I've found that happens when one version of asterisk is 1.2 and the
other end is running 1.0.9 and you are connecting over IAX2.
If you bridge the two servers with SIP it will be fine.
-bill
On 9-Nov-05, at 11:52 AM, Olivier Taylor wrote:
That's a call to pstn
Callee and caller have 9729 but asterisk (astlinux and soekris)
tell me that
there is no match and give me an error :(
Any idea?
Kind regards,
Olivier
9 headers, 11 lines
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 82.146.123.246:38098
Found description format G729
Found description format telephone-event
Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer -
audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723),
combined - 0x1
(g723)
Nov 9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format:
Unable to
find a path from g729 to gsm
Nov 9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format:
Unable to
find a path from ilbc to g729
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