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 <[EMAIL PROTECTED]> 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 < [EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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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