I upgrade Asterisk from 1.2.X to 1.4.0 and after that when I make a call via VoipDiscount (g729 buy from Digum) on my internel SIP phone (ulaw) I got silent.
When I make same call with internel analog phone connected to Digium TDM4xx everything is ok.
When I make same call with external (gsm) trunk - everyting is ok.
When I make same call with mISDN - everything is ok.
Only when translate beetwen g729 and SIP a have silent.
Problem is because I do not have any error or message. Asterisk looks work fine, but I have silent in my phone.
In previous all verion this working perfect.
The g729 codeck I have for 1.4 version.
dembinski3*CLI> show g729
0/0 encoders/decoders of 3 licensed channels are currently in use
When I make a call the value 0/0 not change.
But:
dembinski3*CLI> core show channels
Channel Location State Application(Data)
SIP/voipdiscount01-0 (None) Up Bridged Call(SIP/grandstream01
SIP/grandstream01-b7 848228339677@default Up Dial(SIP/48228339677@voipdisco
2 active channels
1 active call
I found the problem.
In sip.conf in this version MUST be set does the trunk/client are behind nat or not. In previous version Asterisk done this by himself, or just it works witout any setting.