[asterisk-users] Why asterisk changes RTP destination port when it receives first RTP packet in opposite direction despite canreinvite=no

[rob.r374 at gmail.com]

Hi,

I'm connecting Asterisk v. 1.4.10 to Zanzibar Open IVR that acts as a SIP 
trunk. Since recognition didn't work correctly, I've troubleshot with 
Wireshark and saw that RTP stream is first send to one port on SIP trunk and 
then when first RTP packet arrives in opposite direction (from TTS part of 
Zanzibar - it's a prompt) Asterisk starts sending to the same RTP port - 
therefore changing destination port.

I'm calling sip trunk with IAX2 client, have put canreinvite=no in sip.conf 
and in trunk definition....

What am I doing wrong ?
Anyone else tried Zanzibar (for those not familiar - it's a first open 
source based way of doing speech recognition and synthesis in convenient 
way) ?


Thanks in advance,

regards,

Rob.

Trunk definition :
[Zanzibar]

type=peer

host=192.168.0.50

port=5090

dtmfmode=info

canreinvite=no

qualify=no

Extension definition for calling IVR :

exten => 510,1,Answer

exten => 510,2,SIPAddHeader(x-channel:${CHANNEL})

exten => 510,3,SIPAddHeader(x-application:beanId|Parrot)

exten => 510,4,Dial(SIP/Zanzibar)