[asterisk-users] regexten & regcontext broken for SIP?

[Andrew Joakimsen]

I was trying:

[000000000000L1]
Telco Systems AC-211 00-00-00-00-00-00 Line 1
type=friend
secret=password
host=dynamic
context=envision-out
callerid=0000000000
nat=yes
qualify=5000
allow=all
disallow=g723
disallow=g729
dtmfmode=rfc2833
canreinvite=no
mailbox=0000000000
regcontext=sip_autoreg
regexten=10000000000

the calls come into the context inboundcalls...


[inboundcalls]
include => sip_autoreg
exten => _1xxxxxxxxxx,2,Set(dialto=${DB(regexten/${EXTEN})})
exten => _1xxxxxxxxxx,3,Dial(SIP/${dialto},25,r)
exten => _1xxxxxxxxxx,4,VoiceMail(${EXTEN:1}|u)
exten => _1xxxxxxxxxx,5,Hangup()


[vm]
exten => _1xxxxxxxxxx,1,VoiceMail(${EXTEN:1}|b)

On 11/2/06, Watkins, Bradley <Bradley.Watkins at compuware.com> wrote:
>
>  Can either or both of you post the relevant sections of your sip.conf and
> extensions.conf?
>
> - Brad
>
>  ------------------------------
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Andrew Joakimsen
> *Sent:* Thursday, November 02, 2006 1:51 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] regexten & regcontext broken for SIP?
>
> I am having the same issues. Did you ever file a bug report?
>
> On 10/6/06, Philipp von Klitzing < klitzing at pool.informatik.rwth-aachen.de>
> wrote:
> >
> > Hi ho,
> >
> > is there anyone out here that is making use of the regcontext and
> > regexten settings in sip.conf? I've tried this on two Asterisk boxes
> > (1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1
> > being created upon SIP client registration, "show dialplan xxx" reveals
> > no change.
> >
> > And yes, I have also read and checked bug 7144; if I go down that route
> > and no SIP client is registered I get a CLI warning that my standard
> > context tries to include an empty context - go figure...
> > http://bugs.digium.com/view.php?id=7144
> >
> > So, do I need to file a bug report, or is it working OK for others?
> >
> > Cheers, Philipp
> >
> > P.S.: Of course I am aware of this Wiki page:
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+regcontext
> >
> >
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