a2billing is your default context in the a2billing.conf file from setup,
if you have changed this, and also changed your context for your
card/sip entry to callingcard you should, when you dial from your phone,
be using the callingcard context. Please check your SIP entry for your
phone in additional_a2billing_sip.conf to ensure that your context is
set to callingcard, not a2billing (default), and then reload you sip
config from the CLI.
Your phone that is registered sounds like it is not assigned to that
context by default.
J-
On Sun, 2007-02-18 at 20:38 +0200, Ioan Indreias wrote:
> Hello,
> I'm not familiar with A2billing but for me it is strange that you "dial"
> SIP/777 - 777 should be an extension.
>
> Could you post your "user" context - or at least the one which direct
> you to:
> Dial("SIP/9614-3896", "SIP/777|200|rt")
>
> Best regards,
> ## nini @ www.modulo.ro ##
>
>
>
> [EMAIL PROTECTED] wrote:
> > Thanks Rob, that helped a little bit but now getting a different kind
> > of error:
> >
> > -- Executing Dial("SIP/9614-3896", "SIP/777|200|rt") in new stack
> > Feb 18 06:00:30 NOTICE[12236]: app_dial.c:1011 dial_exec_full: Unable
> > to create channel of type 'SIP' (cause 3 - No route to destination)
> > == Everyone is busy/congested at this time (1:0/0/1)
> > == Auto fallthrough, channel 'SIP/9614-3896' status is 'CHANUNAVAIL'
> >
> >
> > -------------- Original message --------------
> > From: Rob Hillis <[EMAIL PROTECTED]>
> > I guess the obvious question would be whether the "callingcard"
> > context is included into the context that the call is coming
> > from. That's the usual reason for a failure like this.
> >
> >
> > [EMAIL PROTECTED] wrote:
> >> I have followed all the install note for A2billing and have
> >> everything installed and configured and my asterisk works except
> >> the callingcard application.
> >> Added the following
> >> [callingcard]
> >> ; CallingCard application
> >> exten => 777,1,Answer
> >> exten => 777,2,Wait,2
> >> exten => 777,3,DeadAGI,a2billing.php
> >> exten => 777,4,Wait,2
> >> exten => 777,5,Hangup
> >> I am using 777 as the calling card application. when I call that
> >> extension, instead of getting " please enter you pin number" it
> >> fails and this is the output from the cli:
> >> -- Executing Dial("SIP/9614-e7ba", "SIP/777|200|rt") in new stack
> >> Feb 18 05:03:38 WARNING[11725]: chan_sip.c:1968 create_addr: No
> >> such host: 777
> >> Feb 18 05:03:38 NOTICE[11725]: app_dial.c:1011 dial_exec_full:
> >> Unable to create channel of type 'SIP' (cause 3 - No route to
> >> destination)
> >> == Everyone is busy/congested at this time (1:0/0/1)
> >> == Auto fallthrough, channel 'SIP/9614-e7ba' status is
> >> 'CHANUNAVAIL'
> >> Any Help will be greatly appreciated.
> >>
> >> ------------------------------------------------------------------------
> >>
> >> _______________________________________________
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> >>
> >
> >
> > ------------------------------------------------------------------------
> >
> > Subject:
> > Re: [asterisk-users] chan_sip.c:1968 create_addr: No such host:
> > From:
> > Rob Hillis <[EMAIL PROTECTED]>
> > Date:
> > Sun, 18 Feb 2007 12:43:59 +0000
> > To:
> > Asterisk Users Mailing List - Non-Commercial Discussion
> > <asterisk-users@lists.digium.com>
> >
> > To:
> > Asterisk Users Mailing List - Non-Commercial Discussion
> > <asterisk-users@lists.digium.com>
> >
> >
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> > ------------------------------------------------------------------------
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