[asterisk-users] sip_header=value?

Karl J. Vesterling kjv at ken-ton.com
Tue Apr 10 05:01:34 MST 2007


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
> <mailto: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
>>     <mailto: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
>>>         <mailto: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
>>>             <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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>>
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>>
>>
>>     -- 
>>     Regards
>>     Rizwan Hisham
>>     Software Engineer
>>     ------------------------------------------------------------------------
>>
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>
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>
>
> -- 
> Regards
> Rizwan Hisham
> Software Engineer
> ------------------------------------------------------------------------
>
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