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: [email protected] | SIP: [email protected] | 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