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

[Klaus Darilion]

Arnold P. Siboro wrote:
> I did not install the AMR codec, but then there was no codec related
> error message. I then did instll AMR by patching and recompiling
> Asterisk as instructed on this URL
> http://lists.digium.com/pipermail/asterisk-dev/2007-March/026446.html
> And I believe it's put in place, since Asterisk log shows as follows:
> 
> logger.c:   == Registered translator 'amrtolin' from format amr to slin,
> cost 2
> logger.c:   == Registered translator 'lintoamr' from format slin to amr,
> cost 11
> logger.c: codec_amr.so => (AMR Coder/Decoder)
> 
> With this dialplan:
> 
> [from-zaptel]
> exten => s,1,h324m_gw(s at threegvideo)
> 
> [threegvideo]
> exten => s,1,Dial(SIP/1002)
> 
> I could finally got the call on X-lite (X-Lite rings) but when it is
> picked up the call was disconnected.
> 
>     -- Accepting data call from '08017734983' 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-6e38,2", "SIP/1002") in new stack
>     -- Called 1002
>     -- SIP/1002-08d9ee60 is ringing
>     -- SIP/1002-08d9ee60 answered Local/s at threegvideo-6e38,2
>   == Spawn extension (threegvideo, s, 1) exited non-zero on 'Local/s at threegvideo-6e38,2'
> 
> I wonder how to find out what is the reason of disconnection (are there
> ways to investigate via logs for example)? X-Lite does not support AMR 
> (Supported audio codecs are Broadvoice-32, Broadvoice-32 FEC, G.711aLaw,
> G.711uLaw, GSM, iLBC, Speex), but Asterisk is supposed to handle
> transcoding between AMR and for example G.711, isn't?

Yes - asterisk should handle the transcoding - in theorie. I also could 
not get it working too. Maybe it is related to the "byte-alignment" 
problem within AMR. Sergio posted a new patch which should fix this but 
the patch did not work. :-(

regards
klaus

> 
> 
> Pada Mon, 23 Jul 2007 13:54:54 +0200
> si Klaus Darilion <klaus.mailinglists at pernau.at> bilang:
> 
>> 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)
> 
> "Anyone who has never made a mistake has never tried anything new." 
>                                           --  Albert Einstein
> 
> 
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