SIP dialog ‘3856312003’ Method: REGISTER
Really destroying SIP dialog ‘2110813098’ Method: REGISTER
Really destroying SIP dialog ‘1336392893’ Method: REGISTER
Really destroying SIP dialog ‘1406768095’ Method: REGISTER
Really destroying SIP dialog ‘517800500’ Method: REGISTER
Really destroying SIP dialog ‘1463453200’ Method: REGISTER
li1481-56*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@li1481-56 ~]#

Sorry for the large CLI log text, I have activated the protocol debugging via “sip set debug on” command.
and this is the results

please help! i need to resolve this asap

Contact your service provider.

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