Asterisk 18, TLS NAT calls have no video but auido , local udp calls have bothe viedo and audio

dmsherazi · March 28, 2022, 1:07pm

Hi ,

First of all I would appoliogise for being not able to debug the sip flows from history to debug my issue.
I have a setup where one of the device registered as 141 is on a 4G cellular network. when my dialplan is activated I place a call to local devices and after a delay of about 6 seconds dial the 141 devices.

If I pick the call at 141 then there is two way audio but now video . Can someone help in debugging why is there no audio

Below are the logs

    -- Called Local/mobilephones@default
    -- Executing [mobilephones@default:1] Ringing("Local/mobilephones@default-00000000;2", "") in new stack
    -- Local/mobilephones@default-00000000;1 is ringing
<--- Transmitting SIP response (473 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK494062518
Call-ID: 1574771622
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=761291098
To: <sip:601@192.168.1.17>;tag=794e99f9-d347-4408-9d03-3b2d8606dd67
CSeq: 21 INVITE
Server: SHAULA-001(7.4.0)
Contact: <sip:192.168.1.17:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


    -- Executing [mobilephones@default:2] System("Local/mobilephones@default-00000000;2", "/bin/sleep 6") in new stack
    -- Executing [mobilephones@default:3] Dial("Local/mobilephones@default-00000000;2", "&PJSIP/141/sip:141@188.66.182.161:60760;transport=TLS;x-ast-orig-host=10.184.189.62:46450") in new stack
    -- Called PJSIP/141/sip:141@188.66.182.161:60760;transport=TLS;x-ast-orig-host=10.184.189.62:46450
<--- Transmitting SIP request (727 bytes) to TLS:188.66.182.161:60760 --->
INVITE sip:141@188.66.182.161:60760;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 82.178.154.154:5061;rport;branch=z9hG4bKPjfc09c0ed-dc87-4c8a-a3a9-f5ceb2173884;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=9c2c56c3-28c2-4fc6-963c-bd16a982f03d
To: <sip:141@188.66.182.161>
Contact: <sip:asterisk@82.178.154.154:5061;transport=TLS>
Call-ID: a62b55a1-4e5b-42ec-8f05-8aa4a5bbde62
CSeq: 20162 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "161" <sip:Door_1@192.168.1.17>
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP response (494 bytes) from TLS:188.66.182.161:60760 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPjfc09c0ed-dc87-4c8a-a3a9-f5ceb2173884;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=9c2c56c3-28c2-4fc6-963c-bd16a982f03d
To: <sip:141@188.66.182.161>
Call-ID: a62b55a1-4e5b-42ec-8f05-8aa4a5bbde62
CSeq: 20162 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (595 bytes) from TLS:188.66.182.161:60760 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPjfc09c0ed-dc87-4c8a-a3a9-f5ceb2173884;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=9c2c56c3-28c2-4fc6-963c-bd16a982f03d
To: <sip:141@188.66.182.161>;tag=12444390
Call-ID: a62b55a1-4e5b-42ec-8f05-8aa4a5bbde62
CSeq: 20162 INVITE
Contact: <sip:141@10.184.189.62:46450;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- PJSIP/141-00000001 is ringing
    -- Local/mobilephones@default-00000000;1 is ringing
<--- Received SIP response (2141 bytes) from TLS:188.66.182.161:60760 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPjfc09c0ed-dc87-4c8a-a3a9-f5ceb2173884;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=9c2c56c3-28c2-4fc6-963c-bd16a982f03d
To: <sip:141@188.66.182.161>;tag=12444390
Call-ID: a62b55a1-4e5b-42ec-8f05-8aa4a5bbde62
CSeq: 20162 INVITE
Contact: <sip:141@10.184.189.62:46450;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length:  1489

v=0
o=141 8000 8000 IN IP4 10.184.189.62
s=SIP Call
c=IN IP4 10.184.189.62
t=0 0
m=audio 16758 RTP/SAVP 0 8 9 123 2 97 3 18 101
a=sendrecv
a=rtcp:16759 IN IP4 10.184.189.62
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap       > 0x17cc610 -- Strict RTP learning after remote address set to: 10.184.189.62:16758
       > 0x179ae20 -- Strict RTP learning after remote address set to: 10.184.189.62:37218
<--- Transmitting SIP request (1355 bytes) to TLS:188.66.182.161:60760 --->
ACK sip:141@188.66.182.161:60760;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 82.178.154.154:5061;rport;branch=z9hG4bKPjfa7af291-22de-4131-9d68-eff9713ac66b;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=9c2c56c3-28c2-4fc6-963c-bd16a982f03d
To: <sip:141@188.66.182.161>;tag=12444390
Call-ID: a62b55a1-4e5b-42ec-8f05-8aa4a5bbde62
CSeq: 20162 ACK
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Type: application/sdp
Content-Length:   904

v=0
o=- 8000 8002 IN IP4 82.178.154.154
s=Asterisk
c=IN IP4 82.178.154.154
t=0 0
a=msid-semantic:WMS *
m=audio 37220 RTP/SAVP 0
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 12:17:EF:77:F6:9D:57:03:42:0B:17:CD:C6:85:2E:4E:1A:E2:50:FD:04:7F:85:FA:0D:DA:9B:DB:D7:00:99:6A
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:20
a=sendrecv
a=msid:38ccff69-7f29-4de0-adb8-a73746d96821 deab48f6-c433-4660-b89f-dffe394caabd
a=rtcp-fb:* transport-cc
m=video 3    -- PJSIP/141-00000001 answered Local/mobilephones@default-00000000;2
    -- Local/mobilephones@default-00000000;1 answered PJSIP/161-00000000
    -- Channel PJSIP/141-00000001 joined 'simple_bridge' basic-bridge <83953fc1-2842-4163-b61d-baa9a08acbfc>
    -- Channel Local/mobilephones@default-00000000;2 joined 'simple_bridge' basic-bridge <83953fc1-2842-4163-b61d-baa9a08acbfc>
       > 0x17f0c80 -- Strict RTP learning after remote address set to: 82.178.154.154:6000
       > 0x177c310 -- Strict RTP learning after remote address set to: 82.178.154.154:6200
<--- Transmitting SIP response (992 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK494062518
Call-ID: 1574771622
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=761291098
To: <sip:601@192.168.1.17>;tag=794e99f9-d347-4408-9d03-3b2d8606dd67
CSeq: 21 INVITE
Server: SHAULA-001(7.4.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.17:5060>
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: "141" <sip:MobileExten141@192.168.1.17>
Content-Type: application/sdp
Content-Length:   380

v=0
o=- 700760678 700760680 IN IP4 192.168.1.17
s=Asterisk
c=IN IP4 192.168.1.17
t=0 0
m=audio 33190 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 39348 RTP/AVP 102
a=rtpmap:102 H264/900    -- Channel Local/mobilephones@default-00000000;1 joined 'simple_bridge' basic-bridge <39daf2de-cfa7-4dd2-891f-673b38781f6b>
    -- Channel PJSIP/161-00000000 joined 'simple_bridge' basic-bridge <39daf2de-cfa7-4dd2-891f-673b38781f6b>
<--- Received SIP request (369 bytes) from UDP:192.168.1.163:5060 --->
ACK sip:192.168.1.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1224752597
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=761291098
To: <sip:601@192.168.1.17>;tag=794e99f9-d347-4408-9d03-3b2d8606dd67
Call-ID: 1574771622
CSeq: 21 ACK
Contact: <sip:161@192.168.1.163:5060>
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Received SIP request (801 bytes) from TLS:188.66.182.161:60760 --->
INFO sip:asterisk@82.178.154.154:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 10.184.189.62:46450;branch=z9hG4bK1048656949;rport
From: <sip:141@188.66.182.161>;tag=12444390
To: <sip:Door_1@192.168.1.17>;tag=9c2c56c3-28c2-4fc6-963c-bd16a982f03d
Call-ID: a62b55a1-4e5b-42ec-8f05-8aa4a5bbde62
CSeq: 20163 INFO
Contact: <sip:141@10.184.189.62:46450;transport=tls>
Max-Forwards: 70
Supported: replaces, path, timer, 100rel, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/media_control+xml
Content-Length:   164

<?xml version="1.0" encoding="utf-8" ?><media_control>  <vc_primitive>    <to_encoder>      <picture_fast_update/>    </to_encoder>  </vc_primitive></media_control>
<--- Transmitting SIP request (597 bytes) to UDP:192.168.1.163:5060 --->
INFO sip:161@192.168.1.163:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPje04d48ca-e1e5-4d5b-9362-714af8b292be
From: <sip:601@192.168.1.17>;tag=794e99f9-d347-4408-9d03-3b2d8606dd67
To: "F-1-1-001" <sip:161@192.168.1.17>;tag=761291098
Call-ID: 1574771622
CSeq: 32672 INFO
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>

<--- Transmitting SIP response (346 bytes) to TLS:188.66.182.161:60760 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.184.189.62:46450;rport=60760;received=188.66.182.161;branch=z9hG4bK1048656949
Call-ID: a62b55a1-4e5b-42ec-8f05-8aa4a5bbde62
From: <sip:141@188.66.182.161>;tag=12444390
To: <sip:Door_1@192.168.1.17>;tag=9c2c56c3-28c2-4fc6-963c-bd16a982f03d
CSeq: 20163 INFO
Server: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP response (368 bytes) from UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPje04d48ca-e1e5-4d5b-9362-714af8b292be
From: <sip:601@192.168.1.17>;tag=794e99f9-d347-4408-9d03-3b2d8606dd67
To: "F-1-1-001" <sip:161@192.168.1.17>;tag=761291098
Call-ID: 1574771622
CSeq: 32672 INFO
Contact: <sip:161@192.168.1.163:5060>
User-Agent: DnakeVoip v1.0
Content-Length: 0


       > 0x177c310 -- Strict RTP qualifying stream type: video
       > 0x177c310 -- Strict RTP switching source address to 192.168.1.163:6200
       > Move-swap optimizing Local/mobilephones@default-00000000;2 <-- PJSIP/161-00000000.
    -- Channel PJSIP/161-00000000 left 'simple_bridge' basic-bridge <39daf2de-cfa7-4dd2-891f-673b38781f6b>
    -- Channel Local/mobilephones@default-00000000;2 left 'simple_bridge' basic-bridge <83953fc1-2842-4163-b61d-baa9a08acbfc>
    -- Channel PJSIP/161-00000000 swapped with Local/mobilephones@default-00000000;2 into 'simple_bridge' basic-bridge <83953fc1-2842-4163-b61d-baa9a08acbfc>
    -- Channel Local/mobilephones@default-00000000;1 left 'simple_bridge' basic-bridge <39daf2de-cfa7-4dd2-891f-673b38781f6b>
  == Spawn extension (default, mobilephones, 3) exited non-zero on 'Local/mobilephones@default-00000000;2'
       > 0x17f0c80 -- Strict RTP qualifying stream type: audio
       > 0x17f0c80 -- Strict RTP switching source address to 192.168.1.163:6000
       > 0x17cc610 -- Strict RTP qualifying stream type: audio
       > 0x17cc610 -- Strict RTP switching source address to 188.66.182.161:60761
       > 0x179ae20 -- Strict RTP qualifying stream type: video
       > 0x179ae20 -- Strict RTP switching source address to 188.66.182.161:60763
       > 0x17cc610 -- Strict RTP learning complete - Locking on source address 188.66.182.161:60761
       > 0x177c310 -- Strict RTP learning complete - Locking on source address 192.168.1.163:6200
       > 0x179ae20 -- Strict RTP learning complete - Locking on source address 188.66.182.161:60763
       > 0x17f0c80 -- Strict RTP learning complete - Locking on source address 192.168.1.163:6000
<--- Received SIP request (586 bytes) from TLS:188.66.182.161:60760 --->
BYE sip:asterisk@82.178.154.154:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 10.184.189.62:46450;branch=z9hG4bK1621882673;rport
From: <sip:141@188.66.182.161>;tag=12444390
To: <sip:Door_1@192.168.1.17>;tag=9c2c56c3-28c2-4fc6-963c-bd16a982f03d
Call-ID: a62b55a1-4e5b-42ec-8f05-8aa4a5bbde62
CSeq: 20164 BYE
Contact: <sip:141@10.184.189.62:46450;transport=tls>
Max-Forwards: 70
Supported: replaces, path, timer, 100rel, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (345 bytes) to TLS:188.66.182.161:60760 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.184.189.62:46450;rport=60760;received=188.66.182.161;branch=z9hG4bK1621882673
Call-ID: a62b55a1-4e5b-42ec-8f05-8aa4a5bbde62
From: <sip:141@188.66.182.161>;tag=12444390
To: <sip:Door_1@192.168.1.17>;tag=9c2c56c3-28c2-4fc6-963c-bd16a982f03d
CSeq: 20164 BYE
Server: SHAULA-001(7.4.0)
Content-Length:  0


    -- Channel PJSIP/141-00000001 left 'simple_bridge' basic-bridge <83953fc1-2842-4163-b61d-baa9a08acbfc>
    -- Channel PJSIP/161-00000000 left 'simple_bridge' basic-bridge <83953fc1-2842-4163-b61d-baa9a08acbfc>
  == Spawn extension (fullrights, 601, 3) exited non-zero on 'PJSIP/161-00000000'
<--- Transmitting SIP request (393 bytes) to UDP:192.168.1.163:5060 --->
BYE sip:161@192.168.1.163:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPj15af4b81-dbb9-4610-87a2-c9e9ce619796
From: <sip:601@192.168.1.17>;tag=794e99f9-d347-4408-9d03-3b2d8606dd67
To: "F-1-1-001" <sip:161@192.168.1.17>;tag=761291098
Call-ID: 1574771622
CSeq: 32673 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP response (328 bytes) from UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPj15af4b81-dbb9-4610-87a2-c9e9ce619796
From: <sip:601@192.168.1.17>;tag=794e99f9-d347-4408-9d03-3b2d8606dd67
To: "F-1-1-001" <sip:161@192.168.1.17>;tag=761291098
Call-ID: 1574771622
CSeq: 32673 BYE
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Received SIP request (300 bytes) from UDP:192.168.1.163:5060 --->
OPTIONS sip:192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK218572494
From: <sip:161@192.168.1.17>;tag=1548433885
To: <sip:192.168.1.17>
Call-ID: 184183705
CSeq: 20 OPTIONS
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (444 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK218572494
Call-ID: 184183705
From: <sip:161@192.168.1.17>;tag=1548433885
To: <sip:192.168.1.17>;tag=z9hG4bK218572494
CSeq: 20 OPTIONS
WWW-Authenticate: Digest realm="asterisk",nonce="1648471715/659906ab1182dbc347d1023d4f485e2c",opaque="060e4e84711d63b0",algorithm=md5,qop="auth"
Server: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP request (562 bytes) from UDP:192.168.1.163:5060 --->
OPTIONS sip:192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK324437058
From: <sip:161@192.168.1.17>;tag=1548433885
To: <sip:192.168.1.17>
Call-ID: 184183705
CSeq: 21 OPTIONS
Authorization: Digest username="161", realm="asterisk", nonce="1648471715/659906ab1182dbc347d1023d4f485e2c", uri="sip:192.168.1.17", response="9bdf4b2b663a3aac49bca837a08c3f3c", algorithm=MD5, cnonce="0a4f113b", opaque="060e4e84711d63b0", qop=auth, nc=00000001
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (773 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK324437058
Call-ID: 184183705
From: <sip:161@192.168.1.17>;tag=1548433885
To: <sip:192.168.1.17>;tag=z9hG4bK324437058
CSeq: 21 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: SHAULA-001(7.4.0)
Content-Length:  0

TheMark · March 28, 2022, 1:16pm

hi witch sub version are you using 18.XX.X
also are you using chan_sip or chan_pjsip

can you try to only allow only 1 sound codec and see if that affect the problem
if yes them you may be affected by this bug 1 ASTERISK-29978
only relevant if you are using chan_sip, to fix upgrade to chan_pjsip

dmsherazi · March 28, 2022, 1:19pm

Asterisk 18.5.1
PJPROJECT version currently running against: 2.10

Pjsip is being used

david551 · March 28, 2022, 1:25pm

This is another Asterisk late offer SDP, which I didn’t think it did. We never came to a conclusion on the last one.

chan_sip definitely never originated late offers, so I doubt this is chan_sip.

Asterisk has done a late offer for video, but the remote party never offered a video stream, so the best you have at the moment is video towards Asterisk, but not away.

For some reason, the ACK has been truncated in the logs, and it also looks like the corresponding A side OK has also been truncated, so I can’t confirm that they had a=sendrecv.

Could we have your configuration please.

Also does anyone know if there has been a change in policy to late offer, or if there are specific circumstances that will result in late offer?

jcolp · March 28, 2022, 1:27pm

There might be some specific configuration combination to result in late SDP when it shouldn’t, it’s not something that is really outright configurable. I don’t know how well the code would fundamentally support it if we did it.

david551 · March 28, 2022, 1:50pm

You haven’t specified any codecs for 141. I think that means allow=all. That is known to cause problems.

dmsherazi · March 28, 2022, 2:01pm

I added the following instead of allow=all

disallow = all
allow = ulaw
allow = alaw
allow = gsm
allow = g726
allow = h264
allow = mpeg4
allow = vp8
allow = h263p

but gain only audio and the gs wave app shows the call has PCMU codec. So I removed some and kept only video and now it says 488 error

disallow = all
allow = h264
allow = mpeg4
allow = vp8
allow = h263p

.c:2604 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
    -- Called Local/mobilephones@default
    -- Executing [mobilephones@default:1] Ringing("Local/mobilephones@default-00000000;2", "") in new stack
    -- Local/mobilephones@default-00000000;1 is ringing
    -- Executing [mobilephones@default:2] System("Local/mobilephones@default-00000000;2", "/bin/sleep 6") in new stack
<--- Transmitting SIP response (474 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1468477880
Call-ID: 1749590592
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=782630731
To: <sip:601@192.168.1.17>;tag=6a64e547-0911-449e-b5f0-896d34fbe8c6
CSeq: 21 INVITE
Server: SHAULA-001(7.4.0)
Contact: <sip:192.168.1.17:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Received SIP request (301 bytes) from UDP:192.168.1.163:5060 --->
OPTIONS sip:192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1728869817
From: <sip:161@192.168.1.17>;tag=923349907
To: <sip:192.168.1.17>
Call-ID: 1726151344
CSeq: 20 OPTIONS
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (446 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1728869817
Call-ID: 1726151344
From: <sip:161@192.168.1.17>;tag=923349907
To: <sip:192.168.1.17>;tag=z9hG4bK1728869817
CSeq: 20 OPTIONS
WWW-Authenticate: Digest realm="asterisk",nonce="1648475886/23bcd790c57a4cd67ba60efc8241b7f0",opaque="3f8f0f063a5a624f",algorithm=md5,qop="auth"
Server: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP request (563 bytes) from UDP:192.168.1.163:5060 --->
OPTIONS sip:192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1869279257
From: <sip:161@192.168.1.17>;tag=923349907
To: <sip:192.168.1.17>
Call-ID: 1726151344
CSeq: 21 OPTIONS
Authorization: Digest username="161", realm="asterisk", nonce="1648475886/23bcd790c57a4cd67ba60efc8241b7f0", uri="sip:192.168.1.17", response="642a8d5d128e967f2c669cbd4fd873f2", algorithm=MD5, cnonce="0a4f113b", opaque="3f8f0f063a5a624f", qop=auth, nc=00000001
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (775 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1869279257
Call-ID: 1726151344
From: <sip:161@192.168.1.17>;tag=923349907
To: <sip:192.168.1.17>;tag=z9hG4bK1869279257
CSeq: 21 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: SHAULA-001(7.4.0)
Content-Length:  0


    -- Executing [mobilephones@default:3] Dial("Local/mobilephones@default-00000000;2", "&PJSIP/141/sip:141@188.66.182.161:60996;transport=TLS;x-ast-orig-host=10.184.189.62:48658") in new stack
    -- Called PJSIP/141/sip:141@188.66.182.161:60996;transport=TLS;x-ast-orig-host=10.184.189.62:48658
<--- Transmitting SIP request (1954 bytes) to TLS:188.66.182.161:60996 --->
INVITE sip:141@188.66.182.161:60996;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 82.178.154.154:5061;rport;branch=z9hG4bKPj6b39063d-b888-481b-9a29-8d117e680d36;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=1929965a-f4a3-4ad4-a54e-f7ef17c38f81
To: <sip:141@188.66.182.161>
Contact: <sip:asterisk@82.178.154.154:5061;transport=TLS>
Call-ID: ba674af9-8622-4bca-bbb2-15a0e807a007
CSeq: 6775 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "161" <sip:Door_1@192.168.1.17>
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Type: application/sdp
Content-Length:  1194

v=0
o=- 842565884 842565884 IN IP4 82.178.154.154
s=Asterisk
c=IN IP4 82.178.154.154
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE video-0
m=video 34552 UDP/<--- Received SIP response (493 bytes) from TLS:188.66.182.161:60996 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPj6b39063d-b888-481b-9a29-8d117e680d36;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=1929965a-f4a3-4ad4-a54e-f7ef17c38f81
To: <sip:141@188.66.182.161>
Call-ID: ba674af9-8622-4bca-bbb2-15a0e807a007
CSeq: 6775 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (624 bytes) from TLS:188.66.182.161:60996 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPj6b39063d-b888-481b-9a29-8d117e680d36;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=1929965a-f4a3-4ad4-a54e-f7ef17c38f81
To: <sip:141@188.66.182.161>;tag=1717747345
Call-ID: ba674af9-8622-4bca-bbb2-15a0e807a007
CSeq: 6775 INVITE
Contact: <sip:141@10.184.189.62:48658;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow-Events: talk, hold
Require: 100rel
RSeq: 100
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (445 bytes) to TLS:188.66.182.161:60996 --->
PRACK sip:141@188.66.182.161:60996;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 82.178.154.154:5061;rport;branch=z9hG4bKPj9f5588cd-5ae8-49f4-988f-099b729a799e;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=1929965a-f4a3-4ad4-a54e-f7ef17c38f81
To: <sip:141@188.66.182.161>;tag=1717747345
Call-ID: ba674af9-8622-4bca-bbb2-15a0e807a007
CSeq: 6776 PRACK
RAck: 100 6775 INVITE
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Length:  0


    -- PJSIP/141-00000001 is ringing
    -- Local/mobilephones@default-00000000;1 is ringing
<--- Received SIP response (564 bytes) from TLS:188.66.182.161:60996 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPj9f5588cd-5ae8-49f4-988f-099b729a799e;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=1929965a-f4a3-4ad4-a54e-f7ef17c38f81
To: <sip:141@188.66.182.161>;tag=1717747345
Call-ID: ba674af9-8622-4bca-bbb2-15a0e807a007
CSeq: 6776 PRACK
Contact: <sip:141@10.184.189.62:48658;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (572 bytes) from TLS:188.66.182.161:60996 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPj6b39063d-b888-481b-9a29-8d117e680d36;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=1929965a-f4a3-4ad4-a54e-f7ef17c38f81
To: <sip:141@188.66.182.161>;tag=1717747345
Call-ID: ba674af9-8622-4bca-bbb2-15a0e807a007
CSeq: 6775 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Warning: 304 GS "Media type not available"
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (418 bytes) to TLS:188.66.182.161:60996 --->
ACK sip:141@188.66.182.161:60996;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 82.178.154.154:5061;rport;branch=z9hG4bKPj6b39063d-b888-481b-9a29-8d117e680d36;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=1929965a-f4a3-4ad4-a54e-f7ef17c38f81
To: <sip:141@188.66.182.161>;tag=1717747345
Call-ID: ba674af9-8622-4bca-bbb2-15a0e807a007
CSeq: 6775 ACK
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [mobilephones@default:4] Hangup("Local/mobilephones@default-00000000;2", "") in new stack
  == Spawn extension (default, mobilephones, 4) exited non-zero on 'Local/mobilephones@default-00000000;2'
  == Everyone is busy/congested at this time (25:0/0/25)
    -- Executing [601@fullrights:4] Hangup("PJSIP/161-00000000", "") in new stack
  == Spawn extension (fullrights, 601, 4) exited non-zero on 'PJSIP/161-00000000'
<--- Transmitting SIP response (538 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1468477880
Call-ID: 1749590592
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=782630731
To: <sip:601@192.168.1.17>;tag=6a64e547-0911-449e-b5f0-896d34fbe8c6
CSeq: 21 INVITE
Server: SHAULA-001(7.4.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Reason: Q.850;cause=58
P-Asserted-Identity: "141" <sip:MobileExten141@192.168.1.17>
Content-Length:  0


<--- Received SIP request (313 bytes) from UDP:192.168.1.163:5060 --->
ACK sip:601@192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1468477880
Route: <sip:192.168.1.17;lr>
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=782630731
To: <sip:601@192.168.1.17>;tag=6a64e547-0911-449e-b5f0-896d34fbe8c6
Call-ID: 1749590592
CSeq: 21 ACK
Content-Length: 0


<--- Received SIP request (577 bytes) from TLS:82.178.154.154:60428 --->
REGISTER sip:Kindows.ddns.net:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.242:60428;branch=z9hG4bK1229525322;rport;alias
From: <sip:141@Kindows.ddns.net:5061>;tag=24873490
To: <sip:141@Kindows.ddns.net:5061>
Call-ID: 596385609-29210-1@BJC.BGI.B.CCJ
CSeq: 2198 REGISTER
Contact: <sip:141@192.168.1.242:60428;transport=tls>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B8227D179>";expires=0
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (490 bytes) to TLS:82.178.154.154:60428 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.1.242:60428;rport=60428;received=82.178.154.154;branch=z9hG4bK1229525322;alias
Call-ID: 596385609-29210-1@BJC.BGI.B.CCJ
From: <sip:141@Kindows.ddns.net>;tag=24873490
To: <sip:141@Kindows.ddns.net>;tag=z9hG4bK1229525322
CSeq: 2198 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1648475897/ad08b6d5bdfbb7fd693030976f72845a",opaque="30f78fa47240bb08",algorithm=md5,qop="auth"
Server: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP request (847 bytes) from TLS:82.178.154.154:60428 --->
REGISTER sip:Kindows.ddns.net:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.242:60428;branch=z9hG4bK914669859;rport;alias
From: <sip:141@Kindows.ddns.net:5061>;tag=24873490
To: <sip:141@Kindows.ddns.net:5061>
Call-ID: 596385609-29210-1@BJC.BGI.B.CCJ
CSeq: 2199 REGISTER
Contact: <sip:141@192.168.1.242:60428;transport=tls>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B8227D179>";expires=0
Authorization: Digest username="141", realm="asterisk", nonce="1648475897/ad08b6d5bdfbb7fd693030976f72845a", uri="sip:Kindows.ddns.net:5061", response="5612b80385c871e0e6066695d412ecbc", algorithm=md5, cnonce="13877890", opaque="30f78fa47240bb08", qop=auth, nc=00000009
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- Attempted to remove non-existent contact 'sip:141@82.178.154.154:60428;transport=TLS;x-ast-orig-host=192.168.1.242:60428' from AOR '141' by request
<--- Transmitting SIP response (436 bytes) to TLS:82.178.154.154:60428 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.1.242:60428;rport=60428;received=82.178.154.154;branch=z9hG4bK914669859;alias
Call-ID: 596385609-29210-1@BJC.BGI.B.CCJ
From: <sip:141@Kindows.ddns.net>;tag=24873490
To: <sip:141@Kindows.ddns.net>;tag=z9hG4bK914669859
CSeq: 2199 REGISTER
Date: Mon, 28 Mar 2022 13:58:17 GMT
Contact: <sip:141@10.184.189.62:48658;transport=TLS>;expires=3580
Server: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP request (869 bytes) from TLS:82.178.154.154:60428 --->
REGISTER sip:Kindows.ddns.net:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.242:60428;branch=z9hG4bK722505944;rport;alias
From: <sip:141@Kindows.ddns.net:5061>;tag=24873490
To: <sip:141@Kindows.ddns.net:5061>
Call-ID: 596385609-29210-1@BJC.BGI.B.CCJ
CSeq: 2200 REGISTER
Contact: <sip:141@192.168.1.242:60428;transport=tls>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B8227D179>"
Authorization: Digest username="141", realm="asterisk", nonce="1648475897/ad08b6d5bdfbb7fd693030976f72845a", uri="sip:Kindows.ddns.net:5061", response="4fc7c95e85a34d0c1e62bb698aa27cb4", algorithm=md5, cnonce="05318394", opaque="30f78fa47240bb08", qop=auth, nc=0000000a
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.34
Supported: path
Expires: 3600
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- Added contact 'sip:141@82.178.154.154:60428;transport=TLS;x-ast-orig-host=192.168.1.242:60428' to AOR '141' with expiration of 3600 seconds
<--- Transmitting SIP response (518 bytes) to TLS:82.178.154.154:60428 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.1.242:60428;rport=60428;received=82.178.154.154;branch=z9hG4bK722505944;alias
Call-ID: 596385609-29210-1@BJC.BGI.B.CCJ
From: <sip:141@Kindows.ddns.net>;tag=24873490
To: <sip:141@Kindows.ddns.net>;tag=z9hG4bK722505944
CSeq: 2200 REGISTER
Date: Mon, 28 Mar 2022 13:58:17 GMT
Contact: <sip:141@10.184.189.62:48658;transport=TLS>;expires=3580
Contact: <sip:141@192.168.1.242:60428;transport=TLS>;expires=3599
Expires: 3600
Server: SHAULA-001(7.4.0)
Content-Length:  0


<--- Transmitting SIP request (457 bytes) to TLS:82.178.154.154:60428 --->
OPTIONS sip:141@82.178.154.154:60428;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 82.178.154.154:5061;rport;branch=z9hG4bKPj19b63400-8799-47b2-8507-d167086f381d;alias
From: <sip:141@192.168.1.17>;tag=6ff2566c-d812-4a8b-8277-ab6917b8f613
To: <sip:141@82.178.154.154>
Contact: <sip:141@82.178.154.154:5061;transport=TLS>
Call-ID: 9b989594-7c63-491d-a006-783c36953e53
CSeq: 64782 OPTIONS
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Length:  0


<--- Transmitting SIP request (456 bytes) to TLS:188.66.182.161:60996 --->
OPTIONS sip:141@188.66.182.161:60996;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 82.178.154.154:5061;rport;branch=z9hG4bKPjefeafffb-fa56-4500-824c-e63b42d81867;alias
From: <sip:141@192.168.1.17>;tag=caf224a0-ffa2-4cb2-bca4-1b9417402c64
To: <sip:141@188.66.182.161>
Contact: <sip:141@82.178.154.154:5061;transport=TLS>
Call-ID: f6e9adf0-b3a8-4f23-91c2-c85b82f28a0c
CSeq: 3570 OPTIONS
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP response (497 bytes) from TLS:82.178.154.154:60428 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPj19b63400-8799-47b2-8507-d167086f381d;alias
From: <sip:141@192.168.1.17>;tag=6ff2566c-d812-4a8b-8277-ab6917b8f613
To: <sip:141@82.178.154.154>;tag=1596163976
Call-ID: 9b989594-7c63-491d-a006-783c36953e53
CSeq: 64782 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- Contact 141/sip:141@82.178.154.154:60428;transport=TLS;x-ast-orig-host=192.168.1.242:60428 is now Reachable.  RTT: 47.312 msec
<--- Received SIP response (496 bytes) from TLS:188.66.182.161:60996 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPjefeafffb-fa56-4500-824c-e63b42d81867;alias
From: <sip:141@192.168.1.17>;tag=caf224a0-ffa2-4cb2-bca4-1b9417402c64
To: <sip:141@188.66.182.161>;tag=1470897389
Call-ID: f6e9adf0-b3a8-4f23-91c2-c85b82f28a0c
CSeq: 3570 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


[Mar 28 13:58:17] NOTICE[14263]: res_pjsip/pjsip_transport_management.c:170 idle_sched_cb: Shutting down transport 'TLS to 188.135.92.52:47709' since no request was received in 32 seconds
<--- Received SIP request (302 bytes) from UDP:192.168.1.163:5060 --->
OPTIONS sip:192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1248808894
From: <sip:161@192.168.1.17>;tag=1054043002
To: <sip:192.168.1.17>
Call-ID: 1940863734
CSeq: 20 OPTIONS
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (447 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1248808894
Call-ID: 1940863734
From: <sip:161@192.168.1.17>;tag=1054043002
To: <sip:192.168.1.17>;tag=z9hG4bK1248808894
CSeq: 20 OPTIONS
WWW-Authenticate: Digest realm="asterisk",nonce="1648475915/e5f93ff7857452bb02078242df281ce9",opaque="68ff350f253cd3ab",algorithm=md5,qop="auth"
Server: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP request (564 bytes) from UDP:192.168.1.163:5060 --->
OPTIONS sip:192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1053174873
From: <sip:161@192.168.1.17>;tag=1054043002
To: <sip:192.168.1.17>
Call-ID: 1940863734
CSeq: 21 OPTIONS
Authorization: Digest username="161", realm="asterisk", nonce="1648475915/e5f93ff7857452bb02078242df281ce9", uri="sip:192.168.1.17", response="d3eb149f6d3d5898867654ed3c5d3c60", algorithm=MD5, cnonce="0a4f113b", opaque="68ff350f253cd3ab", qop=auth, nc=00000001
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (776 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1053174873
Call-ID: 1940863734
From: <sip:161@192.168.1.17>;tag=1054043002
To: <sip:192.168.1.17>;tag=z9hG4bK1053174873
CSeq: 21 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: SHAULA-001(7.4.0)
Content-Length:  0


<--- Transmitting SIP request (457 bytes) to UDP:192.168.1.163:5060 --->
OPTIONS sip:161@192.168.1.163:5060;line=e63b8695feffcc4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPjfbf2a581-d2d5-411f-b8e4-4117f4f46d2e
From: <sip:161@192.168.1.17>;tag=f3229aff-dc58-4318-9cef-55ea2e1d8041
To: <sip:161@192.168.1.163;line=e63b8695feffcc4>
Contact: <sip:161@192.168.1.17:5060>
Call-ID: c401855d-cdac-4659-b6d6-c6914340b76c
CSeq: 8999 OPTIONS
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP response (367 bytes) from UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPjfbf2a581-d2d5-411f-b8e4-4117f4f46d2e
From: <sip:161@192.168.1.17>;tag=f3229aff-dc58-4318-9cef-55ea2e1d8041
To: <sip:161@192.168.1.163;line=e63b8695feffcc4>;tag=993864417
Call-ID: c401855d-cdac-4659-b6d6-c6914340b76c
CSeq: 8999 OPTIONS
User-Agent: DnakeVoip v1.0
Content-Length: 0

david551 · March 28, 2022, 2:06pm

Asterisk requires an audio stream.

You now have the, normal, early offer.

You didn’t provide a log for the one with both audio and video codecs.

I thought that WebRTC required Opus, but you don’t have it.

dmsherazi · March 28, 2022, 2:09pm

so now my endpoint is as follows

[141]
type = endpoint
rewrite_contact=yes
context=mobile
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = alaw
allow = gsm
allow = g726
allow = h264
allow = mpeg4
allow = vp8
allow = h263p
rtp_timeout = 30
timers = yes
direct_media = no
callerid=141 <Mobile Exten 141>
send_pai = yes
auth = 141
outbound_auth = 141
aors = 141
dtmf_mode=inband
media_encryption=sdes
transport = transport-tls
rtp_symmetric=yes
force_rport=yes
dtls_auto_generate_cert=yes
webrtc=yes
; Setting webrtc=yes is a shortcut for setting the following options:
use_avpf=yes
; media_encryption=dtls
; dtls_verify=fingerprint
; dtls_setup=actpass
; ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes

and still now video …

and here are the cli logs

 Executing [mobilephones@default:1] Ringing("Local/mobilephones@default-00000000;2", "") in new stack
    -- Local/mobilephones@default-00000000;1 is ringing
    -- Executing [mobilephones@default:2] System("Local/mobilephones@default-00000000;2", "/bin/sleep 6") in new stack
<--- Transmitting SIP response (473 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK801094803
Call-ID: 994379434
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=1664394079
To: <sip:601@192.168.1.17>;tag=7029b327-09cc-4fb1-be25-ec2655be0e2b
CSeq: 21 INVITE
Server: SHAULA-001(7.4.0)
Contact: <sip:192.168.1.17:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


    -- Removed contact 'sip:141@82.178.154.154:34258;transport=TLS;x-ast-orig-host=192.168.1.242:34258' from AOR '141' due to shutdown
  == Contact 141/sip:141@82.178.154.154:34258;transport=TLS;x-ast-orig-host=192.168.1.242:34258 has been deleted
<--- Transmitting SIP request (457 bytes) to UDP:192.168.1.163:5060 --->
OPTIONS sip:161@192.168.1.163:5060;line=e63b8695feffcc4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPj9533b9dc-2903-4060-a79d-462242188b92
From: <sip:161@192.168.1.17>;tag=43e2306f-3b13-49ef-80e6-0392531f5f77
To: <sip:161@192.168.1.163;line=e63b8695feffcc4>
Contact: <sip:161@192.168.1.17:5060>
Call-ID: 4de634d9-ec84-44c6-9189-b33030d96559
CSeq: 9783 OPTIONS
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP response (368 bytes) from UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPj9533b9dc-2903-4060-a79d-462242188b92
From: <sip:161@192.168.1.17>;tag=43e2306f-3b13-49ef-80e6-0392531f5f77
To: <sip:161@192.168.1.163;line=e63b8695feffcc4>;tag=1108843322
Call-ID: 4de634d9-ec84-44c6-9189-b33030d96559
CSeq: 9783 OPTIONS
User-Agent: DnakeVoip v1.0
Content-Length: 0


    -- Executing [mobilephones@default:3] Dial("Local/mobilephones@default-00000000;2", "&PJSIP/141/sip:141@82.178.154.154:38380;transport=TLS;x-ast-orig-host=192.168.1.190:38380") in new stack
    -- Called PJSIP/141/sip:141@82.178.154.154:38380;transport=TLS;x-ast-orig-host=192.168.1.190:38380
<--- Transmitting SIP request (2508 bytes) to TLS:82.178.154.154:38380 --->
INVITE sip:141@82.178.154.154:38380;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 82.178.154.154:5061;rport;branch=z9hG4bKPja0eddc70-7624-4107-a946-0078f967668a;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=3ebf8c05-66b7-48ca-923f-9b4d1b443114
To: <sip:141@82.178.154.154>
Contact: <sip:asterisk@82.178.154.154:5061;transport=TLS>
Call-ID: 56a6c131-795f-4862-9698-b938484c4bf3
CSeq: 19146 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "161" <sip:Door_1@192.168.1.17>
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Type: application/sdp
Content-Length:  1747

v=0
o=- 969310959 969310959 IN IP4 82.178.154.154
s=Asterisk
c=IN IP4 82.178.154.154
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0 video-1
m=audio 3<--- Received SIP response (494 bytes) from TLS:82.178.154.154:38380 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPja0eddc70-7624-4107-a946-0078f967668a;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=3ebf8c05-66b7-48ca-923f-9b4d1b443114
To: <sip:141@82.178.154.154>
Call-ID: 56a6c131-795f-4862-9698-b938484c4bf3
CSeq: 19146 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (625 bytes) from TLS:82.178.154.154:38380 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPja0eddc70-7624-4107-a946-0078f967668a;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=3ebf8c05-66b7-48ca-923f-9b4d1b443114
To: <sip:141@82.178.154.154>;tag=1264300688
Call-ID: 56a6c131-795f-4862-9698-b938484c4bf3
CSeq: 19146 INVITE
Contact: <sip:141@192.168.1.190:38380;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow-Events: talk, hold
Require: 100rel
RSeq: 100
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (447 bytes) to TLS:82.178.154.154:38380 --->
PRACK sip:141@82.178.154.154:38380;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 82.178.154.154:5061;rport;branch=z9hG4bKPj3be9d359-7a24-40e6-ae28-66424021f6dc;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=3ebf8c05-66b7-48ca-923f-9b4d1b443114
To: <sip:141@82.178.154.154>;tag=1264300688
Call-ID: 56a6c131-795f-4862-9698-b938484c4bf3
CSeq: 19147 PRACK
RAck: 100 19146 INVITE
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Length:  0


    -- PJSIP/141-00000001 is ringing
    -- Local/mobilephones@default-00000000;1 is ringing
<--- Received SIP response (565 bytes) from TLS:82.178.154.154:38380 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPj3be9d359-7a24-40e6-ae28-66424021f6dc;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=3ebf8c05-66b7-48ca-923f-9b4d1b443114
To: <sip:141@82.178.154.154>;tag=1264300688
Call-ID: 56a6c131-795f-4862-9698-b938484c4bf3
CSeq: 19147 PRACK
Contact: <sip:141@192.168.1.190:38380;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (1068 bytes) from TLS:82.178.154.154:38380 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPja0eddc70-7624-4107-a946-0078f967668a;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=3ebf8c05-66b7-48ca-923f-9b4d1b443114
To: <sip:141@82.178.154.154>;tag=1264300688
Call-ID: 56a6c131-795f-4862-9698-b938484c4bf3
CSeq: 19146 INVITE
Contact: <sip:141@192.168.1.190:38380;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length:   414

v=0
o=141 8000 8000 IN IP4 192.168.1.190
s=SIP Call
c=IN IP4 192.168.1.190
t=0 0
m=audio 47572 RTP/AVP 0 8 3
a=sendrecv
a=rtcp:47573 IN IP4 192.168.1.190
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
m=video 48336 RTP/AVP 99
       > 0x85c210 -- Strict RTP learning after remote address set to: 192.168.1.190:47572
       > 0x85ecb0 -- Strict RTP learning after remote address set to: 192.168.1.190:48336
<--- Transmitting SIP request (419 bytes) to TLS:82.178.154.154:38380 --->
ACK sip:141@82.178.154.154:38380;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 82.178.154.154:5061;rport;branch=z9hG4bKPj3851ece0-de97-4c86-9ef2-1731c01f2f6c;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=3ebf8c05-66b7-48ca-923f-9b4d1b443114
To: <sip:141@82.178.154.154>;tag=1264300688
Call-ID: 56a6c131-795f-4862-9698-b938484c4bf3
CSeq: 19146 ACK
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Length:  0


    -- PJSIP/141-00000001 answered Local/mobilephones@default-00000000;2
    -- Local/mobilephones@default-00000000;1 answered PJSIP/161-00000000
    -- Channel PJSIP/141-00000001 joined 'simple_bridge' basic-bridge <6cee2c86-f395-4f8d-86ac-1814f373ce47>
    -- Channel Local/mobilephones@default-00000000;2 joined 'simple_bridge' basic-bridge <6cee2c86-f395-4f8d-86ac-1814f373ce47>
       > 0x596310 -- Strict RTP learning after remote address set to: 82.178.154.154:6000
       > 0x859770 -- Strict RTP learning after remote address set to: 82.178.154.154:6200
<--- Transmitting SIP response (994 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK801094803
Call-ID: 994379434
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=1664394079
To: <sip:601@192.168.1.17>;tag=7029b327-09cc-4fb1-be25-ec2655be0e2b
CSeq: 21 INVITE
Server: SHAULA-001(7.4.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.17:5060>
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: "141" <sip:MobileExten141@192.168.1.17>
Content-Type: application/sdp
Content-Length:   382

v=0
o=- 1053206910 1053206912 IN IP4 192.168.1.17
s=Asterisk
c=IN IP4 192.168.1.17
t=0 0
m=audio 34420 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 39064 RTP/AVP 102
a=rtpmap:102 H264/9    -- Channel Local/mobilephones@default-00000000;1 joined 'simple_bridge' basic-bridge <e7e3c096-1c31-4c13-88fa-575563723886>
    -- Channel PJSIP/161-00000000 joined 'simple_bridge' basic-bridge <e7e3c096-1c31-4c13-88fa-575563723886>
<--- Received SIP request (802 bytes) from TLS:82.178.154.154:38380 --->
INFO sip:asterisk@82.178.154.154:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.1.190:38380;branch=z9hG4bK956965339;rport
From: <sip:141@82.178.154.154>;tag=1264300688
To: <sip:Door_1@192.168.1.17>;tag=3ebf8c05-66b7-48ca-923f-9b4d1b443114
Call-ID: 56a6c131-795f-4862-9698-b938484c4bf3
CSeq: 19147 INFO
Contact: <sip:141@192.168.1.190:38380;transport=tls>
Max-Forwards: 70
Supported: replaces, path, timer, 100rel, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/media_control+xml
Content-Length:   164

<?xml version="1.0" encoding="utf-8" ?><media_control>  <vc_primitive>    <to_encoder>      <picture_fast_update/>    </to_encoder>  </vc_primitive></media_control>
<--- Transmitting SIP response (347 bytes) to TLS:82.178.154.154:38380 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.1.190:38380;rport=38380;received=82.178.154.154;branch=z9hG4bK956965339
Call-ID: 56a6c131-795f-4862-9698-b938484c4bf3
From: <sip:141@82.178.154.154>;tag=1264300688
To: <sip:Door_1@192.168.1.17>;tag=3ebf8c05-66b7-48ca-923f-9b4d1b443114
CSeq: 19147 INFO
Server: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP request (368 bytes) from UDP:192.168.1.163:5060 --->
ACK sip:192.168.1.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK429057899
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=1664394079
To: <sip:601@192.168.1.17>;tag=7029b327-09cc-4fb1-be25-ec2655be0e2b
Call-ID: 994379434
CSeq: 21 ACK
Contact: <sip:161@192.168.1.163:5060>
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP request (597 bytes) to UDP:192.168.1.163:5060 --->
INFO sip:161@192.168.1.163:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPjca620fff-a440-42af-8d18-dfda31e90aa2
From: <sip:601@192.168.1.17>;tag=7029b327-09cc-4fb1-be25-ec2655be0e2b
To: "F-1-1-001" <sip:161@192.168.1.17>;tag=1664394079
Call-ID: 994379434
CSeq: 13326 INFO
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>

       > 0x859770 -- Strict RTP qualifying stream type: video
<--- Received SIP response (368 bytes) from UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPjca620fff-a440-42af-8d18-dfda31e90aa2
From: <sip:601@192.168.1.17>;tag=7029b327-09cc-4fb1-be25-ec2655be0e2b
To: "F-1-1-001" <sip:161@192.168.1.17>;tag=1664394079
Call-ID: 994379434
CSeq: 13326 INFO
Contact: <sip:161@192.168.1.163:5060>
User-Agent: DnakeVoip v1.0
Content-Length: 0


       > 0x859770 -- Strict RTP switching source address to 192.168.1.163:6200
       > Move-swap optimizing Local/mobilephones@default-00000000;2 <-- PJSIP/161-00000000.
    -- Channel PJSIP/161-00000000 left 'simple_bridge' basic-bridge <e7e3c096-1c31-4c13-88fa-575563723886>
    -- Channel Local/mobilephones@default-00000000;2 left 'simple_bridge' basic-bridge <6cee2c86-f395-4f8d-86ac-1814f373ce47>
    -- Channel PJSIP/161-00000000 swapped with Local/mobilephones@default-00000000;2 into 'simple_bridge' basic-bridge <6cee2c86-f395-4f8d-86ac-1814f373ce47>
    -- Channel Local/mobilephones@default-00000000;1 left 'simple_bridge' basic-bridge <e7e3c096-1c31-4c13-88fa-575563723886>
  == Spawn extension (default, mobilephones, 3) exited non-zero on 'Local/mobilephones@default-00000000;2'
       > 0x596310 -- Strict RTP qualifying stream type: audio
       > 0x596310 -- Strict RTP switching source address to 192.168.1.163:6000
       > 0x85c210 -- Strict RTP qualifying stream type: audio
       > 0x85c210 -- Strict RTP switching source address to 82.178.154.154:47572
       > 0x85c210 -- Strict RTP learning complete - Locking on source address 82.178.154.154:47572
       > 0x596310 -- Strict RTP learning complete - Locking on source address 192.168.1.163:6000
       > 0x859770 -- Strict RTP learning complete - Locking on source address 192.168.1.163:6200
<--- Received SIP request (642 bytes) from UDP:192.168.1.163:5060 --->
BYE sip:192.168.1.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1117529567
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=1664394079
To: <sip:601@192.168.1.17>;tag=7029b327-09cc-4fb1-be25-ec2655be0e2b
Call-ID: 994379434
CSeq: 22 BYE
Contact: <sip:161@192.168.1.163:5060>
Proxy-Authorization: Digest username="161", realm="asterisk", nonce="1648476397/95a9be6a5ca92610363977cc7bac39f0", uri="sip:192.168.1.17:5060", response="46a2eb8c230a9381a580287112aeee5f", algorithm=MD5, cnonce="0a4f113b", opaque="335a834303e6549d", qop=auth, nc=00000002
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (321 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1117529567
Call-ID: 994379434
From: "F-1-1-001" <sip:161@192.168.1.17>;tag=1664394079
To: <sip:601@192.168.1.17>;tag=7029b327-09cc-4fb1-be25-ec2655be0e2b
CSeq: 22 BYE
Server: SHAULA-001(7.4.0)
Content-Length:  0


    -- Channel PJSIP/161-00000000 left 'simple_bridge' basic-bridge <6cee2c86-f395-4f8d-86ac-1814f373ce47>
  == Spawn extension (fullrights, 601, 3) exited non-zero on 'PJSIP/161-00000000'
    -- Channel PJSIP/141-00000001 left 'simple_bridge' basic-bridge <6cee2c86-f395-4f8d-86ac-1814f373ce47>
<--- Transmitting SIP request (443 bytes) to TLS:82.178.154.154:38380 --->
BYE sip:141@82.178.154.154:38380;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 82.178.154.154:5061;rport;branch=z9hG4bKPj750ea5b6-1afd-4f95-a379-f6fdfad95e79;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=3ebf8c05-66b7-48ca-923f-9b4d1b443114
To: <sip:141@82.178.154.154>;tag=1264300688
Call-ID: 56a6c131-795f-4862-9698-b938484c4bf3
CSeq: 19148 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP response (563 bytes) from TLS:82.178.154.154:38380 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 82.178.154.154:5061;rport=5061;branch=z9hG4bKPj750ea5b6-1afd-4f95-a379-f6fdfad95e79;alias
From: "161" <sip:Door_1@192.168.1.17>;tag=3ebf8c05-66b7-48ca-923f-9b4d1b443114
To: <sip:141@82.178.154.154>;tag=1264300688
Call-ID: 56a6c131-795f-4862-9698-b938484c4bf3
CSeq: 19148 BYE
Contact: <sip:141@192.168.1.190:38380;transport=tls>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP request (301 bytes) from UDP:192.168.1.163:5060 --->
OPTIONS sip:192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1126246447
From: <sip:161@192.168.1.17>;tag=673022285
To: <sip:192.168.1.17>
Call-ID: 1685044823
CSeq: 20 OPTIONS
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (446 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1126246447
Call-ID: 1685044823
From: <sip:161@192.168.1.17>;tag=673022285
To: <sip:192.168.1.17>;tag=z9hG4bK1126246447
CSeq: 20 OPTIONS
WWW-Authenticate: Digest realm="asterisk",nonce="1648476425/d9aab596b8795f4c3fbd0b2cc2df03f8",opaque="15b2fb096a66712d",algorithm=md5,qop="auth"
Server: SHAULA-001(7.4.0)
Content-Length:  0


<--- Received SIP request (563 bytes) from UDP:192.168.1.163:5060 --->
OPTIONS sip:192.168.1.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1377064993
From: <sip:161@192.168.1.17>;tag=673022285
To: <sip:192.168.1.17>
Call-ID: 1685044823
CSeq: 21 OPTIONS
Authorization: Digest username="161", realm="asterisk", nonce="1648476425/d9aab596b8795f4c3fbd0b2cc2df03f8", uri="sip:192.168.1.17", response="46de60cc18f72b3f0bb13fad8d95da12", algorithm=MD5, cnonce="0a4f113b", opaque="15b2fb096a66712d", qop=auth, nc=00000001
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (775 bytes) to UDP:192.168.1.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1377064993
Call-ID: 1685044823
From: <sip:161@192.168.1.17>;tag=673022285
To: <sip:192.168.1.17>;tag=z9hG4bK1377064993
CSeq: 21 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: SHAULA-001(7.4.0)
Content-Length:  0

david551 · March 28, 2022, 2:16pm

Please don’t screen-scrape the logs. Please use the full log file, after enabling it.

You are missing parts of the SDP in your logs.

dmsherazi · March 28, 2022, 4:32pm

Do you mean to make pjsip log to a…pcap file and then share that here?

david551 · March 28, 2022, 5:02pm

To /var/log/asterisk/full as a text file.

dmsherazi · March 29, 2022, 6:31am

This is the full log. enabled full in logger.conf and then reloaded asterisk.

Let 141 register via the cellular network.
Called from 161 to a dialplan 601 , that is as follows

exten => 601,1,NoOp(EXECUTING 601 call)
exten => 601,n,System(/usr/lib/shaula720/some_one_at_door.sh&)
exten => 601,n,Dial(PJSIP/101&PJSIP/102&PJSIP/103&PJSIP/104&PJSIP/105&PJSIP/106&PJSIP/107&PJSIP/108&PJSIP/109&PJSIP/110&PJSIP/111&PJSIP/112&PJSIP/113&PJSIP/114&PJSIP/115&PJSIP/116&PJSIP/117&PJSIP/118&PJSIP/119&PJSIP/120&PJSIP/121&PJSIP/122&PJSIP/123&PJSIP/124&Local/mobilephones@default)
exten => 601,n,Hangup()


exten => mobilephones,1,Ringing()
;same => n,AGI(Push.agi)
;same => n,Wait(6)
same => n,System(/bin/sleep 6)
;same => n,StopRinging()
;same => n,Dial(${PJSIP_DIAL_CONTACTS(140)})
;same => n,Set(VOLUME(RX,p)=4)
;same => n,Set(VOLUME(TX,p)=4)
;same => n,Set(VOLUME(RX)=4)
;same => n,Set(VOLUME(TX)=4)
same => n,Dial(${PJSIP_DIAL_CONTACTS(140)}&${PJSIP_DIAL_CONTACTS(141)})
;same => n,Dial(${PJSIP_DIAL_CONTACTS(140@mobile)})
same => n,Hangup()

Result , there is audio but no video

full_log.txt (176.3 KB)

david551 · March 29, 2022, 10:48am

The SDP is still being truncated. I don’t know if that is a problem with logging in chan_pjsip, or if it is really truncated, but it is difficult to use. Can anyone else confirm whether chan_pjsip should log the complete message?

jcolp · March 29, 2022, 10:55am

I’d expect it to log the full message, unless Asterisk is built with LOW_MEMORY in which case it may not. I vaguely recall it may not then. There was also a recent fix[1] for building with musl where it was truncated.

[1] [ASTERISK-29928] logging messages truncated when using MUSL runtime - Digium/Asterisk JIRA

dmsherazi · March 29, 2022, 11:50am

No I didnt use the low memory .

To gte full logs I changed my logger.conf and its as below now

;
; Logging Configuration
;
; In this file, you configure logging to files or to
; the syslog system.
;
; "logger reload" at the CLI will reload configuration
; of the logging system.

[general]
;
; Customize the display of debug message time stamps
; this example is the ISO 8601 date format (yyyy-mm-dd HH:MM:SS)
;
; see strftime(3) Linux manual for format specifiers.  Note that there is also
; a fractional second parameter which may be used in this field.  Use %1q
; for tenths, %2q for hundredths, etc.
;
;dateformat=%F %T       ; ISO 8601 date format
;dateformat=%F %T.%3q   ; with milliseconds
;
;
; This makes Asterisk write callids to log messages
; (defaults to yes)
;use_callids = no
;
; This appends the hostname to the name of the log files.
;appendhostname = yes
;
; This determines whether or not we log queue events to a file
; (defaults to yes).
;queue_log = no
;
; Determines whether the queue_log always goes to a file, even
; when a realtime backend is present (defaults to no).
;queue_log_to_file = yes
;
; Set the queue_log filename
; (defaults to queue_log)
;queue_log_name = queue_log
;
; When using realtime for the queue log, use GMT for the timestamp
; instead of localtime.  The default of this option is 'no'.
;queue_log_realtime_use_gmt = yes
;
; Log rotation strategy:
; none:  Do not perform any logrotation at all.  You should make
;        very sure to set up some external logrotate mechanism
;        as the asterisk logs can get very large, very quickly.
; sequential:  Rename archived logs in order, such that the newest
;              has the highest sequence number [default].  When
;              exec_after_rotate is set, ${filename} will specify
;              the new archived logfile.
; rotate:  Rotate all the old files, such that the oldest has the
;          highest sequence number [this is the expected behavior
;          for Unix administrators].  When exec_after_rotate is
;          set, ${filename} will specify the original root filename.
; timestamp:  Rename the logfiles using a timestamp instead of a
;             sequence number when "logger rotate" is executed.
;             When exec_after_rotate is set, ${filename} will
;             specify the new archived logfile.
;rotatestrategy = rotate
;
; Run a system command after rotating the files.  This is mainly
; useful for rotatestrategy=rotate. The example allows the last
; two archive files to remain uncompressed, but after that point,
; they are compressed on disk.
;
; exec_after_rotate=gzip -9 ${filename}.2
;
;
; For each file, specify what to log.
;
; For console logging, you set options at start of
; Asterisk with -v for verbose and -d for debug
; See 'asterisk -h' for more information.
;
; Directory for log files is configures in asterisk.conf
; option astlogdir
;
; All log messages go to a queue serviced by a single thread
; which does all the IO.  This setting controls how big that
; queue can get (and therefore how much memory is allocated)
; before new messages are discarded.
; The default is 1000
;logger_queue_limit = 250
;
;
[logfiles]
;
; Format is:
;
; logger_name => [formatter]levels
;
; The name of the logger dictates not only the name of the logging
; channel, but also its type. Valid types are:
;   - 'console'  - The root console of Asterisk
;   - 'syslog'   - Linux syslog, with facilities specified afterwards with
;                  a period delimiter, e.g., 'syslog.local0'
;   - 'filename' - The name of the log file to create. This is the default
;                  for log channels.
;
; Filenames can either be relative to the standard Asterisk log directory
; (see 'astlogdir' in asterisk.conf), or absolute paths that begin with
; '/'.
;
; An optional formatter can be specified prior to the log levels sent
; to the log channel. The formatter is defined immediately preceeding the
; levels, and is enclosed in square brackets. Valid formatters are:
;   - [default] - The default formatter, this outputs log messages using a
;                 human readable format.
;   - [json]    - Log the output in JSON. Note that JSON formatted log entries,
;                 if specified for a logger type of 'console', will be formatted
;                 per the 'default' formatter for log messages of type VERBOSE.
;                 This is due to the remote consoles intepreting verbosity
;                 outside of the logging subsystem.
;
; Log levels include the following, and are specified in a comma delineated
; list:
;    debug
;    notice
;    warning
;    error
;    verbose(<level>)
;    dtmf
;    fax
;    security
;
; Verbose takes an optional argument, in the form of an integer level.
; Verbose messages with higher levels will not be logged to the file.  If
; the verbose level is not specified, it will log verbose messages following
; the current level of the root console.
;
; Special level name "*" means all levels, even dynamic levels registered
; by modules after the logger has been initialized (this means that loading
; and unloading modules that create/remove dynamic logger levels will result
; in these levels being included on filenames that have a level name of "*",
; without any need to perform a 'logger reload' or similar operation).
; Note that there is no value in specifying both "*" and specific level names
; for a filename; the "*" level means all levels.  The only exception is if
; you need to specify a specific verbose level. e.g, "verbose(3),*".
;
; We highly recommend that you DO NOT turn on debug mode if you are simply
; running a production system.  Debug mode turns on a LOT of extra messages,
; most of which you are unlikely to understand without an understanding of
; the underlying code.  Do NOT report debug messages as code issues, unless
; you have a specific issue that you are attempting to debug.  They are
; messages for just that -- debugging -- and do not rise to the level of
; something that merit your attention as an Asterisk administrator.  Debug
; messages are also very verbose and can and do fill up logfiles quickly;
; this is another reason not to have debug mode on a production system unless
; you are in the process of debugging a specific issue.
;
debug => debug
;security => security
console => notice,warning,error
;console => notice,warning,error,debug
messages => notice,warning,error
full => notice,warning,error,debug,verbose,dtmf,fax
;
;full-json => [json]debug,verbose,notice,warning,error,dtmf,fax
;
;syslog keyword : This special keyword logs to syslog facility
;
;syslog.local0 => notice,warning,error
;

after that used the command pjsip set logger on

and later collect the file at /var/log/asterisk/full.

Do i need increse the debug and verbose levels by using the command core set debug x command and core set verbose x and if yes to what values?

dmsherazi · March 29, 2022, 12:19pm

full.txt (1021.3 KB)

can you check this ? Its with debug and verbose set to 5

david551 · March 29, 2022, 7:17pm

Still truncated in the logs:

Content-Length:   695

v=0
o=- 1646642290 1646642290 IN IP4 5.37.239.43
s=Asterisk
c=IN IP4 5.37.239.43
t=0 0
m=audio 31888 RTP/SAVP 0 8 3 111 101
a=crypto:1

The content length actually logged is more like 142 than 695.

dmsherazi · March 31, 2022, 11:03am

if I capture the packets by wireshart will that help?

david551 · March 31, 2022, 11:36am

That should show the complete packets. Ideally, get the UDP exchange as text (which can be done, although I have to work it out each time).