Registration for 163841498sipcallwithuscom Timed Out Trying Again

Hello all, I created a carrier trunk:

Code: Select all: Registration String: myusername:mypassword@host.fr:5060/myusernameAccount Entry [xxx] allow=ulaw type=peer username=myusername secret=mypassword host=host.fr dtmfmode=machine nat=force_rport insecure=very maxexpiry=3600 qualify=yes fromuser=myusername fromdomain=host.fr Protocol: SIP Globals String: Dialplan Entry: exten => _933.,i,AGI(agi://127.0.0.1:4577/call_log) exten => _933.,two,Dial(xxxx@host.fr:xxxx@host.fr:5060/${EXTEN:2},,tTor) exten => 933.,3,Hangup Server IP: 192.168.147.128

And the carrier was unreachable . And I noticed that there is no line for the registration in the sip_vicidial.conf and then I added this line within it:
annals => myusername:mypassword@host.fr:5060/myusername

But subsequently that, i got

Code: Select all: registration for myusername@host.fr timed out, trying again (Attempt #2)

.
So, can anyone help me with this please?

by **carpenox** » Wed Jul 15, 2020 2:39 am

does your sip trunk have channels? you have to pay for channels, theres no free ones....

past **Tom75** » Midweek Jul 15, 2020 2:43 am

Yep, I paid for channels now. Information technology is prodived by ovh.

by **carpenox** » Mon Jul 20, 2020 2:12 am

What is the issue you are having now? What is happening when you try to punch? Delight provide some CLI output from asterisk equally you lot endeavor to call so I tin can come across the logs.

When yous are in asterisks turn on sip debug: sip set debug on

past **Tom75** » Mon Jul 20, 2020 8:11 am

It yet can't registering on ovh. And here is what I got when I gear up the debug on in asterisk cli:

And when I make call, this is the output on cli:

by **carpenox** » Mon Jul 20, 2020 8:53 am

neither image loads....delight re-create and paste the output into a code bracket on hither

by **Tom75** » Monday Jul 20, 2020 xi:29 am

Hither it is, the output of cli after debug on:

Lawmaking: Select all: [Jul xx 12:45:01] Actually destroying SIP dialog '56581183325e72d152115ac05a8edd8a@197.215.204.181:5060' Method: OPTIONS [Jul 20 12:45:01] Retransmitting #4 (NAT) to 192.168.147.1:31130: [Jul 20 12:45:01] OPTIONS sip:77718192.168.147.i:31130;rinstance=8d095f038e2b6f4 c SIP/two.0 [Jul 20 12:45:01] Via: SIP/2.0/UDP 192.168.147.128:5060;branch=z9hG4H7565992f:r port [Jul 20 12:45:01) Max-Forwards: 70 [Jul 20 12:45:01] From: "asterisk" <sip:asterisk@192.168.147.128>;tag=as4c962d4b [Jul 20 12:45:013 To: <sip:7771@192.168.147.ane:31130;rinstance=8d095f038e2b6f4c> [Jul 20 12:45:01] Contact: <sip:asterisk@192.168.147.128:5060> [Jul 20 12:45:01] Phone call-ID: 5646e3d53f8e441b36bf8e282dlefe2f8192.168.147.128:5060 [Jul 20 12:45:01] CSeq: 102 OPTIONS [Jul twenty 12:45:01] User-Agent: Asterisk PBX 13.29.2-vici [Jul 20 12:45:01] Appointment: Mon, 20 Jul 2020 12:44:57 GMT [Jul 20 12:45:01] Allow: INVITE, ACK, Cancel, OPTIONS, Good day, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, Bulletin [Jul twenty 12:45:01] Supported: replaces, timer [Jul 20 12:45:01] Content-Length: 0 [Jul xx 12:45:01] [Jul xx 12:45:01] [Jul xx 12:45:01] --- [Jul 20 12:45:01] Actually destroying SIP dialog '5646e3d53f8e441b36bf8e282dlefe2f 8192.168.147.128:5060' Method: OPTIONS

And this is what i get when I brand telephone call from eyebeam:

Lawmaking: Select all: 001", "agi://127.0.0.1:4577/call_log") in new stack [Jul 20 12:37:25] <SIP/7771-00000001>AGI Script agi://127.0.0.ane:4577/telephone call _log completed, returning 0 [Jul 20 12:37:25] Executing [0033xxxx@default:21 Punch("SIP/7771-0000 0001", "0033xxxx:twenty@sip3.ovh.fr:5060/033xxxx,,tTor") in new stack [Jul twenty 12:37:25] WARNING[3343][C-00000001]: channel.c:6354 ast_request: No aqueduct type registered for '0033xxx@sip3.ovh.fr:5060' [Jul 20 12:37:25] Alert[3343][C-00000001]: app_dial.c:2591 dial_exec_full: Unaible to create channel of type '0033xxxx:xx@sip3.ovh.fr:5060' (cause 66 - Channel not implemented) [Jul 20 12:37:25] Everyone is decorated/congested at this time (1:0/0/1) [Jul 20 12:37:25] Executing [0033xxxx@default:iii] Hangup("SIP/7771-00 000001", "") in new stack [Jul twenty 12:37:25] Spawn extension (default, 0033xxx, 3) exited non-ze ro on 'SIP/7771-00000001' [Jul 20 12:37:25] WARMING[3343][C-00000001]: func_hangupcause.c:140 hangupcause_read: Unable to observe information for channel [Jul 20 12:37:25] Executing [h@default:11 AGI("SIP/7771-00000001", "agi://127.0.0.i:4577/call_log--Hcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL-----)") in new stack [Jul 20 12:37:25] <SIP/7771-00000001>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----MODEBUG-----66-----CHANUNAVAIL---------------) completed , returning 0

by **carpenox** » Mon Jul twenty, 2020 three:43 pm

Jul twenty 12:37:25] Warning[3343][C-00000001]: aqueduct.c:6354 ast_request: No channel type registered for

'0033xxx@sip3.ovh.fr

:5060'
[Jul 20 12:37:25] WARNING[3343][C-00000001]: app_dial.c:2591 dial_exec_full: Unaible to create channel of blazon '0033xxxx:xx@sip3.ovh.fr:5060' (cause 66 - Channel non implemented)
[Jul 20 12:37:25] Everyone is decorated/congested at this fourth dimension (1:0/0/1)

this shows your channels are not going thru to sip provider.... did you lot put the correct dialplan for the carrier? did you update sip.conf? extensions.conf? Are the channels ACTIVE?

by **Tom75** » Tue Jul 21, 2020 ane:fifteen am

here is my dialplan:

Lawmaking: Select all: exten => _0033XXXXXXXXX,i,AGI(agi://127.0.0.1:4577/call_log) exten => _0033XXXXXXXXX,2,Dial(xxx:xxx@sip3.ovh.fr:5060/${EXTEN},,tTor) exten => _0033XXXXXXXXX,3,Hangup

what kind of information should I update in sip.conf and extensions.conf, I only update the sip-vicidial.conf :

Lawmaking: Select all: register => username:password@sip3.ovh.fr:5060/username

by **carpenox** » Tue Jul 21, 2020 12:29 pm

are your phones registering?

by **Tom75** » Tue Jul 21, 2020 11:43 pm

Yes, the phone is registered and when I use sip evidence peers, the phone is the only online

Code: Select all: Proper name/username Host Dyn Forcerport Comedia ACL Port Status Description 5551/5551 (Unspecified) D Yes Aye 0 UNKNOWN 7771/7771 192.168.147.1 D Aye Yes 11778 OK (xiv ms) gs102/gs102 (Unspecified) D Yes Yes 0 UNKNOWN ovh/00339xxxxxxxx 91.121.129.159 Aye Yes 5060 UNREACHABLE 4 sip peers [Monitored: i online, 3 offline Unmonitored: 0 online, 0 offline]

by **carpenox** » Wednesday Jul 22, 2020 12:11 am

yea its your carrier. you need to figure out why your box isnt communicating with it which is why i initially told you that you prolly didnt have any channels. This comes down to dialplan which Bill would probably be able to help you with. Or the carrier itself if you inquire them for a white paper. I am nevertheless rather new to the whole dialplan matter.

by **Tom75** » Wednesday Jul 22, 2020 7:24 am

OK, I will enquire them once more. But can I know what kind of information should I edit in extensions.conf and sip.conf?

Vroland Obtlyn

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