I have a tricky situation. I have a polycom phone with number 3254103. I have
configured the phone to forward to a new number, 18059999999.
Here's my dialplan:
exten => 3254103,1,Dial(SIP/3254103,10,tr)
exten => 18059999999,1,Dial(SIP/11101553818059999999@proxy2,40,tr)
When Asterisk dials 3254103, here's what comes up on the console:
hestia*CLI>
-- Executing Dial("SIP/2944093-6935",
"SIP/3254103|10|tr") in new stack
-- Called 3254103
-- Got SIP response 302 "Moved Temporarily" back from
xxx.187.128.19
-- Now forwarding SIP/2944093-6935 to
'Local/18059999999@betty_start' (thanks to SIP/3254103-47ab)
-- Executing Dial("Local/18059999999@betty_start-5e7d,2",
"SIP/11101553818059999999@proxy2|40|tr") in new stack
-- Called 11101553818059999999@proxy2
-- Local/18059999999@betty_start-5e7d,1 is ringing
-- SIP/proxy2-5adc is ringing
-- SIP/proxy2-5adc is making progress passing it to
Local/18059999999@betty_start-5e7d,2
-- Nobody picked up in 10000 ms
== Spawn extension (betty_start, 18059999999, 1) exited non-zero on
'Local/18059999999@betty_start-5e7d,2'
== Auto fallthrough, channel 'SIP/2944093-6935' status is
'NOANSWER'
hestia*CLI>
You can see that the phone tells Asterisk that the number has been forwarded.
Asterisk re-enters the dialplan logic and tries to contact the forwarded number.
That's all great...
We have a problem of timeouts here. In this situation, Asterisk drops all call
flow at 10 seconds, which was the timeout set for the original number, 3254103,
eventhough it has now re-entered the dialplan logic, dialling a new number with
a timeout of 40 seconds. It's as if the timeout of the original number sets
the timeout for the forwarded call.
Shouldn't the timeout used to dial 18059999999 be 40 seconds? Why does
Asterisk use the original timeout of 10s? This causes all sorts of problems.
Thanks,
Doug.
