<font size=2 face="sans-serif">According to the default sip.conf file:</font>
<br>
<br><font size=2 face="sans-serif">sendrpid=yes ; If Remote-Party-ID should
be sent (defaults to no)</font>
<br>
<br><font size=2 face="sans-serif">sendrpid=rpid ; Use the "Remote-Party-ID"
header to send the identity of the remote party. This is identical to sendrpid=yes</font>
<br>
<br><font size=2 face="sans-serif">sendrpid=pai ; Use the "P-Asserted-Identity"
header to send the identity of the remote party.</font>
<br>
<br><font size=2 face="sans-serif">In my case, pai works. I could also
see yes or the equivalent rpid also working depending on what the phone
expects. I have to think that the reason the options are there is because
different endpoints behave differently.</font>
<br>
<br><font size=2 face="sans-serif">I believe the pai option was added in
1.8.</font>
<br><font size=2 face="sans-serif"><br>
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208</font>
<br>
<br>
<br>
<br><font size=1 color=#5f5f5f face="sans-serif">From:
</font><font size=1 face="sans-serif">Frank <frank@efirehouse.com></font>
<br><font size=1 color=#5f5f5f face="sans-serif">To:
</font><font size=1 face="sans-serif">Asterisk Users Mailing
List - Non-Commercial Discussion <asterisk-users@lists.digium.com>,
</font>
<br><font size=1 color=#5f5f5f face="sans-serif">Date:
</font><font size=1 face="sans-serif">02/04/2013 09:47 AM</font>
<br><font size=1 color=#5f5f5f face="sans-serif">Subject:
</font><font size=1 face="sans-serif">Re: [asterisk-users]
CallerID external call after Attended Transfer</font>
<br><font size=1 color=#5f5f5f face="sans-serif">Sent by:
</font><font size=1 face="sans-serif">asterisk-users-bounces@lists.digium.com</font>
<br>
<hr noshade>
<br>
<br>
<br><tt><font size=2>What is the PAI option below that you are talking
about, for sendrpid ?<br>
The manual only says that yes or no can be used..<br>
<br>
<br>
On 2/4/13 9:39 AM, Kevin Larsen wrote:<br>
> One thing you can try is to set the following in your sip.conf.<br>
><br>
> sendrpid=pai<br>
> trustrpid=yes<br>
><br>
> You can put that on individual phone configurations in sip.conf or,
as I<br>
> do, in a template that is applied to a set of phones.<br>
><br>
> I believe that was what I had to set so that the remote caller ID
would<br>
> show up properly on my Polycom phones. I made no changes to the Polycom<br>
> configuration to make it work. It might work with the Yealink T32G<br>
> phones as well.<br>
><br>
> In the case originally presented, I get the following:<br>
><br>
> Call comes into Operator showing cell phone caller id. Operator performs<br>
> an attended transfer. I get the Operator caller ID. Upon completion
of<br>
> the transfer, I get the cell phone caller ID. If a blind transfer
is<br>
> performed, I get the cell phone caller ID (there might be a flash
of the<br>
> operators caller ID for just the split second it takes her to hit
the<br>
> transfer button a second time to turn it from attended to blind transfer<br>
> on my phones).<br>
><br>
> Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208<br>
><br>
><br>
><br>
> From: Steven Howes <steve-lists@geekinter.net><br>
> To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
> <asterisk-users@lists.digium.com>,<br>
> Date: 02/04/2013 08:31 AM<br>
> Subject: Re: [asterisk-users] CallerID external call after Attended<br>
> Transfer<br>
> Sent by: asterisk-users-bounces@lists.digium.com<br>
> ------------------------------------------------------------------------<br>
><br>
><br>
><br>
> On 4 Feb 2013, at 13:45, Jonas Kellens wrote:<br>
> The IP-phones in this case are Yealink T32G.<br>
><br>
> What setting is needed in this IP-phone ?<br>
><br>
> Quick google doesn't turn up any results. Handsets probably dont support<br>
> it.<br>
><br>
> Steve--<br>
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