Re: [Asterisk-Users] g729 translation to zap (ISDN) doesn´t work

[Klaus Peras]

I thougt i have some problems with ztdummy and removed that # in front 
of ztdummy in the zaptel Makefile before compiling. But still no change.
I even tried it with another Phone, a Planet VIP-150T. Still the same 
Problem, i don´t hear anything from the SIP Phone on the ISDN Phone, but 
i hear everything fine the other way.

Any Ideas? Thanks a lot for help.

regards

Klaus Peras






Klaus Peras schrieb:

> Hi, i just figured out, that there is also a problem by going in a 
> conference with the sip phone that runs the g729a codec.
> Could it be, that i have timing problems? I don´t have digium hardware 
> installed, but i have ztdummy:
>
> asterisk3:/etc/asterisk# lsmod | grep ztdummy
> ztdummy                 3748  0
> zaptel                225540  24 ztdummy,qozap
>
> Does anybody have a advice for me?
>
> Mit freundlichen Grüßen
> With kind regards
>
> Klaus Peras
>
>
>
>
>
>
> Klaus Peras schrieb:
>
>> Hi Asterisk Users,
>>
>> i have a bristuffed-0.2.0-RC8q Asterisk 1.0.9 System running on a 
>> Debian 3.1. With a quadbri card installad, wich is running on the 
>> bristuff drivers.
>> Everything seems to be fine so far.
>> but now i wanted to use the g.729A Codec. I bought 5 licences and 
>> installed them:
>> asterisk3*CLI> show g729
>> 0/0 encoders/decoders of 5 licensed channels are currently in use
>>
>> When i do sip to sip calls, everything is working fine (from a snom 
>> 190 wich is running with that codec to a sip phone with g.711a), 
>> asterisk is translating correct.
>> the output on the CLI is:
>> asterisk3*CLI> show g729
>> 1/0 encoders/decoders of 5 licensed channels are currently in use
>>
>> But if i try to call a zap channel from that sip phone (snom 190) 
>> wich runs that g729 Codec, i don´t hear anything on the ISDN Phone. 
>> the output on the CLI:
>> asterisk3*CLI> show g729
>> 1/1 encoders/decoders of 5 licensed channels are currently in use
>>
>> Here is the output of the show channel command for the SIP Channel 
>> and the ZAP Channel:
>>
>> asterisk3*CLI> show channel SIP/71-d293
>> -- General --
>>           Name: SIP/71-d293
>>           Type: SIP
>>       UniqueID: asterisk-2204-1134137006.49
>>      Caller ID: 30071
>>    DNID Digits: 329
>>          State: Up (6)
>>          Rings: 0
>>   NativeFormat: 256
>>    WriteFormat: 256
>>     ReadFormat: 64
>> 1st File Descriptor: 31
>>      Frames in: 7949
>>     Frames out: 7956
>> Time to Hangup: 0
>>   Elapsed Time: 0h2m39s
>> --   PBX   --
>>        Context: default
>>      Extension: 329
>>       Priority: 2
>>     Call Group: 0
>>   Pickup Group: 0
>>    Application: Dial
>>           Data: Zap/g1/329
>>          Stack: 0
>>    Blocking in: ast_waitfor_nandfds
>> asterisk3*CLI> show channel Zap/1-1
>> -- General --
>>           Name: Zap/1-1
>>           Type: Zap
>>       UniqueID: asterisk-2204-1134137006.50
>>      Caller ID: 30071
>>    DNID Digits: 329
>>          State: Up (6)
>>          Rings: 0
>>   NativeFormat: 72
>>    WriteFormat: 64
>>     ReadFormat: 256
>> 1st File Descriptor: 13
>>      Frames in: 8255
>>     Frames out: 8246
>> Time to Hangup: 0
>>   Elapsed Time: 0h0m0s
>> --   PBX   --
>>        Context: default
>>      Extension: s
>>       Priority: 1
>>     Call Group: 0
>>   Pickup Group: 0
>>    Application: Bridged Call
>>           Data: SIP/71-d293
>>          Stack: -1
>>    Blocking in: ast_waitfor_nandfds
>>
>> I don´t know what i can do on this problem and would be pleased to 
>> get some help.
>>
>> Thank you very much!
>>
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