Hello, Did anyone manage to force asterisk to put Remote-party-ID attribute on the SIP outgoing call? I.e. When A calls B, I want that A gets a name of B displayed on his phone. Note that name of A gets displayed on the B's phone fine, but this is not what I want. This works with Cisco Call manager fine - the RPID is sent as a part of the response to the SIP INVITE this way: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport From: "Ondrej Valousek" <sip:7775 at 192.168.60.20> <sip:7775 at 192.168.60.20> ;tag=as4786d518 To: <sip:1098 at 192.168.62.12> <sip:1098 at 192.168.62.12> ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104 Date: Tue, 30 Mar 2010 13:53:15 GMT Call-ID: 465a9c200587260d164f4514094896fb at 192.168.60.20 CSeq: 102 INVITE Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence *Remote-Party-ID: "Paul Ryan" <sip:1098 at 192.168.62.12> <sip:1098 at 192.168.62.12> ;party=called;screen=yes;privacy=off* Contact: <sip:1098 at 192.168.62.12:5060> <sip:1098 at 192.168.62.12:5060> Content-Length: 0 But I can not make it working with Asterisk. Does anyone have any glue how to achieve this WITHOUT patching asterisk? I am happy to upgrade to the latest/greatest version, I just do not want to patch. Many thanks, Ondrej
Ondrej Valousek wrote:> Hello, > > Did anyone manage to force asterisk to put Remote-party-ID attribute on > the SIP outgoing call? I.e. When A calls B, I want that A gets a name of > B displayed on his phone. > Note that name of A gets displayed on the B's phone fine, but this is > not what I want. > This works with Cisco Call manager fine - the RPID is sent as a part of > the response to the SIP INVITE this way: > > > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport > From: "Ondrej Valousek" <sip:7775 at 192.168.60.20> <sip:7775 at 192.168.60.20> ;tag=as4786d518 > To: <sip:1098 at 192.168.62.12> <sip:1098 at 192.168.62.12> ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104 > Date: Tue, 30 Mar 2010 13:53:15 GMT > Call-ID: 465a9c200587260d164f4514094896fb at 192.168.60.20 > CSeq: 102 INVITE > Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY > Allow-Events: presence > *Remote-Party-ID: "Paul Ryan" <sip:1098 at 192.168.62.12> <sip:1098 at 192.168.62.12> ;party=called;screen=yes;privacy=off* > Contact: <sip:1098 at 192.168.62.12:5060> <sip:1098 at 192.168.62.12:5060> > Content-Length: 0 > > > But I can not make it working with Asterisk. Does anyone have any glue > how to achieve this WITHOUT patching asterisk? I am happy to upgrade to > the latest/greatest version, I just do not want to patch. > Many thanks, > > Ondrej >This feature is in Asterisk trunk and will be present in the upcoming 1.8 release. By setting sendrpid=yes on A's phone, Asterisk will send a Remote-Party-ID header that corresponds to what Asterisk received from B. Also, there is a CONNECTEDLINE() dialplan function that can be used to send this information prior to a call. I actually gave a presentation on this topic at Astricon last year, but for some reason the Astricon '09 archive does not seem to have my presentation video available. Mark Michelson
Try using sendrpid=yes on sip.conf Regards, Juan Ondrej Valousek wrote:> Hello, > > Did anyone manage to force asterisk to put Remote-party-ID attribute on > the SIP outgoing call? I.e. When A calls B, I want that A gets a name of > B displayed on his phone. > Note that name of A gets displayed on the B's phone fine, but this is > not what I want. > This works with Cisco Call manager fine - the RPID is sent as a part of > the response to the SIP INVITE this way: > > > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport > From: "Ondrej Valousek" <sip:7775 at 192.168.60.20> <sip:7775 at 192.168.60.20> ;tag=as4786d518 > To: <sip:1098 at 192.168.62.12> <sip:1098 at 192.168.62.12> ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104 > Date: Tue, 30 Mar 2010 13:53:15 GMT > Call-ID: 465a9c200587260d164f4514094896fb at 192.168.60.20 > CSeq: 102 INVITE > Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY > Allow-Events: presence > *Remote-Party-ID: "Paul Ryan" <sip:1098 at 192.168.62.12> <sip:1098 at 192.168.62.12> ;party=called;screen=yes;privacy=off* > Contact: <sip:1098 at 192.168.62.12:5060> <sip:1098 at 192.168.62.12:5060> > Content-Length: 0 > > > But I can not make it working with Asterisk. Does anyone have any glue > how to achieve this WITHOUT patching asterisk? I am happy to upgrade to > the latest/greatest version, I just do not want to patch. > Many thanks, > > Ondrej > >
https://issues.asterisk.org/view.php?id=6643 CP On Thu, Apr 1, 2010 at 7:36 AM, Ondrej Valousek <webserv at s3group.cz> wrote:> Hello, > > Did anyone manage to force asterisk to put Remote-party-ID attribute on > the SIP outgoing call? I.e. When A calls B, I want that A gets a name of > B displayed on his phone. > Note that name of A gets displayed on the B's phone fine, but this is > not what I want. > This works with Cisco Call manager fine - the RPID is sent as a part of > the response to the SIP INVITE this way: > > > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport > From: "Ondrej Valousek" <sip:7775 at 192.168.60.20> <sip:7775 at 192.168.60.20> ;tag=as4786d518 > To: <sip:1098 at 192.168.62.12> <sip:1098 at 192.168.62.12> ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104 > Date: Tue, 30 Mar 2010 13:53:15 GMT > Call-ID: 465a9c200587260d164f4514094896fb at 192.168.60.20 > CSeq: 102 INVITE > Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY > Allow-Events: presence > *Remote-Party-ID: "Paul Ryan" <sip:1098 at 192.168.62.12> <sip:1098 at 192.168.62.12> ;party=called;screen=yes;privacy=off* > Contact: <sip:1098 at 192.168.62.12:5060> <sip:1098 at 192.168.62.12:5060> > Content-Length: 0 > > > But I can not make it working with Asterisk. Does anyone have any glue > how to achieve this WITHOUT patching asterisk? I am happy to upgrade to > the latest/greatest version, I just do not want to patch. > Many thanks, > > Ondrej > > -- > _____________________________________________________________________ > -- 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 >