[Asterisk-Users] problem with gsm codec

[Daniel Chabrol]

Hello list!

I only can make successful calls if I disable gsm with "disallow=gsm". As soon as I allow gsm the following appears at the console. There are much much more Lines with 
"File dsp.c, Line 1198" but I cut them for a better survey :

--------- Log Start -------------
Asterisk Ready.
WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 0487461566dd401a0759b84a70318c87 at 217.140.94.11 for seqno 103 (Request)
WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 0487461566dd401a0759b84a70318c87 at 217.140.94.11 for seqno 102 (Request)

    -- Executing Dial("SIP/daniel-67b4", "SIP/chabrol/49761800127") in new stack

    -- Called chabrol/49761800127

    -- SIP/chabrol-a5f1 is making progress passing it to SIP/daniel-67b4
WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames

    -- SIP/chabrol-a5f1 answered SIP/daniel-67b4

    -- Attempting native bridge of SIP/daniel-67b4 and SIP/chabrol-a5f1
WARNING[15376]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
WARNING[15376]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 8450f5d2aa052b9e at 217.140.94.10 for seqno 48343 (Response)

  == Spawn extension (daniel, 0049761800127, 1) exited non-zero on 'SIP/daniel-67b4'
--------- Log End -------------

I testet it with the current cvs version.

I have only 2 theories:
1) There is a processor specific optimization which my VIA-processor doesn't understand. But I set "PROC=i586" in the Makefile which should be the right setting.
2) There might be a codec-convert problem.  My grandstream budget tone 100 series  isn't able to speak gsm and therefore asterisk has to convert for example from pcmu to gsm.

Are there other people with this problem?

Best regards,
Daniel