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<div><FONT face="Arial, Helvetica, sans-serif"></FONT>Hi,</div>
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<div>i have the same problem. Trying to use the dialplan function CONNECTEDLINE() this way</div>
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<div> Set(CONNECTEDLINE(name)=${SIPPEER(${EXTEN},callerid_name)})<br>
Set(CONNECTEDLINE(num)=${EXTEN})<br>
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<div>ends with</div>
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<div>[Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function CONNECTEDLINE not registered<br>
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<div>Same happens trying function CALLEDID.</div>
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<div>I am using Asterisk 1.6.1.20.</div>
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<div>What do i have to do to use this function or alternatively the function CALLEDID() described in bug 8824?</div>
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<div>Thanks in advance for help.</div>
<div><br>
<br>
</div>
<div style="CLEAR: both">Ondrej Valousek wrote:<br>
><I> Hello,<br>
</I>><I> <br>
</I>><I> Did anyone manage to force asterisk to put Remote-party-ID attribute on <br>
</I>><I> the SIP outgoing call? I.e. When A calls B, I want that A gets a name of </I>><I> B displayed on his phone.<br>
</I>><I> Note that name of A gets displayed on the B's phone fine, but this is <br>
</I>><I> not what I want.<br>
</I>><I> This works with Cisco Call manager fine - the RPID is sent as a part of <br>
</I>><I> the response to the SIP INVITE this way:<br>
</I>><I> <br>
</I>><I> <br>
</I>><I> SIP/2.0 180 Ringing<br>
</I>><I> Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport<br>
</I>><I> From: "Ondrej Valousek" <sip:<A href="http://lists.digium.com/mailman/listinfo/asterisk-users">7775 at 192.168.60.20</A>> <sip:<A href="http://lists.digium.com/mailman/listinfo/asterisk-users">7775 at 192.168.60.20</A>> ;tag=as4786d518<br>
</I>><I> To: <sip:<A href="http://lists.digium.com/mailman/listinfo/asterisk-users">1098 at 192.168.62.12</A>> <sip:<A href="http://lists.digium.com/mailman/listinfo/asterisk-users">1098 at 192.168.62.12</A>> ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104<br>
</I>><I> Date: Tue, 30 Mar 2010 13:53:15 GMT<br>
</I>><I> Call-ID: <A href="http://lists.digium.com/mailman/listinfo/asterisk-users">465a9c200587260d164f4514094896fb at 192.168.60.20</A><br>
</I>><I> CSeq: 102 INVITE<br>
</I>><I> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY<br>
</I>><I> Allow-Events: presence<br>
</I>><I> *Remote-Party-ID: "Paul Ryan" <sip:<A href="http://lists.digium.com/mailman/listinfo/asterisk-users">1098 at 192.168.62.12</A>> <sip:<A href="http://lists.digium.com/mailman/listinfo/asterisk-users">1098 at 192.168.62.12</A>> ;party=called;screen=yes;privacy=off*<br>
</I>><I> Contact: <sip:<A href="http://lists.digium.com/mailman/listinfo/asterisk-users">1098 at 192.168.62.12</A>:5060> <sip:<A href="http://lists.digium.com/mailman/listinfo/asterisk-users">1098 at 192.168.62.12</A>:5060> <br>
</I>><I> Content-Length: 0<br>
</I>><I> <br>
</I>><I> <br>
</I>><I> But I can not make it working with Asterisk. Does anyone have any glue <br>
</I>><I> how to achieve this WITHOUT patching asterisk? I am happy to upgrade to <br>
</I>><I> the latest/greatest version, I just do not want to patch.<br>
</I>><I> Many thanks,<br>
</I>><I> <br>
</I>><I> Ondrej<br>
</I>><I> <br>
</I><br>
This feature is in Asterisk trunk and will be present in the upcoming 1.8 <br>
release. By setting sendrpid=yes on A's phone, Asterisk will send a <br>
Remote-Party-ID header that corresponds to what Asterisk received from B. Also, <br>
there is a CONNECTEDLINE() dialplan function that can be used to send this <br>
information prior to a call. I actually gave a presentation on this topic at <br>
Astricon last year, but for some reason the Astricon '09 archive does not seem <br>
to have my presentation video available.<br>
<br>
Mark Michelson<br>
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