I have a trunk set up to send long distance calls out to an Audiocodes SIP gateway. My trunk dial plan looks like this…
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN}@10.10.1.202)
When I dial number that matches that pattern from one of my IP phones, it is taking 40 seconds for Asterisk to actually send the SIP invite to the gateway.
I have observed the following from the Asterisk CLI. When I initiate the call from the IP phone, immediately see the following…
== Using SIP RTP CoS mark 5
-- Executing [918642864941@users:1] Dial("SIP/1000-0000010a", "SIP/918642864941@10.10.1.202") in new stack
And then 40 seconds later…
== Using SIP RTP CoS mark 5
-- Called 918642864941@10.10.1.202
-- SIP/10.10.1.202-0000010b is ringing
-- SIP/10.10.1.202-0000010b answered SIP/1000-0000010a
-- Native bridging SIP/1000-0000010a and SIP/10.10.1.202-0000010b
== Spawn extension (users, 918642864941, 1) exited non-zero on 'SIP/1000-0000010a'
I have done some research on my own and found that 40 second is the default master timeout for Asterisk. I am just not sure what it is waiting for.
I am having it dial by IP address (SIP/${EXTEN}@10.10.1.202) so I am not sure what DNS would have to do with it. The call is going through successfully; it just takes 40 seconds to do so.
[quote=“david55”]
The easiest approach to this is probably to attach a debugger and then run thread apply all bt. That will allow you to work out where it is waiting.[/quote]
Forgive my ignorance, as I am still kinda new to Asterisk, but what is the best way of doing that?
It looks like this going to require me to recompile Asterisk which I am not sure how to do on an AsteriskNOW box. Is there any other way to debug this?
I have been able to fix this based on david55’s assumption that the delay was coming from a reverse DNS lookup. Instead having is dial an IP address, I created a context in sip.conf that looks like this…
[audiocodes]
;FXO Gateway for PSTN
type=friend
host=10.10.1.166
context=trunkld
and my trunk dialplan looks like this now…
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN}@audiocodes,20)
and it works like a charm. Thanks for the nudge in the right direction david55