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Help configure SIP Trunk, conf don't change on server

Added by Test Test over 3 years ago

Hi everyone,

i got a sip trunk on OVH.
I add the carrier on goautodial link it to a campaign.
But when i log in with an agentand put a number manually i got nothing only a message waiting for ring.
And the format is not right i think : calling: (XXX)XXX-XXXX.
And in sip-vicidial.conf and sip-goautodial.conf i don't get the sip that i added but other values.
The GUI doesnt' update the conf file it's normal ?

If someone can help me please :(.


Replies (9)

RE: Help configure SIP Trunk, conf don't change on server - Added by Wittie Manansala over 3 years ago

Hi,

Please provide the following for us review your concern:

1. sip show peers
2. asterisk CLI log when making call.

Thanks

RE: Help configure SIP Trunk, conf don't change on server - Added by Test Test over 3 years ago

Hi thank you for you response,

i made the installation again from 0 because i think i make some mistake.
Now i think the installation is good but the sip is unreachable :
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
OVH1/0033********* 176.31.163.245 Yes Yes 5060 UNREACHABLE
kamailio 176.31.163.245 Yes Yes 5060 UNREACHABLE

RE: Help configure SIP Trunk, conf don't change on server - Added by Wittie Manansala over 3 years ago

Hi,

Which of the two links below you've used for installation?

https://goautodial.org/projects/goautodialce/wiki/Version_4_How_To_Install_Goautodial_From_Scratch_using_CentOS_7X

OR

https://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev4

Based on your sip show peers, It seems you've run update_server_ip or you've change the server ip on goadmin server settings. I recommend to follow the steps posted on our wiki to avoid issues.

Thank you

RE: Help configure SIP Trunk, conf don't change on server - Added by Test Test over 3 years ago

So now i got all configured but and all working.
But can't make call i got a dial timeout, i think the carrier conf got a problem :

exten => _0XXXXXXXXX.,1,AGI
exten => _0XXXXXXXXX.,2,Dial(SIP/0033${EXTEN:1}@OVH1,,tTo)
exten => _0XXXXXXXXX.,3,Hangup()

can you help me?

RE: Help configure SIP Trunk, conf don't change on server - Added by Test Test over 3 years ago

In the asterisk console i got :

1 SIP registrations.
[Jul 16 20:35:28] WARNING1855[C-00000006]: chan_sip.c:24003 handle_response_invite: Received response: "Forbidden" from '"M7162035280000000001" <sip::5070>;tag=as346c9066'
[Jul 16 20:35:28] ERROR2774[C-00000005]: member.c:389 member_exec: unable to answer call
[Jul 16 20:35:28] WARNING2774[C-00000005]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel

RE: Help configure SIP Trunk, conf don't change on server - Added by Test Test over 3 years ago

I see that the asterisk Time may do this?

RE: Help configure SIP Trunk, conf don't change on server - Added by Wittie Manansala over 3 years ago

Hi,

Try to coordinate to your VOIP Provider about your issue. Ask them if they seeing any call attempts from your server. Provide them your carrier settings and asterisk CLI logs.

Thanks

RE: Help configure SIP Trunk, conf don't change on server - Added by Test Test over 3 years ago

Hi,

I called my sip provider and they don't see anything.
And when i make the call directy from zoiper with the sip conf it work.
I don't know what to do..

RE: Help configure SIP Trunk, conf don't change on server - Added by Sotmir Laci over 3 years ago

Hi,
I had the same error in my asterisk log. And the solution was to call my VOIP provider because is a VOIP provider problem. Initially, they said the configuration that you are using on port 5070 is incorrect. Later they entered my server to verify the configurations, and it wasn't a config problem on my side. Only after that, they managed to resolve the problem on their side.

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