svbrf
(Martin)
May 9, 2024, 12:02pm
1
Hello, I’m having trouble to connect calls running Asterisk 16.30.0-vici built by abuild @ i02-ch2d on a x86_64.
Inbound calls work but outbound calls only flickers one millisecond on reciever screen (see screenshot) then hangs up. The caller receives response “call congestion”.
What could be the reason for this?
asterisk -rx “sip show peer 23761” gives status “OK” but Im not sure If I must simplify trunk settings to get call to connect properly:
[23761]
disallow=all
allow=opus,gsm,g729,ulaw,alaw
type=friend
username=23761
secret=REMOVED
host=REMOVED
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
Log output:
[May 9 15:21:40] – Called 8600051@default
[May 9 15:21:40] – Executing [8600051@default:1] MeetMe(“Local/8600051@default-0000001f;2”, “8600051,F”) in new stack
[May 9 15:21:40] – Local/8600051@default-0000001f;1 answered
[May 9 15:21:40] – Executing [90812162095@default:1] AGI(“Local/8600051@default-0000001f;1”, “agi://127.0.0.1:4577/call_log”) in new stack
[May 9 15:21:40] – AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=KAMPANJ1))
[May 9 15:21:40] – AGI Script Executing Application: (EXEC) Options: (Set(_CAMPDTO=60))
[May 9 15:21:40] – <Local/8600051@default-0000001f;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 9 15:21:40] – Executing [90812162095@default:2] Dial(“Local/8600051@default-0000001f;1”, “SIP/23761/0812162095,To”) in new stack
[May 9 15:21:40] == Using SIP RTP CoS mark 5
[May 9 15:21:40] – Called SIP/23761/0812162095
[May 9 15:21:40] WARNING[3702][C-00000054]: channel.c:6768 ast_channel_make_compatible_helper: No path to translate from SIP/23761-00000024 to Local/8600051@default-0000001f;1
[May 9 15:21:40] == Spawn extension (default, 90812162095, 2) exited non-zero on ‘Local/8600051@default-0000001f;1’
[May 9 15:21:40] WARNING[3702][C-00000054]: func_hangupcause.c:138 hangupcause_read: Unable to find information for channel
[May 9 15:21:40] – Executing [h@default:1] AGI(“Local/8600051@default-0000001f;1”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----0-----CONGESTION---------------)”) in new stack
[May 9 15:21:40] – <Local/8600051@default-0000001f;1>AGI Script agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----0-----CONGESTION---------------) completed, returning 0
[May 9 15:21:40] == Spawn extension (default, 8600051, 1) exited non-zero on ‘Local/8600051@default-0000001f;2’
[May 9 15:21:40] WARNING[3703][C-00000053]: func_hangupcause.c:138 hangupcause_read: Unable to find information for channel
[May 9 15:21:40] – Executing [h@default:1] AGI(“Local/8600051@default-0000001f;2”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----0--------------------)”) in new stack
[May 9 15:21:40] – <Local/8600051@default-0000001f;2>AGI Script agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----0--------------------) completed, returning 0
[May 9 15:21:41] NOTICE[3092][C-00000054]: chan_sip.c:24449 handle_response_invite: Failed to authenticate on INVITE to ‘“M5091521400000000032” sip:[email protected] ;tag=as73655172’
david55
(david55)
May 9, 2024, 1:38pm
2
You don’t have the codec modules installed that are needed to convert between the incoming and outgoing codec, or you have exceeded the licence limit on proprietary codec module.
Asterisk 16 is not supported.
Asterisk 16.30.0 has known security vulnerabilities.
The -vici suffix means it isn’t even the standard Asterisk code, and it isn’t the FreePBX customised one, either.
svbrf
(Martin)
May 9, 2024, 2:04pm
3
Changed trunk config to allow=ulaw , here is new log:
May 9 16:01:27] NOTICE[3092][C-00000064]: chan_sip.c:24449 handle_response_invite: Failed to authenticate on INVITE to '"M5091601260000000036" <sip:[email protected] >;tag=as5674181f'
[May 9 16:01:27] -- SIP/23761-0000002a is circuit-busy
[May 9 16:01:27] == Everyone is busy/congested at this time (1:0/1/0)
[May 9 16:01:27] -- Executing [90812162095@default:3] Hangup("Local/8600051@default-00000025;1", "") in new stack
[May 9 16:01:27] == Spawn extension (default, 90812162095, 3) exited non-zero on 'Local/8600051@default-00000025;1'
[May 9 16:01:27] -- Executing [h@default:1] AGI("Local/8600051@default-00000025;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION---------------SIP 401 Unauthorized)") in new stack
[May 9 16:01:27] -- <Local/8600051@default-00000025;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION---------------SIP 401 Unauthorized) completed, returning 0
[May 9 16:01:27] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000025;2'
[May 9 16:01:27] WARNING[12876][C-00000063]: func_hangupcause.c:138 hangupcause_read: Unable to find information for channel
[May 9 16:01:27] -- Executing [h@default:1] AGI("Local/8600051@default-00000025;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21--------------------)") in new stack
[May 9 16:01:27] -- <Local/8600051@default-00000025;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21--------------------) completed, returning 0
[May 9 16:01:27] == Manager 'sendcron' logged off from 127.0.0.1
[May 9 16:01:28] Really destroying SIP dialog '[email protected] :5060' Method: INVITE
[May 9 16:01:33] Really destroying SIP dialog 'DLGCH_fhEGCGFxaTZ7QwNXa3o1MH9HB1hlcTZqe0IEWWt9YmcPSVVAYntpfX1AUEBifWVpekBTXg--' Method: ACK
[May 9 16:01:36] Reliably Transmitting (NAT) to 192.168.1.102:5062:
[May 9 16:01:36] OPTIONS sip:[email protected] :5062 SIP/2.0
[May 9 16:01:36] Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK7287945b;rport
[May 9 16:01:36] Max-Forwards: 70
[May 9 16:01:36] From: "asterisk" <sip:[email protected] >;tag=as76266a19
[May 9 16:01:36] To: <sip:[email protected] :5062>
[May 9 16:01:36] Contact: <sip:[email protected] :5060>
[May 9 16:01:36] Call-ID: [email protected] :5060
[May 9 16:01:36] CSeq: 102 OPTIONS
[May 9 16:01:36] User-Agent: Asterisk PBX 16.30.0-vici
[May 9 16:01:36] Date: Thu, 09 May 2024 14:01:36 GMT
[May 9 16:01:36] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[May 9 16:01:36] Supported: replaces, timer
[May 9 16:01:36] Content-Length: 0
jcolp
(Joshua Colp)
May 9, 2024, 2:15pm
4
Whatever you called rejected it due to authentication failure.
1 Like
svbrf
(Martin)
May 9, 2024, 2:20pm
5
Thanks man, outbound calling is working now.
system
(system)
Closed
May 16, 2024, 2:21pm
6
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.