Certified Asterisk 16.8-cert5 Codec Negotiation

georgequalley · November 30, 2020, 3:55am

I have two servers running Certified Asterisk 16.8-cert5. My main server (let’s call it Server A) has handled several remote (to it) Digium phones. I recently installed 15 phones at a remote location and configured a second server (Server B) on the premises to serve as a trunk to Server A. My VOIP provider is Amazon Chime. So in simple terms my setup (for the purposes of this question is:

Amazon -> Server A -> Server B -> Digium Phones

Prior to this setup everything was working great and everything runs over G.722 from end to end. After MUCH configuration and tweaking, I figured out that the only way to get the above setup working was to make my Server A -> Server B trunk only support G.711 (ulaw). Anything else fails with a: SIP/2.0 400 re-INVITE SDP encryption settings incorrect message from Amazon when the call is connected on Server B.

I’m currently using: direct_media = no on my trunk between Server A -> Server B. If I connect a call on Server A (for example, Answer() and play a Background()) it works fine. But the minute I send the call to Server B (and try to Answer() it – although ringing works fine) the call is terminated.

Any help would be greatly appreciated!

david551 · November 30, 2020, 10:21am

Unless you have a support contract with Digium, in which case you should be getting support under that, you should be using at least Asterisk 16.15.0. if you need to debug! Generally, you should only use certified versions with a support contract.

You didn’t say you were using PJSIP, although the spelling of direct_media indicates that you are.

I assume you mean the second, not the minute. There are rather a lot of ways a call can fail in the first minute.

Please provide logs from both A and B at a sufficient detail level to see why the call fails.

georgequalley · November 30, 2020, 4:18pm

Hi David,

I do not currently have a support contract but I’m using a Certified version of asterisk because I have Digium/Sangoma phones running on DPMA.

My mistake. I am using PJSIP.

Again, my mistake. You are correct, the disconnect is nearly instantaneous.

The following is the PJSIP logger output from server A.

<--- Received SIP request (989 bytes) from UDP:99.77.253.10:5060 --->
INVITE sip:+13193587700@voip.qsquaredsystems.com:5060;transport=UDP SIP/2.0
Record-Route: <sip:99.77.253.10;lr;ftag=DgKraN67Q4S3a;did=152.7111;nat=yes>
Via: SIP/2.0/UDP 99.77.253.10:5060;branch=z9hG4bK5d44.24f161d0536bd9f06b7fd1f2b4f1c4da.0
Via: SIP/2.0/UDP 10.0.127.189;received=10.0.127.189;rport=5060;branch=z9hG4bK8128tygj5cD2H
Max-Forwards: 69
From: <sip:+15155565058@10.0.127.189:5060>;tag=DgKraN67Q4S3a
To: <sip:+13193587700@voip.qsquaredsystems.com:5060>;transport=UDP
Call-ID: d3300f3b-d270-41ea-abbe-ca9eb31a55e2
CSeq: 28787635 INVITE
Contact: <sip:10.0.127.189:5060;alias=10.0.127.189~5060~1>
Content-Type: application/sdp
Content-Length: 249
X-VoiceConnector-ID: cbdgjywnile6jesj1csh9w
User-Agent: VineProx-v2.3

v=0
o=Sonus_UAC 104564 4057 IN IP4 99.77.253.131
s=SIP Media Capabilities
c=IN IP4 99.77.253.131
t=0 0
m=audio 33394 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:33395
a=ptime:20

  == Setting global variable 'SIPDOMAIN' to 'voip.qsquaredsystems.com'
<--- Transmitting SIP response (519 bytes) to UDP:99.77.253.10:5060 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP 99.77.253.10:5060;rport=5060;received=99.77.253.10;branch=z9hG4bK5d44.24f161d0536bd9f06b7fd1f2b4f1c4da.0
v: SIP/2.0/UDP 10.0.127.189;rport=5060;received=10.0.127.189;branch=z9hG4bK8128tygj5cD2H
Record-Route: <sip:99.77.253.10;lr;ftag=DgKraN67Q4S3a;did=152.7111;nat=yes>
i: d3300f3b-d270-41ea-abbe-ca9eb31a55e2
f: <sip:+15155565058@10.0.127.189>;tag=DgKraN67Q4S3a
t: <sip:+13193587700@voip.qsquaredsystems.com>;transport=UDP
CSeq: 28787635 INVITE
Server: Asterisk PBX
l:  0


    -- Executing [+13193587700@inbound:1] NoOp("PJSIP/aws-000000bc", "") in new stack
    -- Executing [+13193587700@inbound:2] Answer("PJSIP/aws-000000bc", "") in new stack
<--- Transmitting SIP response (966 bytes) to UDP:99.77.253.10:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 99.77.253.10:5060;rport=5060;received=99.77.253.10;branch=z9hG4bK5d44.24f161d0536bd9f06b7fd1f2b4f1c4da.0
v: SIP/2.0/UDP 10.0.127.189;rport=5060;received=10.0.127.189;branch=z9hG4bK8128tygj5cD2H
Record-Route: <sip:99.77.253.10;lr;ftag=DgKraN67Q4S3a;did=152.7111;nat=yes>
i: d3300f3b-d270-41ea-abbe-ca9eb31a55e2
f: <sip:+15155565058@10.0.127.189>;tag=DgKraN67Q4S3a
t: <sip:+13193587700@voip.qsquaredsystems.com>;tag=8ab1e679-a18a-404e-97ae-25da7f7833f6;transport=UDP
CSeq: 28787635 INVITE
Server: Asterisk PBX
m: <sip:52.89.237.252:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
k: 100rel, timer, replaces, norefersub
c: application/sdp
l:   207

v=0
o=- 104564 4059 IN IP4 52.89.237.252
s=Asterisk
c=IN IP4 52.89.237.252
t=0 0
m=audio 16288 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (495 bytes) from UDP:99.77.253.10:5060 --->
ACK sip:52.89.237.252:5060 SIP/2.0
Via: SIP/2.0/UDP 99.77.253.10:5060;branch=z9hG4bK5d44.c94a4f77ae7fbcd49eb4515b042309d5.0
Via: SIP/2.0/UDP 10.0.127.189;received=10.0.127.189;rport=5060;branch=z9hG4bK9av1vS1N2N3mD
Max-Forwards: 69
From: <sip:+15155565058@10.0.127.189:5060>;tag=DgKraN67Q4S3a
To: <sip:+13193587700@voip.qsquaredsystems.com:5060>;transport=UDP;tag=8ab1e679-a18a-404e-97ae-25da7f7833f6
Call-ID: d3300f3b-d270-41ea-abbe-ca9eb31a55e2
CSeq: 28787635 ACK
Content-Length: 0


    -- Executing [+13193587700@inbound:3] Wait("PJSIP/aws-000000bc", "1") in new stack
    -- Executing [+13193587700@inbound:4] Dial("PJSIP/aws-000000bc", "PJSIP/+13193587700@bfs-trunk") in new stack
    -- Called PJSIP/+13193587700@bfs-trunk
    -- PJSIP/aws-000000bc requested media update control 26, passing it to PJSIP/bfs-trunk-000000bd
<--- Transmitting SIP request (826 bytes) to UDP:50.81.142.253:5060 --->
INVITE sip:+13193587700@50.81.142.253:5060 SIP/2.0
v: SIP/2.0/UDP 52.89.237.252:5060;rport;branch=z9hG4bKPj4e59dd94-981f-469e-8c92-e61db67f556b
f: <sip:+15155565058@172.31.24.143>;tag=103d32af-dbf9-4543-8054-19f78508e608
t: <sip:+13193587700@50.81.142.253>
m: <sip:asterisk@52.89.237.252:5060>
i: 9ef93607-098a-4e83-8312-5ff8f3f7e0fa
CSeq: 4595 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
k: 100rel, timer, replaces, norefersub
x: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX
c: application/sdp
l:   219

v=0
o=- 503096982 503096982 IN IP4 52.89.237.252
s=Asterisk
c=IN IP4 52.89.237.252
t=0 0
m=audio 16932 RTP/AVP 0 9 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (348 bytes) from UDP:50.81.142.253:5060 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP 52.89.237.252:5060;rport=1049;received=52.89.237.252;branch=z9hG4bKPj4e59dd94-981f-469e-8c92-e61db67f556b
i: 9ef93607-098a-4e83-8312-5ff8f3f7e0fa
f: <sip:+15155565058@172.31.24.143>;tag=103d32af-dbf9-4543-8054-19f78508e608
t: <sip:+13193587700@50.81.142.253>
CSeq: 4595 INVITE
Server: Asterisk PBX
l:  0


<--- Received SIP response (846 bytes) from UDP:50.81.142.253:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 52.89.237.252:5060;rport=1049;received=52.89.237.252;branch=z9hG4bKPj4e59dd94-981f-469e-8c92-e61db67f556b
i: 9ef93607-098a-4e83-8312-5ff8f3f7e0fa
f: <sip:+15155565058@172.31.24.143>;tag=103d32af-dbf9-4543-8054-19f78508e608
t: <sip:+13193587700@50.81.142.253>;tag=19da2428-ce82-47b6-822a-3fb6126781ae
CSeq: 4595 INVITE
Server: Asterisk PBX
m: <sip:50.81.142.253:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
k: 100rel, timer, replaces, norefersub
x: 1800;refresher=uac
Require: timer
c: application/sdp
l:   219

v=0
o=- 503096982 503096984 IN IP4 50.81.142.253
s=Asterisk
c=IN IP4 50.81.142.253
t=0 0
m=audio 17170 RTP/AVP 9 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

    -- PJSIP/bfs-trunk-000000bd answered PJSIP/aws-000000bc
<--- Transmitting SIP request (396 bytes) to UDP:50.81.142.253:5060 --->
ACK sip:50.81.142.253:5060 SIP/2.0
v: SIP/2.0/UDP 52.89.237.252:5060;rport;branch=z9hG4bKPjdc9040b6-10ab-4844-bb64-1f5cd525ef99
f: <sip:+15155565058@172.31.24.143>;tag=103d32af-dbf9-4543-8054-19f78508e608
t: <sip:+13193587700@50.81.142.253>;tag=19da2428-ce82-47b6-822a-3fb6126781ae
i: 9ef93607-098a-4e83-8312-5ff8f3f7e0fa
CSeq: 4595 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX
l:  0


    -- Channel PJSIP/bfs-trunk-000000bd joined 'simple_bridge' basic-bridge <0d6b9ff8-acc6-417a-83b7-4b7f32542e93>
    -- Channel PJSIP/aws-000000bc joined 'simple_bridge' basic-bridge <0d6b9ff8-acc6-417a-83b7-4b7f32542e93>
<--- Transmitting SIP request (934 bytes) to UDP:99.77.253.10:5060 --->
INVITE sip:10.0.127.189:5060;alias=10.0.127.189~5060~1 SIP/2.0
v: SIP/2.0/UDP 52.89.237.252:5060;rport;branch=z9hG4bKPj0062216d-57a7-4aaa-9c26-687d2d6297f3
f: <sip:+13193587700@voip.qsquaredsystems.com>;tag=8ab1e679-a18a-404e-97ae-25da7f7833f6;transport=UDP
t: <sip:+15155565058@10.0.127.189>;tag=DgKraN67Q4S3a
m: <sip:52.89.237.252:5060>
i: d3300f3b-d270-41ea-abbe-ca9eb31a55e2
CSeq: 13534 INVITE
Route: <sip:99.77.253.10;lr;ftag=DgKraN67Q4S3a;did=152.7111;nat=yes>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
k: 100rel, timer, replaces, norefersub
x: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX
c: application/sdp
l:   211

v=0
o=- 104564 4060 IN IP4 52.89.237.252
s=Asterisk
c=IN IP4 52.89.237.252
t=0 0
m=audio 16288 RTP/AVP 9 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (381 bytes) from UDP:99.77.253.10:5060 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP 52.89.237.252:5060;rport=5060;branch=z9hG4bKPj0062216d-57a7-4aaa-9c26-687d2d6297f3;received=52.89.237.252
f: <sip:+13193587700@voip.qsquaredsystems.com>;tag=8ab1e679-a18a-404e-97ae-25da7f7833f6;transport=UDP
t: <sip:+15155565058@10.0.127.189>;tag=DgKraN67Q4S3a
i: d3300f3b-d270-41ea-abbe-ca9eb31a55e2
CSeq: 13534 INVITE
Content-Length: 0


<--- Received SIP response (462 bytes) from UDP:99.77.253.10:5060 --->
SIP/2.0 400 re-INVITE SDP encryption settings incorrect
v: SIP/2.0/UDP 52.89.237.252:5060;received=52.89.237.252;rport=5060;branch=z9hG4bKPj0062216d-57a7-4aaa-9c26-687d2d6297f3
From: <sip:+13193587700@voip.qsquaredsystems.com>;tag=8ab1e679-a18a-404e-97ae-25da7f7833f6;transport=UDP
To: <sip:+15155565058@10.0.127.189>;tag=DgKraN67Q4S3a
Call-ID: d3300f3b-d270-41ea-abbe-ca9eb31a55e2
CSeq: 13534 INVITE
Contact: <sip:10.0.127.189:5060>
Content-Length: 0


<--- Transmitting SIP request (493 bytes) to UDP:99.77.253.10:5060 --->
ACK sip:10.0.127.189:5060;alias=10.0.127.189~5060~1 SIP/2.0
v: SIP/2.0/UDP 52.89.237.252:5060;rport;branch=z9hG4bKPj0062216d-57a7-4aaa-9c26-687d2d6297f3
f: <sip:+13193587700@voip.qsquaredsystems.com>;tag=8ab1e679-a18a-404e-97ae-25da7f7833f6;transport=UDP
t: <sip:+15155565058@10.0.127.189>;tag=DgKraN67Q4S3a
i: d3300f3b-d270-41ea-abbe-ca9eb31a55e2
CSeq: 13534 ACK
Route: <sip:99.77.253.10;lr;ftag=DgKraN67Q4S3a;did=152.7111;nat=yes>
Max-Forwards: 70
User-Agent: Asterisk PBX
l:  0


<--- Transmitting SIP request (517 bytes) to UDP:99.77.253.10:5060 --->
BYE sip:10.0.127.189:5060;alias=10.0.127.189~5060~1 SIP/2.0
v: SIP/2.0/UDP 52.89.237.252:5060;rport;branch=z9hG4bKPjc5a282fa-a518-4138-b9a9-507202da378b
f: <sip:+13193587700@voip.qsquaredsystems.com>;tag=8ab1e679-a18a-404e-97ae-25da7f7833f6;transport=UDP
t: <sip:+15155565058@10.0.127.189>;tag=DgKraN67Q4S3a
i: d3300f3b-d270-41ea-abbe-ca9eb31a55e2
CSeq: 13535 BYE
Route: <sip:99.77.253.10;lr;ftag=DgKraN67Q4S3a;did=152.7111;nat=yes>
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX
l:  0


<--- Received SIP response (384 bytes) from UDP:99.77.253.10:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 52.89.237.252:5060;received=52.89.237.252;rport=5060;branch=z9hG4bKPjc5a282fa-a518-4138-b9a9-507202da378b
From: <sip:+13193587700@voip.qsquaredsystems.com>;tag=8ab1e679-a18a-404e-97ae-25da7f7833f6;transport=UDP
To: <sip:+15155565058@10.0.127.189>;tag=DgKraN67Q4S3a
Call-ID: d3300f3b-d270-41ea-abbe-ca9eb31a55e2
CSeq: 13535 BYE
Content-Length: 0


    -- Channel PJSIP/aws-000000bc left 'simple_bridge' basic-bridge <0d6b9ff8-acc6-417a-83b7-4b7f32542e93>
  == Spawn extension (inbound, +13193587700, 4) exited non-zero on 'PJSIP/aws-000000bc'
    -- Channel PJSIP/bfs-trunk-000000bd left 'simple_bridge' basic-bridge <0d6b9ff8-acc6-417a-83b7-4b7f32542e93>
<--- Transmitting SIP request (420 bytes) to UDP:50.81.142.253:5060 --->
BYE sip:50.81.142.253:5060 SIP/2.0
v: SIP/2.0/UDP 52.89.237.252:5060;rport;branch=z9hG4bKPj166713a6-5ace-416e-a99a-b6548f3c34a6
f: <sip:+15155565058@172.31.24.143>;tag=103d32af-dbf9-4543-8054-19f78508e608
t: <sip:+13193587700@50.81.142.253>;tag=19da2428-ce82-47b6-822a-3fb6126781ae
i: 9ef93607-098a-4e83-8312-5ff8f3f7e0fa
CSeq: 4596 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX
l:  0

And for the same call being connected to Server B


    -- Executing [+13193587700@inbound:1] Wait("PJSIP/bfs-trunk-00000068", "1") in new stack
<--- Transmitting SIP request (419 bytes) to UDP:192.168.2.42:5060 --->
OPTIONS sip:line-bfs-24@192.168.2.42:5060;ob SIP/2.0
v: SIP/2.0/UDP 192.168.2.100:5060;rport;branch=z9hG4bKPj71e56c60-b279-46c6-970c-f3af1714ea12
f: <sip:line-bfs-24@192.168.2.100>;tag=ae95bbb2-7aed-4053-a125-f34af8f48c1d
t: <sip:line-bfs-24@192.168.2.42;ob>
m: <sip:line-bfs-24@192.168.2.100:5060>
i: e435c47d-6e06-455a-9b49-776c9ad4239f
CSeq: 34390 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX
l:  0


<--- Received SIP response (802 bytes) from UDP:192.168.2.42:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.100:5060;rport=5060;received=192.168.2.100;branch=z9hG4bKPj71e56c60-b279-46c6-970c-f3af1714ea12
Call-ID: e435c47d-6e06-455a-9b49-776c9ad4239f
From: <sip:line-bfs-24@192.168.2.100>;tag=ae95bbb2-7aed-4053-a125-f34af8f48c1d
To: <sip:line-bfs-24@192.168.2.42;ob>;tag=z9hG4bKPj71e56c60-b279-46c6-970c-f3af1714ea12
CSeq: 34390 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: Digium D60 2_9_9
Content-Length:  0


    -- Executing [+13193587700@inbound:2] Goto("PJSIP/bfs-trunk-00000068", "bfs-in,s,1") in new stack
    -- Goto (bfs-in,s,1)
    -- Executing [s@bfs-in:1] Answer("PJSIP/bfs-trunk-00000068", "") in new stack
<--- Transmitting SIP response (849 bytes) to UDP:52.89.237.252:1049 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 52.89.237.252:5060;rport=1049;received=52.89.237.252;branch=z9hG4bKPj9da92ae2-5162-4da6-b586-4f32f77c43a7
i: 27086479-330f-4b1f-b3cb-fdd4b5d4602c
f: <sip:+15155565058@172.31.24.143>;tag=180c636a-771f-466f-908b-81184a8b2801
t: <sip:+13193587700@50.81.142.253>;tag=d0a3706a-8e7a-4bc8-8878-efe2aa36fde7
CSeq: 28650 INVITE
Server: Asterisk PBX
m: <sip:50.81.142.253:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
k: 100rel, timer, replaces, norefersub
x: 1800;refresher=uac
Require: timer
c: application/sdp
l:   221

v=0
o=- 1883855816 1883855818 IN IP4 50.81.142.253
s=Asterisk
c=IN IP4 50.81.142.253
t=0 0
m=audio 10864 RTP/AVP 9 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (397 bytes) from UDP:52.89.237.252:1049 --->
ACK sip:50.81.142.253:5060 SIP/2.0
v: SIP/2.0/UDP 52.89.237.252:5060;rport;branch=z9hG4bKPj16647245-8f26-44c1-8b2e-57463262e766
f: <sip:+15155565058@172.31.24.143>;tag=180c636a-771f-466f-908b-81184a8b2801
t: <sip:+13193587700@50.81.142.253>;tag=d0a3706a-8e7a-4bc8-8878-efe2aa36fde7
i: 27086479-330f-4b1f-b3cb-fdd4b5d4602c
CSeq: 28650 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX
l:  0


    -- Executing [s@bfs-in:2] Wait("PJSIP/bfs-trunk-00000068", "1") in new stack
<--- Received SIP request (421 bytes) from UDP:52.89.237.252:1049 --->
BYE sip:50.81.142.253:5060 SIP/2.0
v: SIP/2.0/UDP 52.89.237.252:5060;rport;branch=z9hG4bKPj1af3fe89-618c-4737-9d37-c549cbbdd3b1
f: <sip:+15155565058@172.31.24.143>;tag=180c636a-771f-466f-908b-81184a8b2801
t: <sip:+13193587700@50.81.142.253>;tag=d0a3706a-8e7a-4bc8-8878-efe2aa36fde7
i: 27086479-330f-4b1f-b3cb-fdd4b5d4602c
CSeq: 28651 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX
l:  0


<--- Transmitting SIP response (383 bytes) to UDP:52.89.237.252:1049 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 52.89.237.252:5060;rport=1049;received=52.89.237.252;branch=z9hG4bKPj1af3fe89-618c-4737-9d37-c549cbbdd3b1
i: 27086479-330f-4b1f-b3cb-fdd4b5d4602c
f: <sip:+15155565058@172.31.24.143>;tag=180c636a-771f-466f-908b-81184a8b2801
t: <sip:+13193587700@50.81.142.253>;tag=d0a3706a-8e7a-4bc8-8878-efe2aa36fde7
CSeq: 28651 BYE
Server: Asterisk PBX
l:  0


  == Spawn extension (bfs-in, s, 2) exited non-zero on 'PJSIP/bfs-trunk-00000068'
<--- Transmitting SIP request (420 bytes) to UDP:192.168.2.254:5060 --->
OPTIONS sip:line-bfs-15@192.168.2.254:5060;ob SIP/2.0
v: SIP/2.0/UDP 192.168.2.100:5060;rport;branch=z9hG4bKPjf211eeae-31a1-4cb2-9731-805a578bc3be
f: <sip:line-bfs-15@192.168.2.100>;tag=421a33d9-a6d0-46a5-ac8b-d8d66916a3f2
t: <sip:line-bfs-15@192.168.2.254;ob>
m: <sip:line-bfs-15@192.168.2.100:5060>
i: 8dc0f823-e8cd-4918-92c0-295d47da248a
CSeq: 6452 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX
l:  0

jcolp · November 30, 2020, 4:44pm

DPMA does not require certified Asterisk. It is supported on normal Asterisk releases. As well, current Asterisk 16 has a change which will stop the re-INVITE from being sent in the first place.

david551 · November 30, 2020, 4:47pm

The initial client has dropped the call when re-invited to add additional codecs (8 and 9).

I’d have to assume the diagnostic is wrong.

Although people talk about negotiating with SDP, codecs aren’t actually negotiated, and it is legitimate to include codecs that weren’t in the initial exchange; one doesn’t even have to match the current offer.

I’ll lave it to PJSIP experts to say whether there is an option to prevent re-invites with additional codecs, or whether that feature wasn’t actually intended, but the correct response of the client is to respond with just codec 0, and not to try to send with 8 or 9.

georgequalley · November 30, 2020, 5:00pm

I was not aware of that. I was under the impression that Certified was required for DPMA. Thank you for clarifying that for me.

georgequalley · November 30, 2020, 5:03pm

What’s confusing to me is that there’s no problem negotiating the correct codec when I’m connecting the call on Server A only. It’s only when the call is bridged between Server A and Server B (which supports the same codecs) that the renegotiation happens and the call drops.

georgequalley · December 1, 2020, 5:10am

Upgrading to Asterisk 18.1.0 did fix this issue.

system · December 31, 2020, 5:10am

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