[asterisk-users] How to add Callee's name into Dial command ?

Olivier oza-4h07 at myamail.com
Sat Oct 4 05:21:55 CDT 2008


2008/10/3 satish patel <satish at linuxbug.org>

>
>
>
> 2008/10/3 Joe Pukepail <pukepail at gmail.com>
>
>> I think this is what you want: http://bugs.digium.com/view.php?id=8824
>>
>
> Thanks : this one very interesting.
>
> Bottom line is it doesn't work at the moment right ?
>
>>  <http://bugs.digium.com/view.php?id=8824>
>>
>>   On Fri, Oct 3, 2008 at 4:21 AM, Olivier <oza-4h07 at myamail.com> wrote:
>>
>>>   Hi,
>>>
>>> When dialing a number, I use :
>>> exten => _123X, 1, Dial (SIP/${EXTEN})
>>>
>>> Then, I get TRYING and RINGING SIP messages which both include this kind
>>> of line :
>>> To: <sip 1234 at 192.168.1.1;user=phone>
>>>
>>> Is it possible, configuring Asterisk 1.4, to get something like this
>>> instead ?
>>> To: "John Doe" <sip 1234 at 192.168.1.1;user=phone>
>>>
>>> This way, I'm hoping to display callee's name beside (or instead of)
>>> callee's number which would offer a double check for caller which might be
>>> confusing extensions, for instance.
>>>
>>>
>>> I tried this :
>>> exten => _123X, 1, SIPAddHeader(To: Doe \<sip 1234 at 192.168.1.1
>>> \;user=phone\>)
>>>
>>> but I still got :
>>> To: <sip 1234 at 192.168.1.1;user=phone>
>>>
>>> Regards
>>>
>>> _______________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>>> Register Now: http://www.astricon.net
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>  why you people need this thing in dial command which can possible
> with sip.conf callerid options
>

Unfortunately,  callerid option in sip.conf is not used to callee's name in
caller's phone screen :
if Alice calls Bob, Alice's phone will display Bob's number but not Bob (ie
callee's name)

If you SIP messages that comes back from Asterisk to Alice's phone, you
won't find the name Bob anywhere, so obviously, as Alice phone will use
those messages to update its own screen, you won't see any sign of callee's
name anywhere.

P-Asserted-Identity is a rather new field which is dedicated to such names
and is supported by several phones.

At the moment, Asterisk won't add this field in any reply to Alice's INVITE.

Cheers

>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081004/765f66c6/attachment.htm 


More information about the asterisk-users mailing list