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.bie...@xponaut.se
> To: asterisk-users@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 [010...@users:1] Dial("IAX2/2200-12940",
> "SIP/010...@192.168.10.11") in new stack
> == Using SIP RTP CoS mark 5
> -- Called 010...@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:2...@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:2...@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
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
_________________________________________________________________
Windows 7: I wanted more reliable, now it's more reliable. Wow!
http://microsoft.com/windows/windows-7/default-ga.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:102009_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
