asterisk users - Oct 2008 - How to add Callee's name into Dial command ?

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
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2008/10/3 Olivier <oza-4h07 at myamail.com>
> 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>
>
I must add I also tried without success :
exten => _123X, 1, Set(SIALPEERNAME=Doe)


>
>
> Regards
>
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Hi,

[snip[
> 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.
you can set callerid per-peer in sip.conf
like:

callerid='Jhon Doe' <1234>

this should work autmagically ;)

cheers

-- 
------------------------------------------------
Daniele Santi       .o.
daniele at santi.vr.it ..o () ascii ribbon campaign
Linux User #415108  ooo /\  www.asciiribbon.org
------------------------------------------------

Mr Shunz wrote:
> [snip]
> 
>> To make myself clear, here is what I'm trying to do : when Alice is
calling
>> Bob (Alice -------> Asterisk------->Bob), I would like Bob's
phone to
>> display Alice's name  (no problem, for that) but I would also like
Alice's
>> phone screen to display Bob's name (instead of Bob's number)
> 
> mmm ... this wasn't clear on your OP ...
> so you need to show the CALLED name on the CALLER phone ...
> 
>> My SIP hardphone is capable of displaying P-Asserted-Identity in
outbound calls (not just inbound) but I
>> couldn't find any way to teach Asterisk to fill this
P-Asserted-Identity header :
> 
> you can try with: (*** untested ***)
> 
> exten => _123X, 1, SIPAddHeader(P-Asserted-Identity:
> '${CALLERID(name)' <${CALLERID(num)>)
Interesting idea - I can see that being very useful. I know on my old 
SPPA-841s they would do that - but it was based on looking up the dialed 
number in the internal directory (which I programmed using a perl script 
in the asterisk server.) So, when I dialed 213, the name appeared that I 
dialed, confirming I had the right person.

Unfortunately, I never have been able to get my Polycom SoundPoint IP 
500 phones to do that.

I just tested the SIPAddHeader command given above - doesn't work with 
the Polycom Soundpoint IP 500 that I tested with. (Even with the missing 
'}' at the end that I fixed - still doesn't work.)

Good idea thought - anybody have any magic that might make that work?

-josiah

-- 
Josiah Bryan
IT Manager
Productive Concepts, Inc.
jbryan at productiveconcepts.com
(765) 964-6009, ext. 224

I think this is what you want: 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
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Why you didn't read the whole thread before saying that is beyond me.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of satish patel
Sent: October 3, 2008 12:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to add Callee's name into Dial command ?

 

[Mark Hamilton] <snip>


 why you people need this thing in dial command which can possible with
sip.conf callerid options 

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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
>
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