Well here is my scenario,<br>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
<br>Caller called user1 did1(123)-----send bellcore1<br>Caller called user1 did2(456)-----send bellcore2<br>and so on....<br>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.
<br><br>So the first problem:<br>For sending the dnid i used the dial application like this:<br>(1)--Dial(SIP/${DNID}@user1) incase when the user is using asterisk as a peer<br>(2)--Dial(SIP.user1) incase when the user is using sipura as a peer
<br>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.
<br><br>The second Problem:<br>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 ${DNID} variable, and i just came to know this problem. I dont know why this i happening. if you know plz help.
<br><br>So.....any ideas<br><br><br><br><div><span class="gmail_quote">On 4/10/07, <b class="gmail_sendername">Karl J. Vesterling</b> <<a href="mailto:kjv@ken-ton.com">kjv@ken-ton.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000">
<font size="+1">There is a difference bwtween the SPA-941 which is a
standalone desktop phone w/ Speakerphone, and your SPA2100<br>
<br>
What header value are you trying to set, and why?<br>
<br>
Because the example I gave is for making the SPA-941 an intercom, and
also requires some changes in it's settings within the phones web
interface. So of course my example isn't going to anything for your
SPA-2100 except give you an example of how to set a SIP header.<br>
<br>
So now my question becomes, "What is it that you are attempting to do
by setting the SIP Header?"<br>
<br>
Best Regards,<br>
Karl J. Vesterling<br>
<br>
</font><br>
Rizwan Hisham wrote:
<blockquote cite="http://mid4809880c0704100438h54ec5e23uaabd30dfbc4b0640@mail.gmail.com" type="cite"><div><span class="e" id="q_111db651105006ac_1">I have discovered another thing,<br>
<br>
exten=> 123,1,Dial(SIP/abc/${EXTEN},,Tt)
<br>
<div>exten=> 123,2,hangup<br>
<br>
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.<br>
<br>
for sipura to ring we have to use the following extension, without
${EXTEN} variable<br>
<br>
Dial(SIP/abc,,Tt)<br>
</div>
<br>
<br>
<br>
<div><span class="gmail_quote">On 4/9/07, <b class="gmail_sendername">Karl
J. Vesterling</b> <<a href="mailto:kjv@ken-ton.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">kjv@ken-ton.com</a>>
wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000"><font size="+1">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.<br>
<br>
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<br>
<br>
Back to work, I'll check in on this thread later this evening.<br>
<br>
</font><br>
Rizwan Hisham wrote:
<blockquote cite="http://mid4809880c0704090802j2151b98bkd95cf5c2e8315cb7@mail.gmail.com" type="cite">
<div><span>I dont understand
it<br>
<br>
<font size="+1">Set(__SIPADDHEADER=Call-Info:\;answer-after=0);<br>
<br>
whats it doing here?<br>
<br>
</font><br>
<div><span class="gmail_quote">On 4/9/07, <b class="gmail_sendername">Karl
J. Vesterling</b> <<a href="mailto:kjv@ken-ton.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
kjv@ken-ton.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000"><font size="+1">I
struggled
with this one too, try this:<br>
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);<br>
<br>
I use the above for intercom w/ Sipura SPA-941 and it works.<br>
Asterisk 1.2.17 / extensions.ael<br>
<br>
<br>
</font><br>
Rizwan Hisham wrote:
<blockquote cite="http://mid4809880c0704090629v769a7140mb30adfb3653a35c6@mail.gmail.com" type="cite">
<div><span>I have tried it,
it doesnt work<br>
<br>
<div><span class="gmail_quote">On 4/9/07, <b class="gmail_sendername">Hermann
Wecke</b> <<a href="mailto:hermann@wecke.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">hermann@wecke.com</a>>
wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Rizwan
Hisham wrote:<br>
> is there anyway i can set SIP_HEADER(To) to the value i like?<br>
<br>
If voip-info is correct, you can read, but you can't change.<br>
<a href="http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
</a><br>
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