I'm trying to complete software which does all call logic via AMI on it's own using Asterisk only as interface to VOIP, SIP/GSM. Almost everything works great, but...:
Here is my scenario: - incoming call is forwarded to announcement and then to MOH forever - my app decides which extensions to dial (7777) using AMI Action: Originate - once somebody picks up on extension, his/her channel (SIP/306-xxxxx for example) is bridged with waiting call's channel using AMI Action: Bridge
Until this point everything is working fine, both connected parties can hear each other, recording on demand works. All is fine.
Now I'm trying to make assisted transfer to another extension (Atxfer) using AMI on one of the bridged channels. And it doesn't work. I got couple of ami events about DTMF's on a channel (audio is muted while they are played). Every DTMF digit couses quick Bridge:unlink and Bridge:link event on AMI.
I tried to change dtmfmode, upgrade from asterisk 1.8 to 11 (asterisk now) and it always was the same.
While having this problems with Atxfer blind transfer on those channels works (using AMI Action: Redirect).
full log shows nothing something like this:
[2013-11-11 20:24:57] DEBUG[9457]: features.c:3740 feature_interpret: Feature interpret: chan=SIP/306-00000017, peer=SIP/GTS-00000016, code=*2, sense=1, features=0, dynamic=apprecord#apprecord