Timeout in Dial application

Asterisk Asterisk Support
1

Hi,

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

2

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.

3

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.

4

Thanks very much for detailed info how it works!,

best regards, KB