Leandro Rzezak ha scritto:
I'm having same problem. Were you able to solve it?
No, codecs became a secondary problem later in our project so we ended up with 711 on all servers and more bandwidth, anyway the post refers to asterisk 1.0.something and I never investigated the problem in more detail... I think it's possible, usually when you receive no answers (as the case of that post) you have made a really silly question :)
On 10/18/05, *Simone Cittadini* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:I've the following installation : |asterisk client| --- > |asterisk server| --- > |other asterisk server| all the connections are made in IAX, the client and first server allows 711 and 729 the other server only allows 729 since it has low bandwidth at disposal all the numbers but a few are routed to a digium card in the first server, the others are routed to the other server, this way : [default] exten => _123X.,1,Dial(IAX2/otherserver/${EXTEN}) exten => _123X.,2,Hangup exten => _X.,1,Dial(Zap/g1/${EXTEN}) exten => _X.,2,Hangup when I call 123456 from the client box ... on the client : Call accepted by asterisk server (format alaw) on the server : Call accepted by other asterisk server (format g729) on the other server : Called [EMAIL PROTECTED] and then on the server in the middle : Oct 18 18:00:37 NOTICE[2846]: channel.c:1724 ast_set_write_format: Unable to find a path from alaw to g729 Oct 18 18:00:37 NOTICE[2846]: channel.c:1757 ast_set_read_format: Unable to find a path from g729 to alaw since that "something" at the end of the call and the paps which sits before the first asterisk server both have g729, I don't like too much having to pay to translate something which need not translation. Is there a clever combination of sip.conf, iax.conf and extensions.conf I'm missing to solve my problem ? _______________________________________________
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