[Asterisk-Users] Unable to forward voice

[administrator tootai]

Hi again,

always latest CVS from 27/06/04. Calling to a SIP gateway I receive:

Unable to find a path from G723 to ALAW
Unable to find a path from ULAW to G723
Asked to transmit frame type 4, while native format is 1 (read/write = 8/4)
Unable to forward voice
[last messages repeated lot of times]
Acked pending invite 102 <- My phone number
...
No path to translate from SIP/... to SIP/...
Had to drop call because I couldn't make SIP/... compatible with SIP/...

Even if I force my sip.conf to use only g723.1 I have the same result.

BTW, if I want to modify my codecs in a sip context, it's not taking in 
account by asterisk. Is'it normal behaviour?

-- 
dash