[asterisk-users] Question about callerid?

[John A. Sullivan III]

On Thu, 2009-11-05 at 20:57 -0800, Martin Joseph wrote:
> Hello again Asterisk people.
> 
> I am running Asterisk 1.42 on an old PowerPC ibook.  I have had this  
> deployed for several years now, with pretty good results.
> 
> Recently I added a callerid service to my landline (qwest).
> 
> I am using the audiocodes MP114 2fxo/2fxs gateway, which is an  
> outstanding piece of hardware once it's configured (lol).
> 
> Anyhow,  I can see that the gateway is passing caller id info to  
> asterisk because the console will display something like:
> 
> [Nov  4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite:  
> Failed to authenticate user "SEATTLE SCHOOLS" <sip:2062524320 at 89.89.89.253 
>  >;tag=1c492497235
> 
> So the caller ID info is right there.
> 
> However on my extensions (or softphones) the id shows as the extension  
> # (ie 2003).
> 
> Is there something I need to do to set the callerid?  I can't seem to  
> find this in the examples?
> 
> Thanks in advance for helping with my (I am sure) stupid question...
<snip>
I'd like to understand this better myself as I know we don't have this
right in our environment.  I believe the reason you see that is because
Asterisk is providing a B2BUA (I think it's called), i.e., your caller
is not actually talking to your phone.  Instead, your caller is talking
to Asterisk on the inbound SIP ID (whatever that is) and then Asterisk
is calling your phone from the extension in the dial plan.  At least I
think that's why the extension shows up in the callerID.

The identity can be overridden in sip.conf with the fromdomain and
fromuser parameters.  However, we found this introduced its own
problems.  I suppose we just need to build more sophisticated logic into
our dialplan.  The problem is, if we set the fromdomain/user, we now
show correct sip sources when we make direct SIP calls and can return
those calls from the phone's call history.  However, it breaks all the
internal dialing which wants to dial to the extension. If we remove
fromdomain/user, the internal dialing works but public SIP calls now
show the extension as the user rather than the user's public SIP ID.

I'm sure as with most things in Asterisk, we can fix it if we just take
the time to think through the programming logic.  Hope this helps - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsullivan at opensourcedevel.com

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