[Asterisk-video] how to setup SIP-324M gateway

Sergio Garcia sergio.garcia at fontventa.com
Thu Jul 26 06:30:56 CDT 2007


I'm currently in Ireland for a training course. I'll be back to daily work next week.

Best regards
Sergio


---------- Original Message ----------------------------------
From: Klaus Darilion <klaus.mailinglists at pernau.at>
Reply-To: Development discussion of video media support in Asterisk<asterisk-video at lists.digium.com>
Date:  Mon, 23 Jul 2007 13:54:54 +0200

>Have you installed the AMR codec?
>
>regards
>klaus
>
>Arnold P. Siboro wrote:
>> Actually the instruction does not mention the Answer line, so I fixed
>> the configuration to as follows:
>> 
>> [from-zaptel]
>> exten => s,1,h324m_gw(s at threegvideo)
>> 
>> [threegvideo]
>> exten => s,1,Dial(SIP/1002)
>> 
>> But the caller keeps ringing while callee (1002) receives nothing, as
>> follows:
>> 
>> Connected to Asterisk 1.4.7.1-BRIstuffed-0.4.0-test2 currently running on asterisk1 (pid = 22790)
>> ionerisk1*CLI>
>> Verbosity is at least 3
>>     -- Going to extension s|1 because of immediate=yes
>>     -- Accepting voice call from '0948523078' to 's' on channel 0/1, span 1
>>     -- Executing [s at from-zaptel:1] h324m_gw("Zap/1-1", "s at threegvideo") in new stack
>>     -- Executing [s at threegvideo:1] Dial("Local/s at threegvideo-3a33,2", "SIP/1002") in new stack
>>     -- Couldn't call 1002
>>   == Everyone is busy/congested at this time (0:0/0/0)
>>   == Auto fallthrough, channel 'Local/s at threegvideo-3a33,2' status is 'CHANUNAVAIL'
>>   == Spawn extension (from-zaptel, s, 1) exited non-zero on 'Zap/1-1'
>>     -- Hungup 'Zap/1-1'
>> 
>> 
>> Pada Mon, 23 Jul 2007 14:54:08 +0900
>> si "Arnold P. Siboro" <asiboro at maltech.jp> bilang:
>> 
>>> I got my Asterisk box running and tested with ISDN line. Furthermore, h324m_loopback()
>>> also worked perfectly. I want to setup a SIP-324M gateway, following the
>>> instruction on libh324m gateway, I set it as follows:
>>>
>>> [from-zaptel]
>>> exten => _.,1,Answer
>>> ;exten => s,10,h324m_gw(SIP/1002)
>>> exten => _X.,1,h324m_gw(s at threegvideo)
>>>
>>> [threegvideo]
>>> exten => s,1,Dial(SIP/1002)
>>>
>>> However, it does not work (caller keeps ringing but callee does not get
>>> does not response), giving the following message:
>>>
>>>
>>> Connected to Asterisk 1.4.7.1-BRIstuffed-0.4.0-test2 currently running on asterisk1 (pid = 22790)
>>> ionerisk1*CLI>
>>> Verbosity is at least 3
>>>   == Parsing '/etc/asterisk/manager.conf': Found
>>>   == Parsing '/etc/asterisk/manager_custom.conf': Found
>>>   == Manager 'admin' logged on from 127.0.0.1
>>>     -- Going to extension s|1 because of immediate=yes
>>>     -- Accepting voice call from '0948523078' to 's' on channel 0/1, span 1
>>>     -- Executing [s at from-zaptel:1] Answer("Zap/1-1", "") in new stack
>>>   == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
>>>     -- Executing [h at from-zaptel:1] Answer("Zap/1-1", "") in new stack
>>>     -- Hungup 'Zap/1-1'
>>>
>>> Is it codec problem? I was kind of expecting some message if it's about
>>> codec. BTW, on the SIP end I am using X-Lite, which I think does not
>>> have the right audio codec to talk with h324m endpoint.
>>>
>>>
>>> Arnold P. Siboro (asiboro at maltech.jp)
>>>
>>> "Imagination is more important than knowledge." 
>>>                                  --Albert Einstein.
>>>
>>>
>>> _______________________________________________
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>>>
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>>>
>> 
>> 
>> Arnold P. Siboro (asiboro at maltech.jp)
>> 
>> The opinions expressed herein are not necessarily those of my employer, 
>> not necessarily mine, and probably not necessary.
>> 
>> 
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>> 
>> asterisk-video mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-video
>
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