[Asterisk-Users] Grandstream GXP-2000 call end

Hi When I make a call with the Grandstream GXP-2000 through Asterisk (and SER) to landline using VSP, after I hang up the call the other party are still connected for another 30-40 seconds. I've notice that the SIP BYE is sent to Asterisk, but Asterisk sends no SIP BYE on to VSP. When I use the

[Asterisk-Users] codec variable for incoming calls

Hi, In Asterisk, is there way to find out which codec is being used by incoming call? Is there some variable or function call that can be done? Thanks -- Shaun *** If you receive this email by mistake, please notify us and do not make any use o

[Asterisk-Users] sip.conf codecs: ulaw, alaw and g729

Hi, When ever I put g729 in allow for trunk the other two codecs (ulaw and alaw) stop working and I get the frame type error for them, but g729 works fine. I've cleared general part of sip.conf of codec info to be on safe side. If ulaw and alaw are the only ones allowed they work fine. Asterisk