VoIP Mailing List Archives

[dan at amtelco.com]

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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@64.2.142.93>
Contact: <sip:15062fef-986e-4fcf-a93e-06b28da02fff@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=9c4172da-1b32-442b-bea0-75ca3530b661
To: <sip:64.2.142.93>
Contact: <sip:b5a0e71d-9216-4f3c-aedb-a48ecd2fa642@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@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@XXX.XXX.XXX.XXX>;tag=86edf61e-01fd-4fb8-8d38-60d2eabec220
To: <sip:291@192.168.10.235>
Contact: <sip:99d5e64e-8f13-4904-9bbf-0a29865e7690@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@192.168.10.235>
From: <sip:99d5e64e-8f13-4904-9bbf-0a29865e7690@XXX.XXX.XXX.XXX>;tag=86edf61e-01fd-4fb8-8d38-60d2eabec220
Call-ID: 9664c7bb-b978-4443-9190-bba0d805be47
CSeq: 62000 OPTIONS
Contact: <sip:Infinity@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@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@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@XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555@64.2.142.93:5060>
Contact: <sip:291@XXX.XXX.XXX.XXX:5060>
Call-ID: 783c897d153242595013ae516ebaf649@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@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@XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555@64.2.142.93:5060>
Call-ID: 783c897d153242595013ae516ebaf649@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@XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555@64.2.142.93:5060>;tag=as2968a7d2
Call-ID: 783c897d153242595013ae516ebaf649@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@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@XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555@64.2.142.93:5060>;tag=as2968a7d2
Call-ID: 783c897d153242595013ae516ebaf649@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@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@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@XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555@64.2.142.93:5060>;tag=as2968a7d2
Contact: <sip:291@XXX.XXX.XXX.XXX:5060>
Call-ID: 783c897d153242595013ae516ebaf649@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@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@64.2.142.93:5060>;tag=as2968a7d2
To: "Dan" <sip:291@XXX.XXX.XXX.XXX>;tag=as5678b23c
Call-ID: 783c897d153242595013ae516ebaf649@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@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@64.2.142.93:5060>;tag=as2968a7d2
To: "Dan" <sip:291@XXX.XXX.XXX.XXX>;tag=as5678b23c
Call-ID: 783c897d153242595013ae516ebaf649@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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[dan at amtelco.com]

Ugh.

I'm having a bad day. The two traces were swapped.

The one on Asterisk 13 is PJSIP.
The one on Asterisk 12 is using chan_sip.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dan Cropp
Sent: Thursday, December 11, 2014 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question

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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@64.2.142.93>
Contact: <sip:15062fef-986e-4fcf-a93e-06b28da02fff@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: <sip:8005555555@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@XXX.XXX.XXX.XXX>;tag=9c4172da-1b32-442b-bea0-75ca3530b661
To: <sip:64.2.142.93>
Contact: <sip:b5a0e71d-9216-4f3c-aedb-a48ecd2fa642@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@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@XXX.XXX.XXX.XXX>;tag=86edf61e-01fd-4fb8-8d38-60d2eabec220
To: <sip:291@192.168.10.235>
Contact: <sip:99d5e64e-8f13-4904-9bbf-0a29865e7690@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@192.168.10.235>
From: <sip:99d5e64e-8f13-4904-9bbf-0a29865e7690@XXX.XXX.XXX.XXX>;tag=86edf61e-01fd-4fb8-8d38-60d2eabec220
Call-ID: 9664c7bb-b978-4443-9190-bba0d805be47
CSeq: 62000 OPTIONS
Contact: <sip:Infinity@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@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@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@XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555@64.2.142.93:5060>
Contact: <sip:291@XXX.XXX.XXX.XXX:5060>
Call-ID: 783c897d153242595013ae516ebaf649@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@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@XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555@64.2.142.93:5060>
Call-ID: 783c897d153242595013ae516ebaf649@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@XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555@64.2.142.93:5060>;tag=as2968a7d2
Call-ID: 783c897d153242595013ae516ebaf649@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@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@XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555@64.2.142.93:5060>;tag=as2968a7d2
Call-ID: 783c897d153242595013ae516ebaf649@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@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@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@XXX.XXX.XXX.XXX>;tag=as5678b23c
To: <sip:8005555555@64.2.142.93:5060>;tag=as2968a7d2
Contact: <sip:291@XXX.XXX.XXX.XXX:5060>
Call-ID: 783c897d153242595013ae516ebaf649@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@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@64.2.142.93:5060>;tag=as2968a7d2
To: "Dan" <sip:291@XXX.XXX.XXX.XXX>;tag=as5678b23c
Call-ID: 783c897d153242595013ae516ebaf649@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@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@64.2.142.93:5060>;tag=as2968a7d2
To: "Dan" <sip:291@XXX.XXX.XXX.XXX>;tag=as5678b23c
Call-ID: 783c897d153242595013ae516ebaf649@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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_____________________________________________________________________
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To UNSUBSCRIBE or update options visit:
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[dan at amtelco.com]

Trying this again after my first away from work in a couple weeks.

Running Asterisk 13.0.0
IP authentication with Vitelity

I can Originate with sip, but not pjsip.
Here is the sip settings and trace.

Action: Originate
ActionID: S8
Channel: SIP/8005555555@outbound.vitelity.net
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
CallerID: John Doe <1234>
Variable: CALLERID(num-pres)=allowed_passed_screen
Async: true

sip.conf
[HVout]
type=friend
dtmfmode=auto
host=64.2.142.93
disallow=all
allow=ulaw
canreinvite=no
trustrpid=yes
sendrpid=yes
nat=yes
context=TestApp


== Using SIP RTP CoS mark 5
Audio is at 18226
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 64.2.142.189:5060:
INVITE sip:8005555555@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 192.168.11.166:5060;branch=z9hG4bK40275183
Max-Forwards: 70
From: "John Doe" <sip:1234@192.168.11.166>;tag=as466267de
To: <sip:8005555555@outbound.vitelity.net>
Contact: <sip:1234@192.168.11.166:5060>
Call-ID: 59e9eff8339e32af271c23541298135d@192.168.11.166:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.0.0
Date: Sun, 21 Dec 2014 20:06:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1422632184 1422632184 IN IP4 192.168.11.166
s=Asterisk PBX 13.0.0
c=IN IP4 192.168.11.166
t=0 0
m=audio 18226 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
-- Called 8005555555@outbound.vitelity.net

<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.166:5060;branch=z9hG4bK40275183;received=192.168.11.166
From: "John Doe" <sip:1234@192.168.11.166>;tag=as466267de
To: <sip:8005555555@outbound.vitelity.net>
Call-ID: 59e9eff8339e32af271c23541298135d@192.168.11.166:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:8005555555@64.2.142.189>
Content-Length: 0

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

<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.11.166:5060;branch=z9hG4bK40275183;received=192.168.11.166
From: "John Doe" <sip:1234@192.168.11.166>;tag=as466267de
To: <sip:8005555555@outbound.vitelity.net>;tag=as5458ca04
Call-ID: 59e9eff8339e32af271c23541298135d@192.168.11.166:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:8005555555@64.2.142.189>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 21997 21997 IN IP4 64.2.142.189
s=session
c=IN IP4 64.2.142.189
t=0 0
m=audio 19282 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 14 lines) ---
sip_route_dump: route/path hop: <sip:8005555555@64.2.142.189>
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.2.142.189:19282
-- SIP/outbound.vitelity.net-00000000 is making progress

[dan at amtelco.com]

I am running PJPROJECT 2.3 and Asterisk 13.0.0.

I answer the call, about 15 seconds later, vitality hangs up on my cell phone.
However, Asterisk is never notified
When the OK (for the answer) occurs, the ACK seems to never be accepted.
The OK recvd with ACK sent occurs several times.

Here are the pjsip.conf settings…
[global]
type = global
debug = yes

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = [url=sip:64.2.142.93]sip:64.2.142.93[/url]

[outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net
dtmf_mode = rfc4733
force_rport = yes
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
trust_id_inbound = yes
disallow = all
allow = ulaw
direct_media = no

[outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.net
match = 64.2.142.93


When I Originate a call via AMI…

Action: Originate
ActionID: S8
Channel: PJSIP/8005555555@outbound.vitelity.net ([email]PJSIP/8005555555@outbound.vitelity.net[/email])
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
CallerID: John Doe <1234>
Variable: CALLERID(num-pres)=allowed_passed_screen
Async: true
The call goes through to my cell phone. I answer on my cell phone and Asterisk sees the call being answered. However, Vitelity disconnects the cell phone about 15 seconds later.
When looking at the PJSIP trace, the ACK repsonse to the 200 OK (Answer) are missing the Contact header. From what I understand that is likely the reason Vitelity doesn’t seem to process the ACK.

*CLI> -- Called 8005555555@outbound.vitelity.net (8005555555@outbound.vitelity.net)
<--- Transmitting SIP request (1018 bytes) to UDP:64.2.142.93:5060 --->
INVITE [url=sip:8005555555@64.2.142.93]sip:8005555555@64.2.142.93[/url] SIP/2.0
Via: SIP/2.0/UDP 192.168.11.166:5060;rport;branch=z9hG4bKPjc0fc94dc-bde3-441c-8a58-cd71e2d326f2
From: "John Doe" <[url=sip:1234@192.168.11.166]sip:1234@192.168.11.166[/url]>;tag=340be959-af87-4b77-a7a7-8aaf83940b03
To: <[url=sip:8005555555@64.2.142.93]sip:8005555555@64.2.142.93[/url]>
Contact: <[url=sip:63240147-e592-47ae-9fbd-4f1cfbb5c5a6@192.168.11.166:5060]sip:63240147-e592-47ae-9fbd-4f1cfbb5c5a6@192.168.11.166:5060[/url]>
Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125
CSeq: 10207 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: "John Doe" <[url=sip:1234@192.168.11.166]sip:1234@192.168.11.166[/url]>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 2134048799 2134048799 IN IP4 192.168.11.166
s=Asterisk
c=IN IP4 192.168.11.166
t=0 0
m=audio 16262 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 (384 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.11.166:5060;rport=5060;branch=z9hG4bKPjc0fc94dc-bde3-441c-8a58-cd71e2d326f2
From: "John Doe" <[url=sip:1234@192.168.11.166]sip:1234@192.168.11.166[/url]>;tag=340be959-af87-4b77-a7a7-8aaf83940b03
To: <[url=sip:8005555555@64.2.142.93]sip:8005555555@64.2.142.93[/url]>
Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125
CSeq: 10207 INVITE
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0


<--- Received SIP response (851 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.11.166:5060;received=192.168.11.166;rport=5060;branch=z9hG4bKPjc0fc94dc-bde3-441c-8a58-cd71e2d326f2
Record-Route: <[url=sip:64.2.142.93;lr=on]sip:64.2.142.93;lr=on[/url]>
From: "John Doe" <[url=sip:1234@192.168.11.166]sip:1234@192.168.11.166[/url]>;tag=340be959-af87-4b77-a7a7-8aaf83940b03
To: <[url=sip:8005555555@64.2.142.93]sip:8005555555@64.2.142.93[/url]>;tag=as466c6135
Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125
CSeq: 10207 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <[url=sip:18005555555@66.241.99.145]sip:18005555555@66.241.99.145[/url]>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 3457 3457 IN IP4 66.241.99.145
s=session
c=IN IP4 66.241.99.145
t=0 0
m=audio 11872 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

-- PJSIP/outbound.vitelity.net-00000000 is making progress

[dan at amtelco.com]

Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0.

Same problem is happening with both of them.

Could this be caused by PJPROJECT 2.3?

Anyone have any suggestions for what I can try?

My boss is giving me until tomorrow to get the PJSIP support working with Vitelity. Otherwise, he’s told me to go back to using chan_sip and wait a year or two for PJSIP to be in the field more.

Have a great day!

Dan

[george.joseph at fairv...]

On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:

[dan at amtelco.com]

Hi George,

Thank you for looking into this.
This is behind a nat…

[global]
type = global
debug = yes

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = [url=sip:64.2.142.93]sip:64.2.142.93[/url]

[outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net
dtmf_mode = rfc4733
force_rport = yes
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
trust_id_inbound = yes
disallow = all
allow = ulaw
direct_media = no

[outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.net
match = 64.2.142.93

Have a great day!

Dan

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 3:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question

On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:
Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0.

Same problem is happening with both of them.

Could this be caused by PJPROJECT 2.3?

Anyone have any suggestions for what I can try?

My boss is giving me until tomorrow to get the PJSIP support working with Vitelity. Otherwise, he’s told me to go back to using chan_sip and wait a year or two for PJSIP to be in the field more.




I have a Vitelity account I can try. Re-post your pjsip config and I'll try it now.

[george.joseph at fairv...]

On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:

[dan at amtelco.com]

Yes, everything is behind the same NAT.

For the application I’m working on, the only endpoint is the endpoint to Vitelity.
We use AMI to Originate calls from Asterisk endpoint through Vitelity to phones.
After that, we control the call through AMI to perform the work we need.



From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 4:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question



On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:
Hi George,

Thank you for looking into this.
This is behind a nat…





Just to be clear...both the pbx and local endpoints are behind the same NAT?

[george.joseph at fairv...]

On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:

[dan at amtelco.com]

Yes, outbound calls are the only ones I’m trying.


From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question



On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:
Yes, everything is behind the same NAT.

For the application I’m working on, the only endpoint is the endpoint to Vitelity.
We use AMI to Originate calls from Asterisk endpoint through Vitelity to phones.
After that, we control the call through AMI to perform the work we need.





And it's outbound calls that aren't working right?

[george.joseph at fairv...]

Ok Dan, try this...  I was able to get this to work behind a NAT and with ip address authentication.

[global]
type = global
debug = yes

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
local_net=<yourlocalnet I.E. 10.10.10.10/24>
external_media_address=<your public ip address>
external_signaling_address=<your public address>
 
[outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93


[outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net
dtmf_mode = rfc4733
force_rport = yes
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
trust_id_inbound = yes
disallow = all
allow = ulaw
direct_media = no
from_user=<your main vitelity account name>  ; Not subaccount


[outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.net
match = 64.2.142.93

[dan at amtelco.com]

Thanks George.


I will remote into and give this a try.

Have a great evening!


Dan

On Dec 15, 2014, at 7:27 PM, George Joseph <george.joseph@fairview5.com (george.joseph@fairview5.com)> wrote:

[dan at amtelco.com]

I am not sure if I entered the correct settings for the transport information.
For the local_net, I entered my local ip address, but no mask.  I will check with the network admin so he can verify the settings I entered.

One minor detail, we are using ip authentication.  When Vitelity changed my account from user based authentication to IP based authentication, they stopped including a user for the account.

Should these settings work without the from_user (IP based authentication) or do I need to get the account name from Vitelity?

Have a great day!

Da

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 7:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question

Ok Dan, try this... I was able to get this to work behind a NAT and with ip address authentication.

[global]
type = global
debug = yes
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
local_net=<yourlocalnet I.E. 10.10.10.10/24>
external_media_address=<your public ip address>
external_signaling_address=<your public address>

[outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = [url=sip:64.2.142.93]sip:64.2.142.93[/url]

[outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net
dtmf_mode = rfc4733
force_rport = yes
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
trust_id_inbound = yes
disallow = all
allow = ulaw
direct_media = no

from_user=<your main vitelity account name> ; Not subaccount

[outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.net
match = 64.2.142.93

[george.joseph at fairv...]

On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote: