[asterisk-users] PJSIP configuration question
Dan Cropp
dan at amtelco.com
Thu Dec 11 15:02:37 CST 2014
I am not sure what you mean by the ful SIP signaling?
Here is the trace for the sip.conf which works successfully.
Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK
---- SIP ---
<--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 --->
INVITE sip:8005555555 at 64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>
Contact: <sip:15062fef-986e-4fcf-a93e-06b28da02fff at XXX.XXX.XXX.XXX:5060>
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, REGISTER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Type: application/sdp
Content-Length: 239
v=0
o=- 540555224 540555224 IN IP4 XXX.XXX.XXX.XXX
s=Asterisk
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 10030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (378 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0
<--- Received SIP response (844 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
Record-Route: <sip:64.2.142.93;lr=on>
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:18005555555 at 64.2.142.192>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 32312 32312 IN IP4 64.2.142.192
s=session
c=IN IP4 64.2.142.192
t=0 0
m=audio 17494 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Phone is ringing.
Next, I answer my cell phone
<--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
Record-Route: <sip:64.2.142.93;lr=on>
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:18005555555 at 64.2.142.192>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 32312 32313 IN IP4 64.2.142.192
s=session
c=IN IP4 64.2.142.192
t=0 0
m=audio 17494 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<--- Transmitting SIP request (437 bytes) to UDP:64.2.142.93:5060 --->
ACK sip:18005555555 at 64.2.142.93:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj744f71bf-b90a-4e49-9dfb-42ff4aa3a641
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 ACK
Route: <sip:64.2.142.93;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length: 0
<--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
Record-Route: <sip:64.2.142.93;lr=on>
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:18005555555 at 64.2.142.192>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 32312 32313 IN IP4 64.2.142.192
s=session
c=IN IP4 64.2.142.192
t=0 0
m=audio 17494 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<--- Transmitting SIP request (437 bytes) to UDP:64.2.142.93:5060 --->
ACK sip:18005555555 at 64.2.142.93:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj744f71bf-b90a-4e49-9dfb-42ff4aa3a641
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 ACK
Route: <sip:64.2.142.93;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length: 0
<--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
Record-Route: <sip:64.2.142.93;lr=on>
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:18005555555 at 64.2.142.192>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 32312 32313 IN IP4 64.2.142.192
s=session
c=IN IP4 64.2.142.192
t=0 0
m=audio 17494 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<--- Transmitting SIP request (437 bytes) to UDP:64.2.142.93:5060 --->
ACK sip:18005555555 at 64.2.142.93:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj744f71bf-b90a-4e49-9dfb-42ff4aa3a641
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 ACK
Route: <sip:64.2.142.93;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length: 0
<--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
Record-Route: <sip:64.2.142.93;lr=on>
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:18005555555 at 64.2.142.192>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 32312 32313 IN IP4 64.2.142.192
s=session
c=IN IP4 64.2.142.192
t=0 0
m=audio 17494 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<--- Transmitting SIP request (437 bytes) to UDP:64.2.142.93:5060 --->
ACK sip:18005555555 at 64.2.142.93:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj744f71bf-b90a-4e49-9dfb-42ff4aa3a641
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 ACK
Route: <sip:64.2.142.93;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length: 0
At this point, my cell phone is disconnected, but Asterisk still thinks there is a call.
Next I issue a hangup to Asterisk and it terminates the call
<--- Transmitting SIP request (456 bytes) to UDP:64.2.142.93:5060 --->
BYE sip:18005555555 at 64.2.142.192 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPjd458d550-7bce-4008-813c-c84a0e446a86
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23192 BYE
Route: <sip:64.2.142.93;lr>
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length: 0
<--- Received SIP response (507 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPjd458d550-7bce-4008-813c-c84a0e446a86
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555 at 64.2.142.93>;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23192 BYE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<--- Transmitting SIP request (473 bytes) to UDP:64.2.142.93:5060 --->
OPTIONS sip:64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPjddc1c054-48ce-4714-ba4e-9690aa9be55b
From: <sip:b5a0e71d-9216-4f3c-aedb-a48ecd2fa642 at XXX.XXX.XXX.XXX>;tag=9c4172da-1b32-442b-bea0-75ca3530b661
To: <sip:64.2.142.93>
Contact: <sip:b5a0e71d-9216-4f3c-aedb-a48ecd2fa642 at XXX.XXX.XXX.XXX:5060>
Call-ID: 0817263d-519b-4bcc-aa11-a5bbd8c21f2f
CSeq: 20166 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length: 0
<--- Transmitting SIP request (487 bytes) to UDP:192.168.10.235:5060 --->
OPTIONS sip:291 at 192.168.10.235 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj46636dcf-d468-4738-a053-22334fe4523b
From: <sip:99d5e64e-8f13-4904-9bbf-0a29865e7690 at XXX.XXX.XXX.XXX>;tag=86edf61e-01fd-4fb8-8d38-60d2eabec220
To: <sip:291 at 192.168.10.235>
Contact: <sip:99d5e64e-8f13-4904-9bbf-0a29865e7690 at XXX.XXX.XXX.XXX:5060>
Call-ID: 9664c7bb-b978-4443-9190-bba0d805be47
CSeq: 62000 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length: 0
<--- Received SIP response (465 bytes) from UDP:192.168.10.235:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj46636dcf-d468-4738-a053-22334fe4523b
To: <sip:291 at 192.168.10.235>
From: <sip:99d5e64e-8f13-4904-9bbf-0a29865e7690 at XXX.XXX.XXX.XXX>;tag=86edf61e-01fd-4fb8-8d38-60d2eabec220
Call-ID: 9664c7bb-b978-4443-9190-bba0d805be47
CSeq: 62000 OPTIONS
Contact: <sip:Infinity at 192.168.10.235>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REGISTER,REFER,NOTIFY
Supported: replaces
Accept: application/sdp
<--- Received SIP response (462 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 200 OPTIONS is almost as pointless as T38
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport=5060;branch=z9hG4bKPjddc1c054-48ce-4714-ba4e-9690aa9be55b
From: <sip:b5a0e71d-9216-4f3c-aedb-a48ecd2fa642 at XXX.XXX.XXX.XXX>;tag=9c4172da-1b32-442b-bea0-75ca3530b661
To: <sip:64.2.142.93>;tag=37c906215f6623e2b0c0b8aa47fb6fb6.bc9b
Call-ID: 0817263d-519b-4bcc-aa11-a5bbd8c21f2f
CSeq: 20166 OPTIONS
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0
--- PJSIP ---
Reliably Transmitting (NAT) to 64.2.142.93:5060:
INVITE sip:8005555555 at 64.2.142.93:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK2f5e2a55;rport
Max-Forwards: 70
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555 at 64.2.142.93:5060>
Contact: <sip:291 at XXX.XXX.XXX.XXX:5060>
Call-ID: 783c897d153242595013ae516ebaf649 at XXX.XXX.XXX.XXX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.2.0
Date: Wed, 10 Dec 2014 21:56:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 1537
v=0
o=root 133352036 133352036 IN IP4 XXX.XXX.XXX.XXX
s=Asterisk PBX 12.2.0
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 13752 RTP/AVP 10 4 3 0 8 111 5 7 18 110 117 97 112 9 118 102 115 116 119 107 96 108 109 113 101
a=rtpmap:10 L16/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:119 speex/32000
a=rtpmap:107 opus/48000/2
a=fmtp:107 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:108 SILK/12000
a=fmtp:108 maxaveragebitrate=12000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:109 SILK/16000
a=fmtp:109 maxaveragebitrate=20000
a=fmtp:109 usedtx=0
a=fmtp:109 useinbandfec=1
a=rtpmap:113 SILK/24000
a=fmtp:113 maxaveragebitrate=30000
a=fmtp:113 usedtx=0
a=fmtp:113 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:20
a=sendrecv
---
<--- SIP read from UDP:64.2.142.93:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK2f5e2a55;rport=5060
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555 at 64.2.142.93:5060>
Call-ID: 783c897d153242595013ae516ebaf649 at XXX.XXX.XXX.XXX:5060
CSeq: 102 INVITE
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:64.2.142.93:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bK2f5e2a55;rport=5060
Record-Route: <sip:64.2.142.93;lr=on>
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555 at 64.2.142.93:5060>;tag=as2968a7d2
Call-ID: 783c897d153242595013ae516ebaf649 at XXX.XXX.XXX.XXX:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:18005555555 at 66.241.99.161>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 15367 15367 IN IP4 66.241.99.161
s=session
c=IN IP4 66.241.99.161
t=0 0
m=audio 11460 RTP/AVP 0 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
list_route: route/path hop: <sip:64.2.142.93;lr=on>
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|opus|vp8|silk8|silk12|silk16|silk24), peer - audio=(gsm|ulaw|g729)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 66.241.99.161:11460
<--- SIP read from UDP:64.2.142.93:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bK2f5e2a55;rport=5060
Record-Route: <sip:64.2.142.93;lr=on>
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555 at 64.2.142.93:5060>;tag=as2968a7d2
Call-ID: 783c897d153242595013ae516ebaf649 at XXX.XXX.XXX.XXX:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:18005555555 at 66.241.99.161>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 15367 15368 IN IP4 66.241.99.161
s=session
c=IN IP4 66.241.99.161
t=0 0
m=audio 11460 RTP/AVP 0 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|opus|vp8|silk8|silk12|silk16|silk24), peer - audio=(gsm|ulaw|g729)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 66.241.99.161:11460
list_route: route/path hop: <sip:64.2.142.93;lr=on>
Transmitting (NAT) to 64.2.142.93:5060:
ACK sip:18005555555 at 66.241.99.161 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4ba6d973;rport
Route: <sip:64.2.142.93;lr=on>
Max-Forwards: 70
From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555 at 64.2.142.93:5060>;tag=as2968a7d2
Contact: <sip:291 at XXX.XXX.XXX.XXX:5060>
Call-ID: 783c897d153242595013ae516ebaf649 at XXX.XXX.XXX.XXX:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.2.0
Content-Length: 0
---
[Dec 10 21:56:25] NOTICE[3691][C-00000001]: channel.c:4163 __ast_read: Dropping incompatible voice frame on SIP/outbound.vitelity.net-00000001 of format ulaw since our native format has changed to (gsm)
<--- SIP read from UDP:64.2.142.93:5060 --->
BYE sip:291 at XXX.XXX.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 64.2.142.93;branch=z9hG4bKd12e.ec461794.0
Via: SIP/2.0/UDP 66.241.99.161:5060;received=66.241.99.161;branch=z9hG4bK071dea37;rport=5060
From: <sip:8005555555 at 64.2.142.93:5060>;tag=as2968a7d2
To: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=as5678b23c
Call-ID: 783c897d153242595013ae516ebaf649 at XXX.XXX.XXX.XXX:5060
CSeq: 102 BYE
User-Agent: packetrino
Max-Forwards: 69
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 64.2.142.93:5060 (NAT)
Scheduling destruction of SIP dialog '783c897d153242595013ae516ebaf649 at XXX.XXX.XXX.XXX:5060' in 32000 ms (Method: BYE)
<--- Transmitting (NAT) to 64.2.142.93:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.2.142.93;branch=z9hG4bKd12e.ec461794.0;received=64.2.142.93;rport=5060
Via: SIP/2.0/UDP 66.241.99.161:5060;received=66.241.99.161;branch=z9hG4bK071dea37;rport=5060
From: <sip:8005555555 at 64.2.142.93:5060>;tag=as2968a7d2
To: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=as5678b23c
Call-ID: 783c897d153242595013ae516ebaf649 at XXX.XXX.XXX.XXX:5060
CSeq: 102 BYE
Server: Asterisk PBX 12.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
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