I am using asterisk 1.6 with G729 in pass through mode. I have converted my ivrs to g729 format and they all play without any issue. My problem is that when i dial out a number, asterisk gets sip notification “ringing” but the calling party does not hear anything until the call is answered. I also get the following warnings in log:
app_dial.c: – SIP/192.168.0.138-0000088b is ringing
channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
indications.c: Unable to set ‘SIP/192.168.0.142-0000088a’ to signed linear format (write)
channel.c: Unable to handle indication 3 for ‘SIP/192.168.0.142-0000088a’
I guess the problem is that asterisk generates ringing tone itself which is in slin codec. Then it tries to transcode it to g729 which is not possible since i dont have g729 codec installed on my system. Can there be a solution to this problem?
I dont want to use r option of Dial application because as it says on voip-info.org
"r" will force Asterisk to generate ring tones, even if it is not appropriate. For example, if you used this option to force ringing but the line was busy the user would hear “RING RIBEEP BEEP BEEP” (thank you tzanger), which is potentially confusing and/or unprofessional.
Please help me with this problem. Thanks