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[Freeswitch-users] Rewriting Remote Party ID

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intralanman at freeswi...
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PostPosted: Wed Mar 11, 2009 5:30 pm    Post subject: [Freeswitch-users] Rewriting Remote Party ID Reply with quote

Anthony Minessale wrote:
Quote:
try now using sip_from_display variable to control the display field
in the from url
It would be awesome if someone from the community would document all of
these nice new variables on the wiki.

-Ray

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kawarod at laposte.net
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PostPosted: Thu Mar 12, 2009 5:37 am    Post subject: [Freeswitch-users] Rewriting Remote Party ID Reply with quote

Thanks a lot Anthony,

it's working great.
I'm just checking the origination_privacy parameter, cause it seems to
do nothing in my setup.

Anthony Minessale wrote:
Quote:
oops hit send too soon,

try now using sip_from_display variable to control the display field
in the from url


On Wed, Mar 11, 2009 at 10:14 AM, Anthony Minessale
<anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote:

try now


On Wed, Mar 11, 2009 at 9:09 AM, rod <kawarod@laposte.net
<mailto:kawarod@laposte.net>> wrote:

ok,

almost perfect :p

using this in dialplan:
{sip_cid_type=rpid,origination_caller_id_number=000000${CALLER_NUMBER},origination_caller_id_name=test,originatio\
n_privacy=screen+hide_name+hide_number,sip_from_uri=sip:anonymous@anonymous.invalid}sofia/ss7/000000${destination_number}@${GW_IP}

I have this on B leg
From: "test" <sip:anonymous@anonymous.invalid>;tag=gQ53Hjp617BSH.
Remote-Party-ID: "test"
<sip:000000262262880001@172.29.0.5
<mailto:sip%3A000000262262880001@172.29.0.5>>;party=calling;screen=yes;privacy=off.

As you can see the caller id name is still modified, do you
think you
could add something like sip_from_uri_name to circumvent this.

Anthony, I tried many settings for origination_privacy and it
seems to
do nothing on the RPID header. Any clue?

regards.

Anthony Minessale wrote:
Quote:


On Wed, Mar 11, 2009 at 3:33 AM, rod <kawarod@laposte.net
<mailto:kawarod@laposte.net>
Quote:
<mailto:kawarod@laposte.net <mailto:kawarod@laposte.net>>>
wrote:
Quote:

thanks a lot Anthony,

I consider the work is done and I did the paypal
transfer of 200$.
Quote:

But I'll be happy if you could do some minor changes:


Done....r12563

--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com
<mailto:MSN%3Aanthony_minessale@hotmail.com>
Quote:
<mailto:MSN%3Aanthony_minessale@hotmail.com
<mailto:MSN%253Aanthony_minessale@hotmail.com>>
Quote:
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
<mailto:PAYPAL%3Aanthony.minessale@gmail.com>
Quote:
<mailto:PAYPAL%3Aanthony.minessale@gmail.com
<mailto:PAYPAL%253Aanthony.minessale@gmail.com>>
Quote:
IRC: irc.freenode.net <http://irc.freenode.net>
<http://irc.freenode.net> #freeswitch
Quote:

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
<mailto:sip%3A888@conference.freeswitch.org>
Quote:
<mailto:sip%3A888@conference.freeswitch.org
<mailto:sip%253A888@conference.freeswitch.org>>
Quote:
iax:guest@conference.freeswitch.org/888
<http://iax:guest@conference.freeswitch.org/888>
Quote:
<http://iax:guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org>
Quote:
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org
<mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>>
Quote:
pstn:213-799-1400

------------------------------------------------------------------------
Quote:

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Quote:
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Quote:
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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com
<mailto:MSN%3Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
<mailto:PAYPAL%3Aanthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch


FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
<mailto:sip%3A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888
<http://iax:guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org>
pstn:213-799-1400




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com
<mailto:MSN%3Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
<mailto:PAYPAL%3Aanthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
<mailto:sip%3A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888
<http://iax:guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org>
pstn:213-799-1400
------------------------------------------------------------------------

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Prometheus001 at gmx.net
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PostPosted: Fri Mar 13, 2009 2:20 pm    Post subject: [Freeswitch-users] Rewriting Remote Party ID Reply with quote

Ok, I tried it. It works fine on dialthru. SDP header is enhanced. Perfect!

However I have a problem which I already discussed before in this round,
and I am very close to solve it, I think.
Scenario is as follows

* A callback application which is done through event socket calls
party A with cid-number 'unknown'
* Party A enters Digits which represents the number of party B
* I set the caller-id-no and effective-caller-id-no to the value of
A and sip_cid_type=rpid
* Call is then transferred from A to Party B via the xml dialplan
* Now I have 2 different variables when I trace the event_socket
o caller-caller-id-number is 'anonymous' and this is submitted
in the INVITE message to B
o variable-caller-id-number is the right number of B, this is
not part of the Sip message
* B sees 'unknown' as the number of party A. This is also
represented in the SIP mssage sent to B
* I want B to see the right number of party A

My question: What can I do to set <effective-caller-id-no> to
<variable-caller-id-number> in the XML dialplan.

Here is the part of the dialplan where I want so set it:
<!-- Dial any 10 digit number -->
<extension name="10 digits">
<condition field="destination_number" expression="^mp_([1-9]\d{9})$">

<action application="bridge" data="sofia/external/1$1@domain.com"/>
</condition>
</extension>


Anybody has a solution?

Best regards
Peter




Ken Rice schrieb:
Quote:
These should be available any time you are going to process a call thru the
dialplan and call a bridge on the call



Quote:
From: Peter P GMX <Prometheus001@gmx.net>
Reply-To: <freeswitch-users@lists.freeswitch.org>
Date: Tue, 10 Mar 2009 23:20:39 +0100
To: <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Rewriting Remote Party ID

Hello,

are these variables only available at call setup time or can they be
changed during a call, e.g. before a call is being transferred to
another destination?

Best regards
Peter

Michael Collins schrieb:

Quote:
On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale
<anthony.minessale@gmail.com> wrote:


Quote:
Latest SVN:

Send no extra caller id info:
{sip_cid_type=none}sofia/default/user@example.com

Send RPID (default)
{sip_cid_type=rpid}sofia/default/user@example.com

Send P-XXX-Identity
{sip_cid_type=pid}sofia/default/user@example.com

Send RPID with chosen content
{sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_num
ber=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul
tuser@example.com


FYI, I added this info to the channel variables page:
http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type

-MC

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Prometheus001 at gmx.net
Guest





PostPosted: Sat Mar 14, 2009 8:40 am    Post subject: [Freeswitch-users] Rewriting Remote Party ID Reply with quote

I got it to work. It was just so easy:

<action application="set"
data="effective_caller_id_number=${caller_id_number}"/>

Best regards
Peter


Peter P GMX schrieb:
Quote:
Ok, I tried it. It works fine on dialthru. SDP header is enhanced. Perfect!

However I have a problem which I already discussed before in this round,
and I am very close to solve it, I think.
Scenario is as follows

* A callback application which is done through event socket calls
party A with cid-number 'unknown'
* Party A enters Digits which represents the number of party B
* I set the caller-id-no and effective-caller-id-no to the value of
A and sip_cid_type=rpid
* Call is then transferred from A to Party B via the xml dialplan
* Now I have 2 different variables when I trace the event_socket
o caller-caller-id-number is 'anonymous' and this is submitted
in the INVITE message to B
o variable-caller-id-number is the right number of B, this is
not part of the Sip message
* B sees 'unknown' as the number of party A. This is also
represented in the SIP mssage sent to B
* I want B to see the right number of party A

My question: What can I do to set <effective-caller-id-no> to
<variable-caller-id-number> in the XML dialplan.

Here is the part of the dialplan where I want so set it:
<!-- Dial any 10 digit number -->
<extension name="10 digits">
<condition field="destination_number" expression="^mp_([1-9]\d{9})$">

<action application="bridge" data="sofia/external/1$1@domain.com"/>
</condition>
</extension>


Anybody has a solution?

Best regards
Peter




Ken Rice schrieb:

Quote:
These should be available any time you are going to process a call thru the
dialplan and call a bridge on the call




Quote:
From: Peter P GMX <Prometheus001@gmx.net>
Reply-To: <freeswitch-users@lists.freeswitch.org>
Date: Tue, 10 Mar 2009 23:20:39 +0100
To: <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Rewriting Remote Party ID

Hello,

are these variables only available at call setup time or can they be
changed during a call, e.g. before a call is being transferred to
another destination?

Best regards
Peter

Michael Collins schrieb:


Quote:
On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale
<anthony.minessale@gmail.com> wrote:



Quote:
Latest SVN:

Send no extra caller id info:
{sip_cid_type=none}sofia/default/user@example.com

Send RPID (default)
{sip_cid_type=rpid}sofia/default/user@example.com

Send P-XXX-Identity
{sip_cid_type=pid}sofia/default/user@example.com

Send RPID with chosen content
{sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_num
ber=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul
tuser@example.com



FYI, I added this info to the channel variables page:
http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type

-MC

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anthony.minessale at g...
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PostPosted: Sat Mar 14, 2009 9:26 am    Post subject: [Freeswitch-users] Rewriting Remote Party ID Reply with quote

origination_privacy was indeed broken
fixed in r12603

On Thu, Mar 12, 2009 at 5:15 AM, rod <kawarod@laposte.net (kawarod@laposte.net)> wrote:
Quote:
Thanks a lot Anthony,

it's working great.
I'm just checking the origination_privacy parameter, cause it seems to
do nothing in my setup.






--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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