inbound call

[roll72]

Hello all,

I was following the tutorial to create a inbound DID (I create a default one and asked the system to push it to a valid voicemail 12345) in my vicidial 2.2.
but despite what I m doing, no inbound traffic is going through, my voip provider is passing the call but my server don't answer it and but a SIP/2.0 403 Forbidden and my inbound call is going to 'not available' ring.

Does someone got similar issue ?


[Nov 11 00:05:00] --- (17 headers 13 lines) ---
[Nov 11 00:05:00] Using INVITE request as basis request - 0080826001239E0C5E6E00001A83@83.167.xxx.yyy
[Nov 11 00:05:00] Sending to 83.167.xxx.yyy : 5060 (NAT)
[Nov 11 00:05:00] Found peer 'siptrunk'
[Nov 11 00:05:00] NOTICE[11657]: chan_sip.c:10704 handle_request_invite: Failed to authenticate user <sip:09xxxxxxxx@83.167.xxx.yyy>;tag=0080826001239E0C5E6E00000D8D
[Nov 11 00:05:00] Reliably Transmitting (NAT) to 83.167.xxx.yyy:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 83.167.xxx.yyy;branch=z9hG4bK8757.46b57b87.0;received=83.167.xxx.yyy
Via: SIP/2.0/UDP 83.167.xxx.yyy:5060;rport=5060;branch=z9hG4bK0080826001239E0C5E700000152A

[mflorell]

Asterisk version?

admin.php version and build?

[roll72]

Asterisk 1.2.30.4 built by root @ william on a i686 running Linux on 2009-10-21 09:52:52 UTC


Vicidial :
VERSION: 2.2.0-220
BUILD: 90930-2107

[mflorell]

We have seen a large improvement in SIP compatibility by upgrading to Asterisk 1.4.21.2 as compared to Asterisk 1.2.

[roll72]

this server was done from vicibox and then updated by svn as defined in documentation.

If there are any improvment to go to asterisk 1.4, I'm ok, does there is any documentation to know how to update vicidial to asterisk 1.4 ?

[mflorell]

There are instructions at the bottom of the REQUIRED_APPS doc to upgrade to Asterisk 1.4.21.2

[roll72]

Does the fact to migrate to Asterisk 1.4.21.2 will solve the issue with inbound call ?

does there is a basic way to test an income call which will work 100% ?

thanks in advance

[mflorell]

Nothin seems to be 100% in Asterisk, but we have fixed a lot of issues with SIP connections by upgrading to 1.4.21.2.