[asterisk-users] SIP interconnection problem

[Tarek Sawah]

you need to post you SIP.conf and your Extensions.conf so someone can have a look at them and see if there is anything missing
what are the contexts you are using with your peers?
what is the dial plan triggered when calling your destination number?
--
AHD Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP

Syria: +963 944 618286

USA: +1 347 562 2308






> Date: Sun, 25 Oct 2009 15:19:28 +0100
> From: robert.bielik at xponaut.se
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] SIP interconnection problem
> 
> Hi all,
> 
> I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using
> IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a 
> Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension
> on the other * I get a "Failed to authenticate on INVITE" on the * to which the Zoiper is registered:
> 
>    -- Accepting AUTHENTICATED call from 192.168.10.113:  << Zoiper IP
>       > requested format = gsm,
>       > requested prefs = (),
>       > actual format = ulaw,
>       > host prefs = (ulaw|alaw|gsm),
>       > priority = mine
>    -- Executing [010001 at users:1] Dial("IAX2/2200-12940", "SIP/010001 at 192.168.10.11") in new stack
>  == Using SIP RTP CoS mark 5
>    -- Called 010001 at 192.168.10.11 << Other *
> [Oct 23 11:08:25] NOTICE[13576]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '"2200" <sip:2200 at 192.168.10.77>;tag=as3e4fedb8'  << 192.168.10.77 == * for Zoiper
>    -- SIP/192.168.10.11-0a1716f8 is circuit-busy
>  == Everyone is busy/congested at this time (1:0/1/0)
>    -- Auto fallthrough, channel 'IAX2/2200-12940' status is 'CONGESTION'
>    -- Hungup 'IAX2/2200-12940' 
> 
> Why does * try to authenticate on sip:2200 at 192.168.10.77, it is IAX for crying out loud :) ? I've set canreinvite=no on
> the IAX phone (not sure this has any meaning in IAX at all)
> 
> Not sure that this is root of the interconnection problem, since I then get SIP/192.168.10.11.. is circuit-busy... ?
> 
> TIA
> /R
> 
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