[asterisk-users] CONNECTEDLINE() updated during SIP events?
Kevin P. Fleming
kpfleming at digium.com
Wed Apr 25 12:05:19 CDT 2012
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
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: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
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