Timeout in Dial application


I noticed that users which are logged between start of the Dial() application in dialplan, but before end of the time set as a timeout are not always connected (not always receive invite), so we have a case when:

  • asterisk is trying to Dial(SIP/123,30), qualify is disabled, so there are no OPTIONS,
  • after 10 secs SIP/123 is logging (register->unauth->reg with digest->ok) from new location,
  • still at least 15 secs left, but there is no retransmission of INVITE to logged endpoint,

Is there any way to force retransmission of invite to such endpoint? If not - can i somehow change INVITE retransission period to force another INVITE in such case?

Im using chan_sip,

Best regards, KB

chan_sip will have latched the IP address before it sends the first INVITE. I imagine that chan_pjsip does the same, as re-translating for every transmission is expensive.

In more detail, chan_sip copies the information in the peer data structure into the private (technology dependent) channel data structure at the time that the channel is created.

Thanks very much for detailed info how it works!,

best regards, KB