[asterisk-users] Called party identification - where to take called name?
Hello, I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true. Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Called party identification - where to take calledname?
Yehavi wrote: I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? Short answer is that you cannot. Longer answer is that it is possible, but requires new functionality to be added to the core and a new API call be added that can check if the called party is a local endpoint and retrieve the caller-id values. At least that was what I found when working on the patch. If anyone knows a way to lookup a peer/friend from the dialplan and collect such details, it would be possible to use the existing patch without any more changes in the core. BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Called party identification - where to takecalledname?
From: Dan Austin [EMAIL PROTECTED] Date: Thu, 3 May 2007 10:01:25 -0700 Yehavi wrote: I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? Short answer is that you cannot. Longer answer is that it is possible, but requires new functionality to be added to the core and a new API call be added that can check if the called party is a local endpoint and retrieve the caller-id values. It will depend on actual application. For some small sites, manually setting up an AstDB family should suffice. This can even be semi automated. Yuan Liu At least that was what I found when working on the patch. If anyone knows a way to lookup a peer/friend from the dialplan and collect such details, it would be possible to use the existing patch without any more changes in the core. BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Called party identification - where to takecalledname?
Yehavi wrote: I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? Asnwering myself: I am using realtime extensions, so I've added call to MYSQL() application to get the called user callerid field. __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Called Party Identification on Polycom IP501
Greetings, The Polycom SIP 1.5 Admin Guide says this: 3.1.8 Connected Party Identification Where possible, the identity of the remote party to which the user has connected is displayed and logged. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion. Does anyone know if * can provide the network signaling required? If so, how? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Called Party Identification on Polycom IP501
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Tuesday, August 16, 2005 1:21 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Called Party Identification on Polycom IP501 Greetings, The Polycom SIP 1.5 Admin Guide says this: 3.1.8 Connected Party Identification Where possible, the identity of the remote party to which the user has connected is displayed and logged. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion. Does anyone know if * can provide the network signaling required? If so, how? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp That is very dependent on how the call egresses from *, ISDN, POTS, SIP, ??? Who are you calling? If I recall correctly it will work when you call another extension on the * box, but the signaling for that info does not exists in PRI/T1/POTS, so it is not an * issue if you area calling out, * cant get the info from the telco, so * cant send it to the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Called Party Identification on Polycom IP501
Anthony Rodgers wrote: Does anyone know if * can provide the network signaling required? If so, how? Not yet, no. I will be working on that after the 1.2 release of Asterisk is made, and we will be anxious for testers to try it out :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Called Party Identification
Steven Critchfield wrote: On Fri, 2004-01-09 at 22:40, Brent Franks wrote: Does * support Called Party Identification? Say for example, I dial extension 2000, SIP sends back John Doe from the sip.conf file where extension 2000 is defined? Would this violate the SIP RFC? Maybe you didn't think about the fact that extensions aren't defined in sip.conf. Also it is possible for many extensions to end up on any physical phone. So sending essentially caller ID back to the calling phone doesn't really make sense. Agreed, the SIP channel doesn't really now anything about extensions, until called. But when getting a call, we match with a user/peer and could in theory send back a name. I don't know if I want this, though, of privacy reasons. Maybe when I accept a call. And I haven't checked the RFCs on where this should be placed in the SIP headers. Interesting question. Propably belongs in the 200 OK message. Anyone else? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Called Party Identification
Brent Franks wrote: No, but the Caller ID Information for a SIP extension is stored in sip.conf, so yes, I did think about that. As far as making sense, many meridian systems do this, and it is quite helpful. This could help with the implementation of gastman, and also end user phones. On the Cisco's and Polycom's, when you place a call on hold, rather than seeing an extension, you would see the name and you could toggle between the calls and see the name, rather than number (O.K. that part is a convenience thing). I know on my Meridian system at work, if you accidentally dial the wrong extension, the name pops up after it starts ringing, and you know your calling the wrong person. You can hang up, or tell the person real quick, hey sorry, I meant to call someone else. It's one thing when you have an internal PBX, but when you open up for external SIP calls from the Internet - do you really want them to always get your full name? Maybe a filter would be good. Anyway, could you provide a SIP trace of a call setup with this feature? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Called Party Identification
Does * support Called Party Identification? Say for example, I dial extension 2000, SIP sends back John Doe from the sip.conf file where extension 2000 is defined? Would this violate the SIP RFC? - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Called Party Identification
No, but the Caller ID Information for a SIP extension is stored in sip.conf, so yes, I did think about that. As far as making sense, many meridian systems do this, and it is quite helpful. This could help with the implementation of gastman, and also end user phones. On the Cisco's and Polycom's, when you place a call on hold, rather than seeing an extension, you would see the name and you could toggle between the calls and see the name, rather than number (O.K. that part is a convenience thing). I know on my Meridian system at work, if you accidentally dial the wrong extension, the name pops up after it starts ringing, and you know your calling the wrong person. You can hang up, or tell the person real quick, hey sorry, I meant to call someone else. Not sure why other PBX's do Called Party Identification, but it would definitely be a nice feature to have, and they have found some use for putting it in their system. I'll begin work on it myself, I was just inquiring as to whether it was implemented or not. The Polycom phones state in the Admin Guide they support Called Party ID, but it is a network provided service. So they found a use for it too. Regards, Brent -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Saturday, January 10, 2004 1:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Called Party Identification On Fri, 2004-01-09 at 22:40, Brent Franks wrote: Does * support Called Party Identification? Say for example, I dial extension 2000, SIP sends back John Doe from the sip.conf file where extension 2000 is defined? Would this violate the SIP RFC? Maybe you didn't think about the fact that extensions aren't defined in sip.conf. Also it is possible for many extensions to end up on any physical phone. So sending essentially caller ID back to the calling phone doesn't really make sense. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
