Hi, i’m having issue when calling out from my gateway.
The problem comes out when i’m trying to call:
when i’m calling the number from my phone i’m getting ~6 sec delay before my phone rings.
It’s not a big problem for me, but when incoming call is being forwarded through one of gateways this 6 seconds disorient the caller and he can hang up his phone, ant it’s not good.
This is asterisk log, which appears instantly after placing a call:
== Using SIP RTP CoS mark 5
– Executing [705XXXXXXXXXX@call-out:2] Dial(“SIP/212-000004ee”, “SIP/705/XXXXXXXXXX”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/705/705XXXXXXXXXX
– SIP/705-000004ef is making progress passing it to SIP/212-000004ee
0xb6f44f78 – Probation passed - setting RTP source address to XXX.XX.XX.207:18910
And after ~6 seconds when phone starts to call log continues with this:
> 0xb6f7b5c0 -- Probation passed - setting RTP source address to XXX.XX.XX.XX:23032
> == Spawn extension (call-out, 705XXXXXXXXXX, 2) exited non-zero on 'SIP/212-000004ee'
my sip.conf:
[gateway](!)
type=friend
context=incoming
host=dynamic
secret=XXXXXXXXXXXXXXXX
nat=no
trustrpid=yes
qualify=yes
canreinvite=no
callgroup=1
pickupgroup=1
call-limit=3
dtmfmode=RFC2833
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=g723
allow=g722
[705](gateway)
callerid="GW2"
Could it be wrong setup of gateway or asterisk?
Any suggestions?
Regards,
Nufay.