James H. Cloos Jr.
2003-Jun-03 19:22 UTC
[Asterisk-Users] ata186 and 9 for outgoing line type dialplans
I tried putting this as the ata's dailplan:
*St4-|#St4-|9|^9t4>$.-
this is sip.conf
[ata2001]
type=friend
username=ata2001
secret=SoMeSeCrEt
host=dynamic
context=fromata
canreinvite=no
and this in extensions.conf
[fromata]
ignorepat => 9
exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
exten => _91888NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
exten => _91877NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
exten => _91866NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
exten => _91800NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
Dialing 91800nxxxxxx does attempt a call to 1800nxxxxxx via iax to
iaxtel, but I do not get dialtone after the 9.
I presume I have to get * to send dialtone via rtp (or is there some
sip packet that will tell the ata to continue dialtone)? How is that
done to a sip extension?
The calls to iaxtel don't seem to be working. I'm also not able to
get Dial(IAX2/guest@misery.digium.com/s@default) to connect, even
though it was working well earlier today.... (I get no reply packets
at all, for either misery.digium.com or iaxtel.com.)
-JimC
Steven Critchfield
2003-Jun-04 02:49 UTC
[Asterisk-Users] ata186 and 9 for outgoing line type dialplans
You problem is that in SIP, your dialtone is provided by the SIP device. Asterisk is not sending dialtone. Your ATA is doing the dialtone the whole time. What you may want to do to make life really easy is to set up the ata186 to work like a hotline phone. There was comments on the list about this. If the ata186 dials asterisk as soon as the phone goes off hook, then asterisk will be sending dialtone. Then your dialplan on the ata won't get in the way. On Tue, 2003-06-03 at 21:22, James H. Cloos Jr. wrote:> I tried putting this as the ata's dailplan: > > *St4-|#St4-|9|^9t4>$.- > > this is sip.conf > > [ata2001] > type=friend > username=ata2001 > secret=SoMeSeCrEt > host=dynamic > context=fromata > canreinvite=no > > and this in extensions.conf > > [fromata] > ignorepat => 9 > exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) > exten => _91888NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) > exten => _91877NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) > exten => _91866NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) > exten => _91800NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) > > > Dialing 91800nxxxxxx does attempt a call to 1800nxxxxxx via iax to > iaxtel, but I do not get dialtone after the 9. > > I presume I have to get * to send dialtone via rtp (or is there some > sip packet that will tell the ata to continue dialtone)? How is that > done to a sip extension? > > The calls to iaxtel don't seem to be working. I'm also not able to > get Dial(IAX2/guest@misery.digium.com/s@default) to connect, even > though it was working well earlier today.... (I get no reply packets > at all, for either misery.digium.com or iaxtel.com.) > > -JimC > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- Steven Critchfield <critch@basesys.com>