vicidial.org

Hi all, I created a carrier trunk:

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Registration String:
myusername:mypassword@host.fr:5060/myusername

Account Entry
[xxx]
allow=ulaw
type=peer
username=myusername
secret=mypassword
host=host.fr
dtmfmode=auto
nat=force_rport
insecure=very
maxexpiry=3600
qualify=yes
fromuser=myusername
fromdomain=host.fr

Protocol:
SIP

Globals String:

Dialplan Entry:
exten => _933.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _933.,2,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 so I added this line inside it:
register => myusername:mypassword@host.fr:5060/myusername

But after that, i got

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

.
So, can anyone help me with this please?




Vicibox 9.0 from .iso | Vicidial 3133-2.14-718a Build 190902-0839| Asterisk 13.27.0 | Single Server | vmware | No Extra Software After Installation

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

Yes, I paid for channels now. It is prodived by ovh.

I need help please

What is the issue you are having now? What is happening when you try to dial? Please provide some CLI output from asterisk as you try to call so I can see the logs.

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

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


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

neither image loads....please copy and paste the output into a code bracket on here

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

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[Jul 20 12:45:01] Really 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:77718192.168.147.1:31130;rinstance=8d095f038e2b6f4 c SIP/2.0
[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@192.168.147.1:31130;rinstance=8d095f038e2b6f4c>
[Jul 20 12:45:01] Contact: <sip:asterisk@192.168.147.128:5060>
[Jul 20 12:45:01] Call-ID: 5646e3d53f8e441b36bf8e282dlefe2f8192.168.147.128:5060
[Jul 20 12:45:01] CSeq: 102 OPTIONS
[Jul 20 12:45:01] User-Agent: Asterisk PBX 13.29.2-vici
[Jul 20 12:45:01] Date: Mon, 20 Jul 2020 12:44:57 GMT
[Jul 20 12:45:01] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 20 12:45:01] Supported: replaces, timer
[Jul 20 12:45:01] Content-Length: 0
[Jul 20 12:45:01]
[Jul 20 12:45:01]
[Jul 20 12:45:01] ---
[Jul 20 12:45:01] Really destroying SIP dialog '5646e3d53f8e441b36bf8e282dlefe2f
8192.168.147.128:5060' Method: OPTIONS


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

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001", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 20 12:37:25] <SIP/7771-00000001>AGI Script agi://127.0.0.1:4577/call _log completed, returning 0
[Jul 20 12:37:25] Executing [0033xxxx@default:21 Dial("SIP/7771-0000 0001", "0033xxxx:xx@sip3.ovh.fr:5060/033xxxx,,tTor") in new stack
[Jul 20 12:37:25] WARNING[3343][C-00000001]: channel.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 type '0033xxxx:xx@sip3.ovh.fr:5060' (cause 66 - Channel not implemented)
[Jul 20 12:37:25] Everyone is busy/congested at this time (1:0/0/1)
[Jul 20 12:37:25] Executing [0033xxxx@default:3] Hangup("SIP/7771-00 000001", "") in new stack
[Jul 20 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 find information for channel
[Jul 20 12:37:25] Executing [h@default:11 AGI("SIP/7771-00000001", "agi://127.0.0.1:4577/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

Jul 20 12:37:25] WARNING[3343][C-00000001]: channel.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 type '0033xxxx:xx@sip3.ovh.fr:5060' (cause 66 - Channel not implemented)
[Jul 20 12:37:25] Everyone is busy/congested at this time (1:0/0/1)

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

here is my dialplan:

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exten => _0033XXXXXXXXX,1,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 :

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

are your phones registering?

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

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Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
5551/5551                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
7771/7771                 192.168.147.1                            D  Yes        Yes            11778    OK (14 ms)
gs102/gs102               (Unspecified)                            D  Yes        Yes            0        UNKNOWN
ovh/00339xxxxxxxx         91.121.129.159                              Yes        Yes            5060     UNREACHABLE
4 sip peers [Monitored: 1 online, 3 offline Unmonitored: 0 online, 0 offline]


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 ask them for a white paper. I am still rather new to the whole dialplan thing.

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