According to the default sip.conf file: sendrpid=yes ; If Remote-Party-ID should be sent (defaults to no)
sendrpid=rpid ; Use the "Remote-Party-ID" header to send the identity of the remote party. This is identical to sendrpid=yes sendrpid=pai ; Use the "P-Asserted-Identity" header to send the identity of the remote party. 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. I believe the pai option was added in 1.8. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: Frank <fr...@efirehouse.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>, Date: 02/04/2013 09:47 AM Subject: Re: [asterisk-users] CallerID external call after Attended Transfer Sent by: asterisk-users-boun...@lists.digium.com What is the PAI option below that you are talking about, for sendrpid ? The manual only says that yes or no can be used.. On 2/4/13 9:39 AM, Kevin Larsen wrote: > One thing you can try is to set the following in your sip.conf. > > sendrpid=pai > trustrpid=yes > > You can put that on individual phone configurations in sip.conf or, as I > do, in a template that is applied to a set of phones. > > I believe that was what I had to set so that the remote caller ID would > show up properly on my Polycom phones. I made no changes to the Polycom > configuration to make it work. It might work with the Yealink T32G > phones as well. > > In the case originally presented, I get the following: > > Call comes into Operator showing cell phone caller id. Operator performs > an attended transfer. I get the Operator caller ID. Upon completion of > the transfer, I get the cell phone caller ID. If a blind transfer is > performed, I get the cell phone caller ID (there might be a flash of the > operators caller ID for just the split second it takes her to hit the > transfer button a second time to turn it from attended to blind transfer > on my phones). > > Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 > > > > From: Steven Howes <steve-li...@geekinter.net> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com>, > Date: 02/04/2013 08:31 AM > Subject: Re: [asterisk-users] CallerID external call after Attended > Transfer > Sent by: asterisk-users-boun...@lists.digium.com > ------------------------------------------------------------------------ > > > > On 4 Feb 2013, at 13:45, Jonas Kellens wrote: > The IP-phones in this case are Yealink T32G. > > What setting is needed in this IP-phone ? > > Quick google doesn't turn up any results. Handsets probably dont support > it. > > Steve-- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com > <http://www.api-digital.com/>-- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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