[asterisk-users] Question about callerid?

[Martin Joseph]

On Nov 6, 2009, at 5:14 AM, John A. Sullivan III wrote:

> 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.
OK,  that makes sense.  So since Asterisk is a back to to back user  
agent (ie the call is always going through it) then the Caller ID data  
isn't magically moved along...

Still, the fact that it's showing up there in the console means there  
should be some way to grab it (the callerID data) and stuff into into  
the proper place for it to be passed along.

I see that the callerid valiable can be set as per:

http://www.voip-info.org/wiki/view/Setting+Callerid

So that's nice,  and the only question is how to I get the callerID  
info from where it show in the console as "failed to authenticate"?

Either that,  or I could reconfigure my audiocodes and my asterisk so  
that instead of incoming calls dialing my desired extension (ie 2020),  
asterisk could accept the calls from the domain of the audiocodes (ie  
it's IP address).  Maybe that's how get the CID data.

Don't really know, but suspect there are lots of people here who do?

Thanks for any help in advance,
Marty


>
> 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
>
> http://www.spiritualoutreach.com
> Making Christianity intelligible to secular society
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users