[asterisk-users] two questions regarding incoming call

[Faisal Hanif]

Try insecure=port,invite

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Oguzhan Kayhan
Sent: Tuesday, March 01, 2011 2:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] two questions regarding incoming call

Update,

My first question solved already.
There was an error on my agi script.

But second problem still valid.


On Tuesday, March 01, 2011 11:04:50 am Oguzhan Kayhan wrote:
> Hello,
> I want to make an agi script to match incoming DIDs with usernames.
> 
> I tried to do such entry in incoming trunk.
> 
> [DID_diddw]
> include = from-didww
> 
> [from-didww]
> exten = 3130XXXXXXX,1,AGI("did.php")
> exten = 3130XXXXXXX,n,DIAL(SIP/${yup_no},20)
> 
> 
> but when i run the rule it says
> chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to 
> extension '31301111111' rejected because extension not found in 
> context 'from-didww' Cant I use such agi scripts on incoming calls?
> 
> PS:
> exten = 3130XXXXXXX,n,DIAL(SIP/1111) works alone.
> 
> 
> My second question.
> I got two incoming trunk sip channels on my server.
> 
> One of them is as follows.
> 
> [46.19.209.1]
> host = 46.19.209.1
> type = friend
> insecure = invite
> context = from-didww
> canreinvite=no
> 
> 
> The other is as follows:
> 
> [62.180.237.73]
> host = 62.180.237.73
> type = friend
> insecure = invite
> context = from-btnet2
> canreinvite = no
> 
> 
> 
> The problem is, i get all calls coming from trunk1(didww) without a 
> problem but, when i receive a call from trunk2(btnet) it tries to 
> authenticate the sip call and denies it. It works only if i allow guest
calls.
> What can be the reason for that?
> Thank you.
> 
> 
> 
> 
> 
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users