Bridging 2 SIP channels: no audio UNLESS playback occurs 1st

Hello All,

I have been helping installing quite a bunch of Asterisk PBX compiled from source, usually with POTS or PRI/ISDN connection to the carrier. Recently I have configured a lot of SIP-trunk-based carriers on PBX, and start seeing the same issue on all those installations. I am wondering if there is something in the configuration I am missing, or if this is a typical behavior when using SIP-trunking carriers. The problem occurs when trying to take an inbound call, place an outbound call, and bridge the two together (for example when someone wants their PBX to send a call to an outside answering service during certain hours).

A typical dialplan will look like this:

[from-voip-carrier] exten => _X.,1,NoOp() same => n,Answer() same => n,Dial(SIP/8005556666@voipcarrier) same => n,Hangup()

When an inbound call arrives, an outbound call will be placed, and the two SIP channels will be bridged. However, no audio will be heard from either side.

However, if using the following code, audio will work perfectly fine both ways: (additional line with a Playback)

[from-voip-carrier] exten => _X.,1,NoOp() same => n,Answer() same => n,Playback(hello-world) same => n,Dial(SIP/8005556666@voipcarrier) same => n,Hangup()

Some additional information:
[] there is no NAT involved in any of the setup. All the Asterisk setup are done with a nic configured with a public IP.
] Internally, Polycom IP phones are utilized. The problem is the same if a polycom phone is setup to forward calls.
[*] I tried with and without the “Answer()” line, the problem is the same.

Any advice on this will be greatly appreciated !


What is the relevance of the reference to the internal phones.

It is difficult to see what difference playback will make, especially as using answer will force Asterisk to originate audio for the call progress tones.

You need to obtain SIP traces of the calls, particularly the SDP.

Try This =>

exten => _X.,1,Dial(SIP/${EXTEN:0},30,r)