I have discovered another thing,<br><br>exten=> 123,1,Dial(SIP/abc/${EXTEN},,Tt)
<br><div id="mb_16">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">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 class="e" id="q_111d7522f5730943_1">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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