Asterisk 15 Dial() application

Asterisk Asterisk Support
1

Hi,

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

2

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

3

Hi,

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.

Answer()
Dial()

Is working,

But in the same time

Noop()
Dial()

Dont and im not sure why… :frowning:

4

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

5

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,

6

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.

7

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!