[asterisk-users] CONNECTEDLINE() updated during SIP events?

[Steve Davies]

On 25 April 2012 18:05, Kevin P. Fleming <kpfleming at 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