[Asterisk-Users] SIP to H323 call timeout

Tue Jul 20 04:20:54 MST 2004

Fred Lee a écrit :

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

My question was "which * h323 channel you're using?"  (h323 Nufone or  
oh323) Don't know about Cisco and Nextone but I also use an ATA186 as 
SIP UA with GnuGK and have this problem. If you install an earlier that 
20/05/04 CVS asterisk version with H323 Nufone channel it works. Don't 
know how it works with the stable branch.

Daniel

>
>
>> 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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>>
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