Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?
On 25 April 2012 18:05, Kevin P. Fleming kpflem...@digium.com wrote: On 04/25/2012 11:54 AM, Steve Davies wrote: A further question... It appears that for SIP endpoints, this facility only updates RPID and PAI headers? I have found that there appear to be 4 different SIP CID-update mechanisms out there as follows: - Update RPID and PAI (ITSP and trunks often understand this) - Update Contact: header (Aastra handsets use this) - A SIP INFO packet if Supported: callerid is specified (Older snom firmware uses this) - Update From: header if Supported: from-change is specified (RFC 4916, snom, Yealink) Are there existing plans to support any of these other methods? If not, I will almost certainly add them for my own use, and submit the code. No, we have no plans at this time to go beyond RPID and PAI support. Those two appear to cover all the current endpoints that we have been able to test with, and many community members have also used with other endpoints and had success. Thanks for that, I'll have to test further and see whether all the devices we use support RPID/PAI. It would certainly be easier than messing about with headers that should not really be changed! Changing the Contact header seems quite wrong; the display-name in a URI in the Contact header is pretty much irrelevant. Changing the From header also seems wrong; that should indicate who sent the initial INVITE, not who redirected it. I don't think we'd want to merge patches that added support for either of those mechanisms. The From: header change is a relatively recent RFC, but I've seen several handsets supporting it, and several non-Asterisk SIP stacks using this to achieve COLP updates. I completely agree that changing the Contact: header is daft, and I have no idea why Aastra use this method. Regards, Steve -- _ -- 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
Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?
I have read the excellent information here: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information and believe I have an understanding of what is offered. I have a couple of questions: - Is it possible to update COLP/COLR when a SIP redirect occurs, or when a SIP divert is in place? If so, how? All redirecting activity is valid only before the associated calls are answered. After the calls are answered, it is connected-line updates. The redirecting interception macros are invoked before the outgoing call is answered when the outgoing call is redirected by an entity further down the line. If your Asterisk server is redirecting the call, the REDIRECTING information is updated by normal dialplan activity before placing the next outgoing call to the redirected to party. - Is it possible to have the COLP/COLR information updated when a SIP attended transfer is completed? If so how? Transfers generate connected line update events automatically. The connected line interception macros give you a chance to alter the connected line information as it is passed between the connected endpoints of the bridge. In both of the above cases, there is no obvious dialplan execution when the calls are redirected, diverted or masqueraded, so we cannot update the CONNECTEDLINE() information trivially. Or am I missing an obvious trick? This is the purpose of the interception macros. Richard -- _ -- 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
Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?
On 25 April 2012 16:55, Richard Mudgett rmudg...@digium.com wrote: [snip] - Is it possible to have the COLP/COLR information updated when a SIP attended transfer is completed? If so how? Transfers generate connected line update events automatically. The connected line interception macros give you a chance to alter the connected line information as it is passed between the connected endpoints of the bridge. In both of the above cases, there is no obvious dialplan execution when the calls are redirected, diverted or masqueraded, so we cannot update the CONNECTEDLINE() information trivially. Or am I missing an obvious trick? This is the purpose of the interception macros. Ah, thank you. I was looking at it back-to-front. The key bit is Transfers generate connected line update events automatically. - I can now see this in the source code in ast_do_masquerade() and elsewhere. This then lets you use the macros as you describe. A further question... It appears that for SIP endpoints, this facility only updates RPID and PAI headers? I have found that there appear to be 4 different SIP CID-update mechanisms out there as follows: - Update RPID and PAI (ITSP and trunks often understand this) - Update Contact: header (Aastra handsets use this) - A SIP INFO packet if Supported: callerid is specified (Older snom firmware uses this) - Update From: header if Supported: from-change is specified (RFC 4916, snom, Yealink) Are there existing plans to support any of these other methods? If not, I will almost certainly add them for my own use, and submit the code. Regards, Steve -- _ -- 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
Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?
On 04/25/2012 11:54 AM, Steve Davies wrote: A further question... It appears that for SIP endpoints, this facility only updates RPID and PAI headers? I have found that there appear to be 4 different SIP CID-update mechanisms out there as follows: - Update RPID and PAI (ITSP and trunks often understand this) - Update Contact: header (Aastra handsets use this) - A SIP INFO packet if Supported: callerid is specified (Older snom firmware uses this) - Update From: header if Supported: from-change is specified (RFC 4916, snom, Yealink) Are there existing plans to support any of these other methods? If not, I will almost certainly add them for my own use, and submit the code. No, we have no plans at this time to go beyond RPID and PAI support. Those two appear to cover all the current endpoints that we have been able to test with, and many community members have also used with other endpoints and had success. Changing the Contact header seems quite wrong; the display-name in a URI in the Contact header is pretty much irrelevant. Changing the From header also seems wrong; that should indicate who sent the initial INVITE, not who redirected it. I don't think we'd want to merge patches that added support for either of those mechanisms. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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
