asterisk users - Aug 2005 - Called Party Identification on Polycom IP501

Greetings,

The Polycom SIP 1.5 Admin Guide says this:

"3.1.8 Connected Party Identification

Where possible, the identity of the remote party to which the user has 
connected is displayed and logged.  The connected party identity is 
derived from the network signaling.  In some cases the remote party 
will be different from the called party identity due  to network call 
diversion."

Does anyone know if * can provide the "network signaling" required? If
so, how?

Regards,
-- 
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp

> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Anthony Rodgers
> Sent: Tuesday, August 16, 2005 1:21 PM
> To: Asterisk-Users@lists.digium.com
> Subject: [Asterisk-Users] Called Party Identification on Polycom IP501
> 
> Greetings,
> 
> The Polycom SIP 1.5 Admin Guide says this:
> 
> "3.1.8 Connected Party Identification
> 
> Where possible, the identity of the remote party to which the user has
> connected is displayed and logged.  The connected party identity is
> derived from the network signaling.  In some cases the remote party
> will be different from the called party identity due  to network call
> diversion."
> 
> Does anyone know if * can provide the "network signaling"
required? If
> so, how?
> 
> Regards,
> --
> Anthony Rodgers
> Business Systems Analyst
> District of North Vancouver
> Web: http://www.dnv.org
> RSS Feed: http://www.dnv.org/rss.asp
> 
That is very dependent on how the call egresses from *, ISDN, POTS, SIP,
???
Who are you calling?

If I recall correctly it will work when you call another extension on
the * box, but the signaling for that info does not exists in
PRI/T1/POTS, so it is not an * issue if you area calling out, * cant get
the info from the telco, so * cant send it to the phone.

Anthony Rodgers wrote:
> Does anyone know if * can provide the "network signaling"
required? If
> so, how?
Not yet, no. I will be working on that after the 1.2 release of Asterisk 
is made, and we will be anxious for testers to try it out :-)

Hi Damon,

It's not working SIP to SIP - I am wondering if there is something I am 
missing in my * config.

What I see on the Polycom display is:

To:2471
2471

Called party entry in sip.conf (calling party entry is identical):

[2471]
type=friend
context=internal
callerid=C***** M**** <2471>
secret=********
host=dynamic
nat=no
canreinvite=no
dtmfmode=rfc2833
mailbox=2471@default

The called party entry in phone2471.cfg (calling party entry is 
identical):

<?xml version="1.0" encoding="UTF-8"
standalone="yes"?>
<!-- Example Per-phone Configuration File -->
<!-- $Revision: 1.59 $  $Date: 2004/05/22 00:50:41 $ -->
<phone2471>
   <reg reg.1.displayName="C***** M****"
reg.1.address="2471"
reg.1.label="2471" reg.1.type="private"
reg.1.auth.userId="2471"
reg.1.auth.password="********"/>
   <msg msg.bypassInstantMessage="1">
   <mwi msg.mwi.1.subscribe=""
msg.mwi.1.callBackMode="contact"
msg.mwi.1.callBack="*98"/>
   </msg>
</phone2471>

Am I missing anything?

Regards,

Anthony
> That is very dependent on how the call egresses from *, ISDN, POTS, 
> SIP,
> ???
> Who are you calling?
>
>
> If I recall correctly it will work when you call another extension on
> the * box, but the signaling for that info does not exists in
> PRI/T1/POTS, so it is not an * issue if you area calling out, * cant 
> get
> the info from the telco, so * cant send it to the phone.

Try quotes and no spaces between name and number.

Callerid="first last"<2471>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Anthony Rodgers
> Sent: Tuesday, August 16, 2005 5:31 PM
> To: Asterisk-Users@lists.digium.com
> Subject: [Asterisk-Users] Re: Called Party Identification on Polycom
IP501
> 
> Hi Damon,
> 
> It's not working SIP to SIP - I am wondering if there is something I
am
> missing in my * config.
> 
> What I see on the Polycom display is:
> 
> To:2471
> 2471
> 
> Called party entry in sip.conf (calling party entry is identical):
> 
> [2471]
> type=friend
> context=internal
> callerid=C***** M**** <2471>
> secret=********
> host=dynamic
> nat=no
> canreinvite=no
> dtmfmode=rfc2833
> mailbox=2471@default
> 
> The called party entry in phone2471.cfg (calling party entry is
> identical):
> 
> <?xml version="1.0" encoding="UTF-8"
standalone="yes"?>
> <!-- Example Per-phone Configuration File -->
> <!-- $Revision: 1.59 $  $Date: 2004/05/22 00:50:41 $ -->
> <phone2471>
>    <reg reg.1.displayName="C***** M****"
reg.1.address="2471"
> reg.1.label="2471" reg.1.type="private"
reg.1.auth.userId="2471"
> reg.1.auth.password="********"/>
>    <msg msg.bypassInstantMessage="1">
>    <mwi msg.mwi.1.subscribe=""
msg.mwi.1.callBackMode="contact"
> msg.mwi.1.callBack="*98"/>
>    </msg>
> </phone2471>
> 
> Am I missing anything?
> 
> Regards,
> 
> Anthony
> 
> > That is very dependent on how the call egresses from *, ISDN, POTS,
> > SIP,
> > ???
> > Who are you calling?
> >
> >
> > If I recall correctly it will work when you call another extension
on
> > the * box, but the signaling for that info does not exists in
> > PRI/T1/POTS, so it is not an * issue if you area calling out, * cant
> > get
> > the info from the telco, so * cant send it to the phone.
> 
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