[Asterisk-Users] CallerID retain on internal transfer
Hi,From http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/148565/focus=149455, you can read that :- SIP allows CallerID to be changed at the point when 2 separate calls are bridged to one ... - May 2006 trunk version of Asterisk did not support this behaviour at that time.Is it still true today ?Is this feature considered for inclusion in 1.4 or 1.6 development cycle ?Regards ___ --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] CallerID retain on internal transfer
Steve Davies wrote: > In the cases previously mentioned, the user is doing an attended > transfer using the handset features, and not Asterisk. I do not know > whether SIP even allows the Caller ID to be changed at the point when > two separate calls are bridged to one... It does, but Asterisk does not currently support that behavior (even in the development branch). I believe Olle's SIP transfer re-write may provide this functionality when Asterisk 1.4 is released, but I am not positive. ___ --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] CallerID retain on internal transfer
On 5/16/06, Avi Miller <[EMAIL PROTECTED]> wrote: Michael J. Tubby B.Sc (Hons) G8TIC wrote: > call then transfers it on to another extension transferee (recipeient) > sees the Caller*ID This behaviour changed in Asterisk 1.2 -- add "o" to your Dial options and Asterisk will retain the original Caller ID on transfer. This does not sound like quite the same thing "o" reverts to 1.0.x behaviour, which is still not presenting the original number after an attended transfer that is managed from the phone handset (perhaps it does work if the *2 feature is used?) In the cases previously mentioned, the user is doing an attended transfer using the handset features, and not Asterisk. I do not know whether SIP even allows the Caller ID to be changed at the point when two separate calls are bridged to one... i.e. The current behaviour (on our system) is: Caller -> Phone A (Caller's ID) Caller -> On Hold Phone A -> Phone B (A's ID) Phone A disconnected Caller -> Phone B (A's ID) But the desired behaviour is: Caller -> Phone A (Caller's ID) Caller -> On Hold Phone A -> Phone B (A's ID) Phone A disconnected Caller -> Phone B (Caller's ID) The "o" option looks as if it changes the number that is initially presented on Phone B, and is only under Asterisk's control if you use Asterisk's built-in attended transfer facility, otherwise the 'Phone A' handset is responsible for the change. I suspect that the desired behaviour is also only possible if 'Phone A' does the "Right Thing (tm)" *THINKS* In fact, when the phone is doing the attended transfer, the caller-ID that should be presented AFTER a transfer will depend entirely upon the final destination of the call. It may not even be possible to change the CID upon transfer if the attended call went (for example) out of a ZAP channel. Looks like this is not an easy one to solve, but I am not 100% sure of which party is responsible for what during this type of transfer so I may be wrong... Cheers, Steve ___ --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] CallerID retain on internal transfer
Michael J. Tubby B.Sc (Hons) G8TIC wrote: call then transfers it on to another extension transferee (recipeient) sees the Caller*ID This behaviour changed in Asterisk 1.2 -- add "o" to your Dial options and Asterisk will retain the original Caller ID on transfer. -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ .>> Open Source - Own it - Squiz.net ./> ___ --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] CallerID retain on internal transfer
You sure you doing a blind xfer? On 5/15/06, Michael J. Tubby B.Sc (Hons) G8TIC <[EMAIL PROTECTED]> wrote: I'm seeing a similar thing... We have Asterisk 1.2.7.1, Chan CAPI and a home-brewed IVR/Auto Attendant that places in-bound calls to role-based riinging groups like "sales", "support", "admin" etc. which works well, but from a 7960G phone (SIP 7.5) if the person that answers a call then transfers it on to another extension transferee (recipeient) sees the Caller*ID of the transferor (internal role-based extension number) and not the Caller*ID of the original calling party. I took this to be a limitation of the way in which Cisco implement Attended/Blind transfers in the SIP firmware of the 7960G phone...? Mike - Original Message - From: "Steve Davies" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, May 10, 2006 10:52 AM Subject: Re: [Asterisk-Users] CallerID retain on internal transfer On 5/10/06, Joseph Rothstein <[EMAIL PROTECTED]> wrote: > From what I have tested, using cisco phones and 1.2.5, the original > callerID > is not kept when making a transfer. > > Any other ideas? > We use SPA, snom and aastra phones, and I had assumed that this was a limitation of the SIP protocol. I would be pleased to be told I am wrong on this one :) Regards, Steve ___ --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 ___ --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 ___ --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] CallerID retain on internal transfer
I'm seeing a similar thing... We have Asterisk 1.2.7.1, Chan CAPI and a home-brewed IVR/Auto Attendant that places in-bound calls to role-based riinging groups like "sales", "support", "admin" etc. which works well, but from a 7960G phone (SIP 7.5) if the person that answers a call then transfers it on to another extension transferee (recipeient) sees the Caller*ID of the transferor (internal role-based extension number) and not the Caller*ID of the original calling party. I took this to be a limitation of the way in which Cisco implement Attended/Blind transfers in the SIP firmware of the 7960G phone...? Mike - Original Message - From: "Steve Davies" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, May 10, 2006 10:52 AM Subject: Re: [Asterisk-Users] CallerID retain on internal transfer On 5/10/06, Joseph Rothstein <[EMAIL PROTECTED]> wrote: From what I have tested, using cisco phones and 1.2.5, the original callerID is not kept when making a transfer. Any other ideas? We use SPA, snom and aastra phones, and I had assumed that this was a limitation of the SIP protocol. I would be pleased to be told I am wrong on this one :) Regards, Steve ___ --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 ___ --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] CallerID retain on internal transfer
On 5/10/06, Joseph Rothstein <[EMAIL PROTECTED]> wrote: From what I have tested, using cisco phones and 1.2.5, the original callerID is not kept when making a transfer. Any other ideas? We use SPA, snom and aastra phones, and I had assumed that this was a limitation of the SIP protocol. I would be pleased to be told I am wrong on this one :) Regards, Steve ___ --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] CallerID retain on internal transfer
>From what I have tested, using cisco phones and 1.2.5, the original callerID is not kept when making a transfer. Any other ideas? Joe ___ --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] CallerID retain on internal transfer
Joe wrote: I was just looking through the Wiki for some info on how to retain the original caller's callerid when make transfers to internal extensions, and I came across the parameter below: useincomingcalleridonzaptransfer=yes There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering if this is still a valid parameter. If not, does anyone know how I can do this? 1.2 will keep the original Caller*ID when doing a transfer. 1.0.x did not do this. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] CallerID retain on internal transfer
I've been wondering about how to do what you're describing as well! Anyone know how it can be done? Regards, Bevan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Sent: Monday, 8 May 2006 4:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] CallerID retain on internal transfer I was just looking through the Wiki for some info on how to retain the original caller's callerid when make transfers to internal extensions, and I came across the parameter below: useincomingcalleridonzaptransfer=yes There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering if this is still a valid parameter. If not, does anyone know how I can do this? Thanks, Joe ___ --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 ___ --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] CallerID retain on internal transfer
I was just looking through the Wiki for some info on how to retain the original caller's callerid when make transfers to internal extensions, and I came across the parameter below: useincomingcalleridonzaptransfer=yes There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering if this is still a valid parameter. If not, does anyone know how I can do this? Thanks, Joe ___ --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
