Asterisk 15 Dial() application


Recently i made an upgrade from Asterisk 13 to Asterisk 15. I noted that something changed with Dial() function. Im using chan_sip.

My problem is that dialplan:

exten => _3.,1,Verbose(“Calling exten SIP/${EXTEN} from ${CALLERID(num)}”)
exten => _3.,n(dial),Dial(SIP/${EXTEN},20)
exten => _3.,n,Hangup()

Stopped working. Call is established, but there is no sound in both endpoints. If i will add Answer() before dial i have sound in one direction, but second one is still dead,

Fully working solution is to add sound played to called party after a call:

exten => _3.,1,Verbose(“Calling exten SIP/${EXTEN} from ${CALLERID(num)}”)
exten => _3.,n,Answer()
exten => _3.,n(dial),Dial(SIP/${EXTEN},20, A(test) )
exten => _3.,n,Hangup()

Why i have such problems with Asterisk 15 simple bridge? Any1 spotted such behaviour?

Best regards, KB

Have you done any investigation to see if media is actually flowing using rtp set debug on?


For sure it does.

Got traces from both cases. With and without Answer() before Dial and both look the same.

It could be important - im using configuration behind nat (rport,comedia). Only thing which i didnt checked is actual RTP content. This configuration is using srtp, so i cannot check rtp content in whireshark, only packages

And im stuck.


Is working,

But in the same time


Dont and im not sure why… :frowning:

Traces from where? From rtp set debug on? From elsewhere? Is Asterisk itself behind NAT?

Yes, from

rtp set debug on

Asterisk is not behind nat, but the clients which are connecting are. Additionally im using turn.

Honestly i doub that this is a routing problem. Previously on the same machine i was using asterisk 13. With the same routers and the same config files. It stopped after upgrade to 15.

Best regards,

You keep adding additional data here. You need to provide the COMPLETE network layout, what you’re doing, SIP traces, the rtp set debug on information.

I understand that it was working before and now it isn’t, but that alone doesn’t provide an explanation of why. The problem has to be isolated.

Found it. Just FYI

Previously i was using chan_sip. When i switched to chan_pjsip everyhing started to work.
In asterisk 15 something changed with chan_sip.

But my problem is solved. Thanks very much for help!

Have a nice day!