[asterisk-users] Call Forwarding in SIP.conf

[Tim St. Pierre]

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, broadbandvoice at comcast.net 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/1234567890 at gafachi-o)      ; 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
> ;mailbox=1234 at default
> ;disallow=all
> ;allow=ulaw
> ;allow=alaw
> ;allow=g723.1                   ; Asterisk only supports g723.1 pass-thru!
>
>
>
> -------------- Original message --------------
> From: "William Piper" <william.piper at gmail.com>
>
> whatever the did is needs to be put in the extensions.conf & told to dial
> your cellphone. Example:
>
> exten => _011123445566,1,Dial,SIP/1234567890 at 1.2.3.4
>
> 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, broadbandvoice at comcast.net <broadbandvoice at comcast.net > 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
> ;mailbox=1234 at default
> ;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" < tim at communicatefreely.net>
>
> > _______________________________________________
> > --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://5101@communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
tim at communicatefreely.net
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 187 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060908/b5c0ccbc/attachment.pgp