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

Steve Davies davies147 at gmail.com
Wed Apr 25 11:54:55 CDT 2012

On 25 April 2012 16:55, Richard Mudgett <rmudgett at digium.com> wrote:
>> - 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


More information about the asterisk-users mailing list