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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