OVH Trunk Setup

Hello, i’m a bit lost with a new trunk that i have to set, impossible to make inbound / outbound calls, trunk is configured with settings provided, my issue is similar than the one in this topic but the solution in this one doesn’t work for me, the issue is the trunk is well registred to my provider, i can see in asterisk infos :

siptrunk.ovh.net:5060 Y 0033XXXXXXXX 105 Registered Fri, 07 Oct 2016

I therefore added to the file extensions_custom.conf :

[Custom-get-did-ovh]
exten => __X.,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)

So if someone can help me ?

OK, you’re new. Remember, we can’t log in to your server to look up things like your FreePBX or Asterisk versions. Also, you didn’t tell us what problems you are actually having - while the issue in the topic might be similar to yours, there are lots of people that have enough experience to see the subtle differences between one problem and another.

Your trunk is registered.

  • Did you set up a trunk definition for your inbound calls?
  • Did you set up an Inbound Route with no DID and no CID so that any call that shows up at your server is answered?
  • Did you set up an Outbound Route that points to your trunk?
  • Did you look in the /var/log/asterisk/full log file to see what error you are getting?

Did you set up a trunk definition for your inbound calls?

USER Details

type=peer
host=ippi.fr
context=custom-get-did-ovh
nat=yes
canreinvite=no

Register String

00333XXXXXXXX:[email protected]

PEER Details

type=peer
host=siptrunk.ovh.net
username=00333XXXXXXXX
secret=password
fromuser=00333XXXXXXXX
fromdomain=siptrunk.ovh.net
nat=yes
canreinvite=no

Did you set up an Inbound Route with no DID and no CID so that any call that shows up at your server is answered?

I don’t need inbound, i set a simple outbound route those is going to the trunk with this dial pattern

0[1-7]XXXXXXXX

FreePBX 13.0.188.8 - Asterisk 13.11.2

Thanks for your help !!

If you are not trying to get inbound working, then you are completely off in the weeds. The trunk registration and user defined context is ONLY for inbound calling. Post a sanitized call trace of an outbound call, you will probably see that the provider is rejecting the call for some reason.

when i try to make outbound calls => all-circuits-busy-now sorry i’m french and i don’t speak very well English, i’m trying to try to understand the hundreds line generated by asterisk for every calls that i try to make …

>     -- Called SIP/ovh/03XXXXXXXX(number i try to call)
> [2016-10-07 18:46:40] WARNING[31078][C-00000867]: chan_sip.c:23840 handle_response_invite: Received response: "Forbidden" from '<sip:[email protected]>;tag=as523e401f'
>   == Everyone is busy/congested at this time (1:0/0/1)
>     -- Executing [[email protected]:24] NoOp("SIP/2-00000b6d", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
>     -- Executing [[email protected]:25] GotoIf("SIP/2-00000b6d", "0?continue,1:s-CHANUNAVAIL,1") in new stack
>     -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
>     -- Executing [[email protected]:1] Set("SIP/2-00000b6d", "RC=21") in new stack
>     -- Executing [[email protected]:2] Goto("SIP/2-00000b6d", "21,1") in new stack
>     -- Goto (macro-dialout-trunk,21,1)
>     -- Executing [[email protected]:1] Goto("SIP/2-00000b6d", "continue,1") in new stack
>     -- Goto (macro-dialout-trunk,continue,1)
>     -- Executing [[email protected]:1] NoOp("SIP/2-00000b6d", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
>     -- Executing [[email protected]:2] ExecIf("SIP/2-00000b6d", "1?Set(CALLERID(number)=2)") in new stack
>     -- Executing [03XXXXXXXX(number i try to call)@from-internal:7] Macro("SIP/2-00000b6d", "outisbusy,") in new stack
>     -- Executing [[email protected]:1] Progress("SIP/2-00000b6d", "") in new stack
>     -- Executing [[email protected]:2] GotoIf("SIP/2-00000b6d", "0?emergency,1") in new stack
>     -- Executing [[email protected]:3] GotoIf("SIP/2-00000b6d", "0?intracompany,1") in new stack
>     -- Executing [[email protected]:4] Playback("SIP/2-00000b6d", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
>     -- <SIP/2-00000b6d> Playing 'all-circuits-busy-now.ulaw' (language 'fr')
>        > 0x7f7db06f93e0 -- Probation passed - setting RTP source address to 62.XXX.XX.XX:62284
>        > 0x7f7db06f93e0 -- Probation passed - setting RTP source address to 62.XXX.XX.XX:62284
> [2016-10-07 18:46:42] WARNING[14701][C-00000867]: file.c:774 ast_openstream_full: File pls-try-call-later does not exist in any format
> [2016-10-07 18:46:42] WARNING[14701][C-00000867]: file.c:1110 ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such file or directory
> [2016-10-07 18:46:42] WARNING[14701][C-00000867]: app_playback.c:494 playback_exec: Playback failed on SIP/2-00000b6d for all-circuits-busy-now&pls-try-call-later, noanswer
>     -- Executing [[email protected]:5] Congestion("SIP/2-00000b6d", "20") in new stack
> [2016-10-07 18:46:42] WARNING[14701][C-00000867]: channel.c:4910 ast_prod: Prodding channel 'SIP/2-00000b6d' failed
>   == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/2-00000b6d' in macro 'outisbusy'
>   == Spawn extension (from-internal, 03XXXXXXXX(number i try to call), 7) exited non-zero on 'SIP/2-00000b6d'
>     -- Executing [[email protected]:1] Macro("SIP/2-00000b6d", "hangupcall") in new stack
>     -- Executing [[email protected]:1] GotoIf("SIP/2-00000b6d", "1?theend") in new stack
>     -- Goto (macro-hangupcall,s,3)
>     -- Executing [[email protected]:3] ExecIf("SIP/2-00000b6d", "0?Set(CDR(recordingfile)=)") in new stack
>     -- Executing [[email protected]:4] Hangup("SIP/2-00000b6d", "") in new stack
>   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/2-00000b6d' in macro 'hangupcall'
>   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/2-00000b6d'

Your provider is rejecting the call, possibly because the number or CID is not formatted correctly, or possibly for some other reason. You might try asking them.

So … after some hours experimenting with sip settings, all is working as expected :smile: and yes i’m happy ! Thanks for your help, i’ll be back tomorow for post my working config for this french sip provider with “NO SUPPORT” !

Hi,

May I ask you to post your configuration because I have the same problem with Asterisk 11 and 13.

Regards

Richard