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

[Klaus Peras]

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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