vicidial.org

I make the calls throught sipura 3000, when I calling with the vicidial I got the same log:



-- Executing Dial("SIP/888-08971998", "SIP/53947@1123|30|rt") in new stack
Dec 14 15:35:02 WARNING[436]: channel.c:506 ast_best_codec: Don't know any of 0x40000 formats
Dec 14 15:35:02 WARNING[436]: channel.c:506 ast_best_codec: Don't know any of 0x40000 formats
-- Called 53947@1123
-- SIP/1123-08969e10 is ringing
Dec 14 15:35:02 WARNING[9127]: channel.c:2380 set_format: Unable to find a codec translation path from ulaw to unknown
Dec 14 15:35:02 WARNING[9127]: channel.c:2380 set_format: Unable to find a codec translation path from ulaw to unknown
Dec 14 15:35:02 WARNING[9127]: channel.c:2380 set_format: Unable to find a codec translation path from ulaw to unknown
Dec 14 15:35:02 WARNING[9127]: channel.c:2380 set_format: Unable to find a codec translation path from ulaw to unknown
-- SIP/1123-08969e10 answered SIP/888-08971998

in the last line says "answered" but it`s only the pulse tone of sipura without made the call throught ptsn, is there a way to bypassing the first command "answered" in order to wait response of the ptsn phone pickup?.

thx

make sure you have a right codec registered in your sipura or on asterisk server..

what codec are you using for outgoing asterisk?