Tim, this is the way I have Gafachi set up in sip.conf and works well with channels that have an ATA attached to it but not the virtual one. I have changed the host in extensions.conf to the xxxxxxxxxxxx.sip.gafachi.com. But I have calls on the server and cannot restart it yet. I'll keep you posted and thanks for the feedback.
[gafachi-out]
type=peer
secret=xxxxxxxxxx
username=xxxxxxxxx
fromuser=xxxxxxxxxxxxx
fromdomain=xxxxxxxxxxxxxxx
host=xxxxxxxxxxxx.sip.gafachi.com
;usereqphone=yes ; This provider requires ";user=phone" on URI
;nat=yes
rtptimeout=60
dtmfmode=rfc2833
type=peer
secret=xxxxxxxxxx
username=xxxxxxxxx
fromuser=xxxxxxxxxxxxx
fromdomain=xxxxxxxxxxxxxxx
host=xxxxxxxxxxxx.sip.gafachi.com
;usereqphone=yes ; This provider requires ";user=phone" on URI
;nat=yes
rtptimeout=60
dtmfmode=rfc2833
-------------- Original message --------------
From: "Tim St. Pierre" <[EMAIL PROTECTED]>
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--- Begin Message ---Do you have gafachi-o in your sip.conf? Since it's not a valid host name, you need to have an entry in sip.conf to tell asterisk how to make a call to gafachi-o.That's why it is telling you "No such host". On September 8, 2006 12:57, [EMAIL PROTECTED] wrote: > It sounds like a good idea, I tried it and get this error > > Sep 8 09:52:17 WARNING[27193]: chan_sip.c:1968 create_addr: No such host: > gafachi-o Sep 8 09:52:17 NOTICE[27193]: app_dial.c:1011 dial_exec_full: > Unable to create channel of type 'SIP' (cause 3 - No route to destination) > > In Extensions.conf I have > exten => 4305,1,Dial(SIP/[EMAIL PROTECTED]) ; permit transfer > > In Sip.conf I have > [4305] > type=friend > user=4305 > secret=xxxxxxxx > ;context=from-sip > callerid= > host=dynamic > nat=yes > canreinvite=no > dtmfmode=rfc2833 > ;incominglimit=1 > ;[EMAIL PROTECTED] > ;disallow=all > ;allow=ulaw > ;allow=alaw > ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! > > > > -------------- Original message -------------- > From: "William Piper" <[EMAIL PROTECTED]> > > whatever the did is needs to be put in the extensions.conf & told to dial > your cellphone. Example: > > exten => _011123445566,1,Dial,SIP/[EMAIL PROTECTED] > > assuming that your using a SIP carrier, replace 1234567890 with your > cellphone & 1.2.3.4 with the carrier's IP or carriers context name in > sip.conf. > > bp > > On 9/8/06, [EMAIL PROTECTED] <[EMAIL PROTECTED] > wrote: > I'm using it for virtual numbers. I have international virtual number from > a DID provider and want to forward it to my cell phone. > > In Sip.conf I have the channel > > [4305] > type=friend > user=4305 > secret=xxxxxxxx > ;context=from-sip > callerid= > host=dynamic > nat=yes > canreinvite=no > dtmfmode=rfc2833 > ;incominglimit=1 > ;[EMAIL PROTECTED] > ;disallow=all > ;allow=ulaw > ;allow=alaw > ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! > > and in extensions.conf I have > > exten => 4305,1,Dial(SIP/4305,120,rt) ; permit transfer > > This had worked in the past when I forwarded it through the Linksys ATA but > now have run out of ATA's. > > > -------------- Original message -------------- > From: "Tim St. Pierre" < [EMAIL PROTECTED]> > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED]
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