[asterisk-users] sip_header=value?

[Rizwan Hisham]

I have discovered another thing,

exten=> 123,1,Dial(SIP/abc/${EXTEN},,Tt)
exten=> 123,2,hangup

If the user is using xlite to register with my asterisk server, then we are
able to call him using the above axtension, and if the user is using sipura
the the above extension does not dial the user instead it displays a
congestion message as before, maybe there is a problem in sipura firmware. I
am using "Linksys/SPA2100-3.3.6". any ideas why is sipra behaving like this.

for sipura to ring we have to use the following extension, without ${EXTEN}
variable

Dial(SIP/abc,,Tt)



On 4/9/07, Karl J. Vesterling <kjv at ken-ton.com> wrote:
>
>  Apparently it sets a SIP_HEADER variable named "Call-Info" to a value of
> "answer-after=0" effectively telling the Sipura to answer the call and put
> it through to speakerphone.
>
> I will say that extensions.ael is a bit different from regular line based
> extensions.conf in that I seem to have to escape all sorts of stuff with
> the \ character that I don't have to in extensions.conf
>
> Back to work, I'll check in on this thread later this evening.
>
>
> Rizwan Hisham wrote:
>
> I dont understand it
>
> Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
>
> whats it doing here?
>
>
> On 4/9/07, Karl J. Vesterling < kjv at ken-ton.com> wrote:
> >
> > I struggled with this one too, try this:
> > Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
> >
> > I use the above for intercom w/ Sipura SPA-941 and it works.
> > Asterisk 1.2.17 / extensions.ael
> >
> >
> >
> > Rizwan Hisham wrote:
> >
> > I have tried it, it doesnt work
> >
> > On 4/9/07, Hermann Wecke <hermann at wecke.com> wrote:
> > >
> > > Rizwan Hisham wrote:
> > > > is there anyway i can set SIP_HEADER(To) to the value i like?
> > >
> > > If voip-info is correct, you can read, but you can't change.
> > > http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
> > > _______________________________________________
> > > --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
> > >
> >
> >
> >
> > --
> > Regards
> > Rizwan Hisham
> > Software Engineer
> >
> > ------------------------------
> >
> > _______________________________________________
> > --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
> >
> >
>
>
> --
> Regards
> Rizwan Hisham
> Software Engineer
>
> ------------------------------
>
> _______________________________________________
> --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
>
>


-- 
Regards
Rizwan Hisham
Software Engineer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070410/0af06ace/attachment.htm