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

Klaus Darilion klaus.mailinglists at pernau.at
Mon Jul 23 06:54:54 CDT 2007


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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> 
> 
> 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.
> 
> 
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> 
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