[asterisk-users] sip_header=value?

[Rizwan Hisham]

Well here is my scenario,
Our users have option to register with one Primary did and 5 secondary dids
for the purpose of distinctive-ring/did-based-routing. If a user is
registered with us and he is using sipura, then we have to send 6 different
bellcores in alert into sip header for his different dids like this
Caller called user1 did1(123)-----send bellcore1
Caller called user1 did2(456)-----send bellcore2
and so on....
If the user is using asterisk to register with us, then we also have to send
the dnid so that when the user receives the dialed num, he has to decide to
route that call to which extension based on dnid.

So the first problem:
For sending the dnid i used the dial application like this:
(1)--Dial(SIP/${DNID}@user1) incase when the user is using asterisk as a
peer
(2)--Dial(SIP.user1) incase when the user is using sipura as a peer
i wanted to use (1) for dialing sipura also but it doesnt work, so i dial
like (1) if user is using asterisk and (2) for every other device. I am
still finding a way to solve this problem so that i dont have to check the
called user's useragent for every call. I was trying to find a way to send
the dnid in some header field. thats why i started this thread.

The second Problem:
Asterisk doesn't set the ${DNID} variable to the dialed extension extension
num. all of my system for setting the bellcore and sending the dnid is based
on ${DNID} variable, and i just came to know this problem. I dont know why
this i happening. if you know plz help.

So.....any ideas



On 4/10/07, Karl J. Vesterling <kjv at ken-ton.com> wrote:
>
>  There is a difference bwtween the SPA-941 which is a standalone desktop
> phone w/ Speakerphone, and your SPA2100
>
> What header value are you trying to set, and why?
>
> Because the example I gave is for making the SPA-941 an intercom, and also
> requires some changes in it's settings within the phones web interface.  So
> of course my example isn't going to anything for your SPA-2100 except give
> you an example of how to set a SIP header.
>
> So now my question becomes, "What is it that you are attempting to do by
> setting the SIP Header?"
>
> Best Regards,
> Karl J. Vesterling
>
>
> Rizwan Hisham wrote:
>
> 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
> > > >
> > > > _______________________________________________
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> > >
> > >
> > >
> > > --
> > > Regards
> > > Rizwan Hisham
> > > Software Engineer
> > >
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> > --
> > Regards
> > Rizwan Hisham
> > Software Engineer
> >
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> Regards
> Rizwan Hisham
> Software Engineer
>
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-- 
Regards
Rizwan Hisham
Software Engineer
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