svbrf
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]REMOVED REMOVED
Log output:
[May 9 15:21:40] – Called 8600051@defaultsip:[email protected] ;tag=as73655172’
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
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
May 9, 2024, 2:15pm
4
Whatever you called rejected it due to authentication failure.
1 Like
svbrf
May 9, 2024, 2:20pm
5
Thanks man, outbound calling is working now.
system
May 16, 2024, 2:21pm
6
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