Output of sip debug
:
`<— SIP read from TCP:myPhoneIP:5066 —>
INVITE sip:CalledNumber@AsteriskIP SIP/2.0
Via: SIP/2.0/TCP myPhoneIP:5066;branch=z9hG4bK-e0193350
From: “myPhoneDefinition” sip:_myPhoneDefinition_@_AsteriskIP_;tag=4f9ea0c15f4b7528o0
To: sip:_CalledNumber_@_AsteriskIP_
Call-ID: d99fc775-2735e04@myPhoneIP
CSeq: 101 INVITE
Max-Forwards: 70
Contact: “myPhoneDefinition” sip:_myPhoneDefinition_@_myPhoneIP_:5066;transport=tcp
Expires: 240
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 401
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 58119286 58119286 IN IP4 myPhoneIP
s=-
c=IN IP4 myPhoneIP
t=0 0
m=audio 16418 RTP/AVP 0 2 8 9 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
— (14 headers 18 lines) —
Sending to myPhoneIP:5066 (NAT)
Sending to myPhoneIP:5066 (NAT)
Using INVITE request as basis request - d99fc775-2735e04@myPhoneIP
Found peer ‘myPhoneDefinition’ for ‘myPhoneDefinition’ from myPhoneIP:5066
<— Reliably Transmitting (NAT) to myPhoneIP:5066 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP myPhoneIP:5066;branch=z9hG4bK-e0193350;received=myPhoneIP;rport=5066
From: “myPhoneDefinition” sip:_myPhoneDefinition_@_AsteriskIP_;tag=4f9ea0c15f4b7528o0
To: sip:_CalledNumber_@_AsteriskIP_;tag=as141d2e4b
Call-ID: d99fc775-2735e04@myPhoneIP
CSeq: 101 INVITE
Server: Asterisk PBX 15.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“595712d7”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘d99fc775-2735e04@myPhoneIP’ in 32000 ms (Method: INVITE)
<— SIP read from TCP:myPhoneIP:5066 —>
ACK sip:CalledNumber@AsteriskIP SIP/2.0
Via: SIP/2.0/TCP myPhoneIP:5066;branch=z9hG4bK-e0193350
From: “myPhoneDefinition” sip:_myPhoneDefinition_@_AsteriskIP_;tag=4f9ea0c15f4b7528o0
To: sip:_CalledNumber_@_AsteriskIP_;tag=as141d2e4b
Call-ID: d99fc775-2735e04@myPhoneIP
CSeq: 101 ACK
Max-Forwards: 70
Contact: “myPhoneDefinition” sip:_myPhoneDefinition_@_myPhoneIP_:5066;transport=tcp
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 0
<------------->
— (10 headers 0 lines) —
<— SIP read from TCP:myPhoneIP:5066 —>
INVITE sip:CalledNumber@AsteriskIP SIP/2.0
Via: SIP/2.0/TCP myPhoneIP:5066;branch=z9hG4bK-1d940c
From: “myPhoneDefinition” sip:_myPhoneDefinition_@_AsteriskIP_;tag=4f9ea0c15f4b7528o0
To: sip:_CalledNumber_@_AsteriskIP_
Call-ID: d99fc775-2735e04@myPhoneIP
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username=“myPhoneDefinition”,realm=“asterisk”,nonce=“595712d7”,uri=“sip:CalledNumber@AsteriskIP”,algorithm=MD5,response=“172cf79e08e3ac3b8f23c612a0092ed4”
Contact: “myPhoneDefinition” sip:_myPhoneDefinition_@_myPhoneIP_:5066;transport=tcp
Expires: 240
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 401
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 58119286 58119286 IN IP4 myPhoneIP
s=-
c=IN IP4 myPhoneIP
t=0 0
m=audio 16418 RTP/AVP 0 2 8 9 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
— (15 headers 18 lines) —
Sending to myPhoneIP:5066 (NAT)
Using INVITE request as basis request - d99fc775-2735e04@myPhoneIP
Found peer ‘myPhoneDefinition’ for ‘myPhoneDefinition’ from myPhoneIP:5066
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format G729a for ID 18
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|g726|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|)
> 0x7f5c88013840 – Strict RTP learning after remote address set to: myPhoneIP:16418
Peer audio RTP is at port myPhoneIP:16418
Looking for CalledNumber in default (domain AsteriskIP)
sip_route_dump: route/path hop: sip:_myPhoneDefinition_@_myPhoneIP_:5066;transport=tcp
<— Transmitting (NAT) to myPhoneIP:5066 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TCP myPhoneIP:5066;branch=z9hG4bK-1d940c;received=myPhoneIP;rport=5066
From: “myPhoneDefinition” sip:_myPhoneDefinition_@_AsteriskIP_;tag=4f9ea0c15f4b7528o0
To: sip:_CalledNumber_@_AsteriskIP_
Call-ID: d99fc775-2735e04@myPhoneIP
CSeq: 102 INVITE
Server: Asterisk PBX 15.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:_CalledNumber_@_AsteriskIP_:5060;transport=tcp
Content-Length: 0
<------------>
– Executing [CalledNumber@default:1] Set(“SIP/myPhoneDefinition-00000000”, “CALLERID(num)=myNumber”) in new stack
– Executing [CalledNumber@default:2] Dial(“SIP/myPhoneDefinition-00000000”, “SIP/CalledNumber@SIPTrunkPeerDefinition”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 10264
Adding codec alaw to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to SIPTrunkIP:5060:
INVITE sip:CalledNumber@SIPTrunkRegServer SIP/2.0
Via: SIP/2.0/TCP AsteriskIP:5060;branch=z9hG4bK32c80056;rport
Max-Forwards: 70
From: “myCallerID” sip:_myNumber_@_SIPTrunkDomain_;tag=as4635f0fb
To: sip:_CalledNumber_@_SIPTrunkRegServer_
Contact: sip:_myNumber_@_AsteriskIP_:5060;transport=tcp
Call-ID: 48b8b025074defc251c4a145277e9e3e@SIPTrunkDomain
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.4.0
Date: Mon, 18 Jun 2018 08:30:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 1897568621 1897568621 IN IP4 AsteriskIP
s=Asterisk PBX 15.4.0
c=IN IP4 AsteriskIP
t=0 0
m=audio 10264 RTP/AVP 8 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
-- Called SIP/_CalledNumber_@_SIPTrunkPeerDefinition_
<— SIP read from TCP:SIPTrunkIP:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TCP AsteriskIP:5060;rport=32798;received=myExtIP;branch=z9hG4bK32c80056
To: sip:_CalledNumber_@_SIPTrunkRegServer_
From: “myCallerID” sip:_myNumber_@_SIPTrunkDomain_;tag=as4635f0fb
Call-ID: 48b8b025074defc251c4a145277e9e3e@SIPTrunkDomain
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from TCP:SIPTrunkIP:5060 —>
SIP/2.0 403 Forbidden (R403_REQUEST_NOT_ALLOWED)
Via: SIP/2.0/TCP AsteriskIP:5060;rport=5060;received=myExtIP;branch=z9hG4bK32c80056
To: sip:_CalledNumber_@_SIPTrunkRegServer_;tag=b2a0e993
From: “myCallerID” sip:_myNumber_@_SIPTrunkDomain_;tag=as4635f0fb
Call-ID: 48b8b025074defc251c4a145277e9e3e@SIPTrunkDomain
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Transmitting (NAT) to SIPTrunkIP:5060:
ACK sip:CalledNumber@SIPTrunkRegServer SIP/2.0
Via: SIP/2.0/TCP AsteriskIP:5060;branch=z9hG4bK32c80056;rport
Max-Forwards: 70
From: “myCallerID” sip:_myNumber_@_SIPTrunkDomain_;tag=as4635f0fb
To: sip:_CalledNumber_@_SIPTrunkRegServer_;tag=b2a0e993
Contact: sip:_myNumber_@_AsteriskIP_:5060;transport=tcp
Call-ID: 48b8b025074defc251c4a145277e9e3e@SIPTrunkDomain
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.4.0
Content-Length: 0
[Jun 18 10:30:15] WARNING[32471][C-00000001]: chan_sip.c:24124 handle_response_invite: Received response: “Forbidden” from ‘“myCallerID” sip:_myNumber_@_SIPTrunkDomain_;tag=as4635f0fb’
Scheduling destruction of SIP dialog ‘48b8b025074defc251c4a145277e9e3e@SIPTrunkDomain’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/myPhoneDefinition-00000000’ status is ‘CHANUNAVAIL’
<— Reliably Transmitting (NAT) to myPhoneIP:5066 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP myPhoneIP:5066;branch=z9hG4bK-1d940c;received=myPhoneIP;rport=5066
From: “myPhoneDefinition” sip:_myPhoneDefinition_@_AsteriskIP_;tag=4f9ea0c15f4b7528o0
To: sip:_CalledNumber_@_AsteriskIP_;tag=as37732786
Call-ID: d99fc775-2735e04@myPhoneIP
CSeq: 102 INVITE
Server: Asterisk PBX 15.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0
<------------>
<— SIP read from TCP:myPhoneIP:5066 —>
ACK sip:CalledNumber@AsteriskIP SIP/2.0
Via: SIP/2.0/TCP myPhoneIP:5066;branch=z9hG4bK-1d940c
From: “myPhoneDefinition” sip:_myPhoneDefinition_@_AsteriskIP_;tag=4f9ea0c15f4b7528o0
To: sip:_CalledNumber_@_AsteriskIP_;tag=as37732786
Call-ID: d99fc775-2735e04@myPhoneIP
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“myPhoneDefinition”,realm=“asterisk”,nonce=“595712d7”,uri=“sip:CalledNumber@AsteriskIP”,algorithm=MD5,response=“172cf79e08e3ac3b8f23c612a0092ed4”
Contact: “myPhoneDefinition” sip:_myPhoneDefinition_@_myPhoneIP_:5066;transport=tcp
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘d99fc775-2735e04@myPhoneIP’ Method: ACK`