I have read around and have only found similar problems, but not exactly as this:
I have a sip trunk [vitelity-inbound] in sip.conf that goes directly to the [a2billing-in] context in extensions.conf, and a SIP extension [5551231234] in extensions.conf:
Code:
sip.conf=
[vitel-inbound]
username=XXXXXX
type=friend
secret=XXXXXXXX
insecure=very
host=inbound3.vitelity.net
disallow=all
context=a2billing-in
canreinvite=no
allow=ulaw
allow=gsm
extensions.conf=
[a2billing-in]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php|2|did)
exten => _X.,n,Hangup
[5551231234]
type=friend
username=5551231234
accountcode=5551231234
regexten=5551231234
callerid=15551231234
amaflags=billing
secret=XXXXXXXX
nat=1
dtmfmode=RFC2833
qualify=1
canreinvite=yes
disallow=all
allow=ulaw
allow= alaw
allow= gsm
allow= g729
host=dynamic
context=a2billing
regseconds=0
cancallforward=yes
I have two did destinations set up for the 5551231234 did. They are:
Code:
#1
SIP/5551231234
and
#2
15551112222
When I dial the DID from an external PSTN line it goes all the way through a2billing.php. When it tries to ring SIP/5551231234 it says
Code:
Jun 20 17:47:36 NOTICE[10219] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Jun 20 17:47:36 VERBOSE[10219] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
Jun 20 17:47:36 DEBUG[10219] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.
And then it immediatly SUCCESSFULLY dials the next destination, which is going out through a voip trunk to a PSTN destination.
Code:
AGI Script Executing Application: (Dial) Options: (SIP/vitel-outbound/15551112222|60|HRrL(17967000:61000:30000))
Why does it not dial the SIP/5551231234 extension when I dial in externally from the 5551231234 DID, but it will successfully dial the (SIP/vitel-outbound/15551112222) ???
I can dial out from that sip extension fine!
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