[Asterisk-Users] SIP to H323 call timeout

[Fred Lee]

My SIP UA is an ATA186 and my H323 gateway is a Cisco 5300 and a Nextone.


>From: administrator tootai <admin at tootai.net>
>Reply-To: asterisk-users at lists.digium.com
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] SIP to H323 call timeout
>Date: Tue, 20 Jul 2004 02:34:31 +0200
>
>Fred Lee a écrit :
>
>>
>>
>>Hi all,
>>
>>I have the following setup:
>>
>>UAs ------------SER ---------- ASTERISK ----------GNUGK --------- GWs
>>
>>SER is configured to route call requests from UAs to Asterisk. Asterisk is 
>>configured to receive the call on SIP channel and dial out to GNUGK over 
>>H323 channel. The problem I'm facing is that asterisk sends out the call 
>>request to GNUGK and times out immediately, so call setup is never 
>>completed. On GNUGK the call request comes in followed by a normal call 
>>drop.
>>
>>Any ideas on what could be the problem ??
>
>Do you use the h323 - Nufone? Is it a recent installation? If so, could be 
>the problem that GW need FastStart and the * h323 don't send it.
>
>>
>>My asterisk configuration, debug and console output are as follow :
>>
>>SIP.CONF
>>======
>>[general]
>>port = 5080
>>bindaddr = 10.10.1.170
>>context = to_GNUGK
>>disallow=all
>>allow=g729
>>
>>
>>H323.CONF
>>=======
>>[general]
>>port = 1720
>>allow = g729
>>gatekeeper = 64.80.103.12
>>allowgkrouted = yes
>>context = to_SER
>>
>>EXTENSIONS.CONF
>>============
>>[general]
>>static = yes
>>writeprotect = yes
>>
>>[to_GNUGK]]
>>exten => _.,1,Dial(h323/${EXTEN}@10.10.1.12:1720,60,C)
>>
>>[to_SER]
>>exten => _.,1,Dial(SIP/${EXTEN}@10.10.1.170:5060,60)
>>
>>
>>
>>DEBUG File
>>==========
>>Jul 15 16:14:10 DEBUG[65541]: Check for res for
>>Jul 15 16:14:10 DEBUG[65541]:  is not a local user
>>Jul 15 16:14:10 DEBUG[65541]: build_route: Record-Route hop: 
>><sip:15613021234 at 10.10.1.170;ftag=661806388;lr=on>
>>Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop: 
>><sip:999012020 at 10.10.1.13:5060;user=phone;transport=udp>
>>Jul 15 16:14:10 DEBUG[311316]: SIMPLE DIAL (NO URL)
>>Jul 15 16:14:10 DEBUG[311316]: type=h323, format=256, 
>>data=15613021234 at 10.10.1.12:1720.
>>Jul 15 16:14:10 DEBUG[311316]: Host: 10.10.1.12:1720  Username: 
>>15613021234
>>Jul 15 16:14:10 DEBUG[311316]: dest=15613021234 at 10.10.1.12:1720, 
>>timeout=0.
>>Jul 15 16:14:13 DEBUG[213006]: Cleaning up our mess
>>Jul 15 16:14:23 DEBUG[311316]: SIMPLE DIAL (NO URL)
>>Jul 15 16:14:23 DEBUG[311316]: type=h323, format=256, 
>>data=t at 10.10.1.12:1720.
>>Jul 15 16:14:23 DEBUG[311316]: Host: 10.10.1.12:1720  Username: t
>>Jul 15 16:14:23 DEBUG[311316]: dest=t at 10.10.1.12:1720, timeout=0.
>>Jul 15 16:14:24 DEBUG[213006]: Cleaning up our mess
>>Jul 15 16:14:31 DEBUG[311316]: SIMPLE DIAL (NO URL)
>>Jul 15 16:14:31 DEBUG[311316]: type=h323, format=256, 
>>data=h at 10.10.1.12:1720.
>>Jul 15 16:14:31 DEBUG[311316]: Host: 10.10.1.12:1720  Username: h
>>Jul 15 16:14:31 DEBUG[311316]: find_user() - decrement inUse counter
>>Jul 15 16:14:31 DEBUG[311316]:  is not a local user
>>Jul 15 16:14:31 DEBUG[65541]: Stopping retransmission on 
>>'842589597 at 10.10.1.13' of Response 1: Found
>>
>>
>>
>>CONSOLE Output
>>==============
>>*CLI>     -- Executing Dial("SIP/-08121388", 
>>"h323/15613021234 at 10.10.1.12:1720|60|C") in new stack
>>   -- Called 15613021234 at 10.10.1.12:1720
>>== No one is available to answer at this time
>>
>>   -- Timeout on SIP/-08121388
>>== CDR updated on SIP/-08121388
>>
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