Ticket #2563 (closed Feature Requests: wontfix)

Opened 4 years ago

Last modified 1 year ago

Updating caller ID informations during a call - Directed call pickup caller id presentation

Reported by: olivier1010 Assigned to: p_lindheimer
Priority: minor Milestone: Undetermined
Component: Core Version:
Keywords: Cc:
Confirmation: Unreviewed SVN Revision (if applicable):
Backend Engine: Asterisk 1.4.x Backend Engine Version:

Description

As it could be interesting for some scenario to update the caller ID during a call, specially for directed call pickup, i've finally found those interesting informations inside the aastra admin manual.

This could be used with **xx call pickup, or *8 group call pickup, to send caller id informations before that the user decide to intercept the call.

Update Caller ID During a Call

It is possible for a proxy or call server to update the Caller ID information that displays on the phone during a call, by modifying the SIP Contact header in the 200 OK message and/or in a re-INVITE message. The phone displays the updated name and number information contained within the Contact header. The following parameter allows the system administrator to enable or disable this feature: sip update callerid: This parameter is configurable via the configuration files only.

Change History

12/16/07 20:15:41 changed by p_lindheimer

  • version deleted.

04/22/08 06:04:27 changed by vgster

I think this is an issue that Digium need to sort. I spoke with Aastra about this, as I was supporting a number of their phones, and I think the answer was that Asterisk needed to send the info.

(follow-up: ↓ 5 ) 02/06/10 11:12:06 changed by jsmith

This feature has been added in the trunk version of Asterisk, and will be available in Asterisk 1.8.

08/22/10 09:29:28 changed by olivier1010

I hope this will be backported to 1.4 and 1.6 as 1.8 is certainly in the two years time range for a final version fully supported by FreePBX and mains IPBX distributions.

(in reply to: ↑ 3 ) 08/22/10 12:44:21 changed by mbrevda

  • status changed from new to closed.
  • resolution set to wontfix.

Replying to jsmith:

This feature has been added in the trunk version of Asterisk, and will be available in Asterisk 1.8.

Thanks, we will wait patiently.