Re: how to setup SIP-324M gateway

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.mailinglistspernau.at>
Reply-To: Development discussion of video media support in
Asterisk<asterisk-videolists.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(sthreegvideo)
>> 
>> [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 [sfrom-zaptel:1]
h324m_gw("Zap/1-1", "sthreegvideo") in new
stack
>>     -- Executing [sthreegvideo:1]
Dial("Local/sthreegvideo-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/sthreegvideo-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" <asiboromaltech.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(sthreegvideo)
>>>
>>> [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 [sfrom-zaptel:1]
Answer("Zap/1-1", "") in new stack
>>>   == Auto fallthrough, channel 'Zap/1-1' status
is 'UNKNOWN'
>>>     -- Executing [hfrom-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 (asiboromaltech.jp)
>>>
>>> "Imagination is more important than
knowledge." 
>>>                                  --Albert
Einstein.
>>>
>>>
>>>
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>>>
>> 
>> 
>> Arnold P. Siboro (asiboromaltech.jp)
>> 
>> The opinions expressed herein are not necessarily
those of my employer, 
>> not necessarily mine, and probably not necessary.
>> 
>> 
>> _______________________________________________
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om--
>> 
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>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-video
>
>_______________________________________________
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om--
>
>asterisk-video mailing list
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>   http://lists.digium.com/mailman/listinfo/asterisk-video
>
 

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