[asterisk-users] Chan_sip Entry

[Kutman.DK at forces.gc.ca]

Hello,

I am trying to get to Jain Sip softphones to call one another via an Asterisk server.  When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways.  When I look at the asterisk log file it has an entry which says:
"Oooh, format changed to 2". 

Would anyone know why this is occuring one way and not the other, and more importantly, how would I fix this.  After some examination I see that when I send the OK to the INVITE, this SDP body should have a 0 for the codec which is ulaw.  When this Ok message gets to the other pc after going through asterisk it seems like asterisk adds a codec because the SDP body now contains the codecs 0 and 3.  I believe the problem has something to do with this but I am not sure why it would work one way but not the other.

Any help would be greatly appreciated.

Thanks very much,

Denis Kutman