I found very useful info here: https://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE
In other words, on the asterisk1 box, you need to fetch from SIPPEER in extensions on asterisk1 box, and then populate connectedline. SIPPEER is the callee leg of the call, and CONNECTEDLINE is the caller. So if you set CONNECTEDLINE on caller (eg the asterisk2 side of the trunk between asterisk1 and asterisk2), You need to fetch this info in extensions for the SIPPEER on asterisk1 side of the trunk between asterisk1 and asterisk2, and copy this info into CONNECTEDLINE (the ISDN PRI leg of the call) on the asterisk1 box. I guess you have a extension on asterisk2, and then call "through" asterisk1 box. (Otherwise, if you are "behind" asterisk2 box and call the Conf line on asterisk1, you need to do the opposite of above, set the things on asterisk2 box.) -----Ursprungligt meddelande----- Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Dmitry Melekhov Skickat: den 15 maj 2017 12:47 Till: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Ämne: [asterisk-users] Callee id over chan_sip trunk Hello! I run two asterisks 13.13.1. Here is how they are connected: me---PBX--isdn pri--asterisk1--sip--asterisk2. If I call something from asterisk1 and have in dial plan: Let's say exten => 6000,n,Set(CONNECTEDLINE(name)=Conf. 6000) exten => 6000,n,Meetme(6000,TL(10800000:60000)) Then I see Conf. 6000 on my phone if I call 6000. If I have the same code for number on asterisk2, then there is no name on my phone, i.e. looks like asterisk doesn't send this info, at least I don't see it in sip debug. Could you tell me is it possible to pass this over sip? Thank you! -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
