[asterisk-users] Connected Line presentation in 1.8.x upwards

[Steve Davies]

On 29 July 2013 16:55, Kevin Larsen <kevin.larsen at pioneerballoon.com> wrote:

>
>
> From:        Steve Davies <davies147 at gmail.com>
> To:        Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com>,
> Date:        07/29/2013 10:53 AM
> Subject:        [asterisk-users] Connected Line presentation in 1.8.x
> upwards
> Sent by:        asterisk-users-bounces at lists.digium.com
> ------------------------------
>
>
>
> Hi,
>
> I've searched the *asterisk.org* <http://asterisk.org/> and voip-info
> wiki sites, but not found an answer that seems to match.
>
> Hopefully this is a simple question. COLP is working very well on our
> system - Unfortunately it is working a bit TOO well in some circumstances.
> We have some "untrusted" trunks. On these trunks, an initial CallerID can
> be used, but any redirected caller numbers, COLP updates etc are not safe
> to accept. Sadly I cannot find how to cause COLP updates to be ignored for
> a trunk.
>
> I need solutions for SIP, IAX and DAHDI, what options do I have? This
> applies to both in- and out-bound calls.
>
> Are there some variables that I can set just before dialling an outbound
> call, and immediately on receiving an inbound call to determine what the
> callerID values will be for the entire duration of the call? (ie. old-style
> pre-COLP behaviour for specific trunks)
>
> Thanks for any pointers.
>
> Regards,
> Steve
>
>
>
> I believe what you are looking for in Dial is the 'I' option.
>
>
>
Ah. Many thanks.

It appears that the normally reliable voip-info wiki is out of date and
does not include that option. I should probably have just used Asterisk's
built-in documentation anyway :)

I guess on an inbound call I will have to conditionally set 'I' on the Dial
command based on the originating channel?

I will also have to go and check what affect this has when a call is SIP
REFER'ed as that might result in an asymmetric requirement. The internal
SIP handset will want updating, but the external SIP trunk call will not.

Regards,
Steve
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