Asterisk 13 no audio on outgoing call

Asterisk Asterisk SIP
1

Hello,

we are running Asterisk 13.29.2 and i have the following Dialplan (shortened):

XXX => 1,SipAddHeader(P-Preferred-Identity: <sip:+xxx@company>)
same => n,Dial(SIP/CARRIER/00xxxxxxxxxx,120,oR)

now this should be something like a call forwarding, so i call and get connected with another external number

this works SOMETIMES

what i figured out with sngrep:
the outgoing call has the following events:

  • 100 Trying
  • 183 Session Progress (SDP)
  • 180 Ringing (SDP)

but sometimes the order of events change to:

  • 100 Trying
  • 180 Ringing (SDP)
  • 183 Session Progress (SDP)

and if this happens, there is no audio, so i can not hear and the other can also not hear

does anybody know what the problem could be?

2

It is probably not worth debugging as Asterisk 13 is several years beyond end of life and chan_sip is no longer in the latest version, and has been, effectively, unsupported, for several years.

Generally we want the Asterisk full log, with, in your case, “sip set debug on” in effect, so see the full contents of the SIP exchange, not just a summary.

When you say no audio, are you just talking about ring back tone. I think the second sequence will stop Asterisk from generating any, as it is basically cancelling ring back.

3

thanks for your response
no audio means, that at the time the channels get bridged, neither the ring-tone of the called nor the voice is hearable when picked up (so no sound in both directions)

4

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