Re: [Freeswitch-users] OpenZap and Sangoma A500 BRI card

2009-03-10 Thread Sergey Kirillov

 I'm no BRI expert but it looks to me like your wanpipe is set up for
 E1/EuroISDN. Where did you get this setup information?
 -MC

It is autoconfigured by wancfg

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Re: [Freeswitch-users] OpenZap and Sangoma A500 BRI card

2009-03-10 Thread Sergey Kirillov

Is there a development version that I can check now?

On Mon, Mar 9, 2009, Anthony MinessaleAnthony Minessale  wrote:
 it's not released yet,

 please wait for the announcement that it has been completed sometime in the
 next week or 2.


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Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread rod
Hi all,

it seems there is no way to do this :(

It could be great to be able to:
- decide if RPID should be present or not in the B leg for an 
outbound call
- make RPID header fully customizable with variables
- filter RPID for inbound call

I saw that kokoska rokoska created a jira bounty for 50$: Make RPID SIP 
header optional

I'll add 150$ for this if I could manage RPID as described above.
Sorry to use mailing list for this, I'm unable to add a note on jira for 
this bounty.

regards,
rod

rod wrote:
 Hi David,

 already tried this :p
 the pbm is that this doesn' modify the RPID header, but it adds a new 
 one so that I have 2 RPID header in the SIP INVITE :(

 rod

 David Knell wrote:
   
 Hi Rod,

 You can set it directly:
 action 
 application=set![CDATA[sip_h_Remote-Party-Identity=${caller_id_number} 
 sip:{$caller_id_numb...@1.2.3.4;screen=yes;privacy=off]]/action
 action application=bridge data=sofia/gateway/wherever/+$1/

 --Dave

 
 using these functions like this did nothing on the SIP INVITE packet :'(

 seven wrote:
   
   
 try  
 bridge 
 ({effective_caller_id_name 
 =your_name,effective_caller_id_number=}sofia/b-leg)

 On Mar 5, 2009, at 9:00 PM, rod wrote:

   
 
 
 the A leg invite looks like this:
 From: Anonymoussip:anonym...@anonymous.invalid;user=phone

 it has been rewritten like this:
 From: Anonymous sip:00anonym...@172.29.0.5

 rod

 rod wrote:
 
   
   
 Hi Brian,

 if I use the function effective_caller_id_number with my INVITE, I  
 get this:

 From: Anonymous sip:00anonym...@172.29.0.5;tag=17geyFjX5p0gS.

 this is not exactly what I'm looking for :p

 rod


 Brian West wrote:

   
 
 
 Well this depends on how you're placing the call.. if its a standard
 bridge you can on the A-Leg set
 effective_caller_id_number=000${caller_id_number} before you call
 bridge.

 Is the from already in the correct format?

 /b

 On Mar 5, 2009, at 6:12 AM, rod wrote:


 
   
   
 Dear list,

 I'd like to rewrite the number in the Remote Party ID header and  
 only in
 this header.

 ex: I'd like to prefix the caller  ID with a prefix code (000 in  
 this
 example) in the  RPID header :

   From: Anonymoussip:anonym...@anonymous.invalid;tag=1208367
   Remote-Party-ID:
 sip: 
 123456 
 @10.10.10.10:5062;user=phone;privacy=full;screen=yes;party=calling

 should become:
   From: Anonymoussip:anonym...@anonymous.invalid;tag=1208367
   Remote-Party-ID:
 sip: 
 000123456 
 @10.10.10.10:5062;user=phone;privacy=full;screen=yes;party=calling

 But the From field should remain unchanged.

 And how to strip this prefix:
   From: Anonymoussip:anonym...@anonymous.invalid;tag=1208367
   Remote-Party-ID:
 sip: 
 000123456 
 @10.10.10.10:5062;user=phone;privacy=full;screen=yes;party=calling

 should become:
   From: Anonymoussip:anonym...@anonymous.invalid;tag=1208367
   Remote-Party-ID:
 sip: 
 123456 
 @10.10.10.10:5062;user=phone;privacy=full;screen=yes;party=calling


 regards.

   
 
 
 

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Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread kokoska rokoska



rod napsal(a):
 Hi all,
 
 it seems there is no way to do this :(
 
 It could be great to be able to:
 - decide if RPID should be present or not in the B leg for an 
 outbound call
 - make RPID header fully customizable with variables
 - filter RPID for inbound call
 
 I saw that kokoska rokoska created a jira bounty for 50$: Make RPID SIP 
 header optional
 
 I'll add 150$ for this if I could manage RPID as described above.
 Sorry to use mailing list for this, I'm unable to add a note on jira for 
 this bounty.


I really don't need to customize RPID (cause it is depricated header),
but... Well let's add another 50 $ if it will work like you describe :-)

Best regards,

kokoska.rokoska


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Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread rod
Yes I know, it's deprecated but many peers still rely on this and 
P-Asserted-ID is not widely spread (my own experience).
moreover if we could strip the RPID, we could write a new one, but It 
could be very convenient to get access to the fields in this header for 
manipulation.

thanks for supporting this request :p

kokoska rokoska wrote:


 rod napsal(a):
   
 Hi all,

 it seems there is no way to do this :(

 It could be great to be able to:
 - decide if RPID should be present or not in the B leg for an 
 outbound call
 - make RPID header fully customizable with variables
 - filter RPID for inbound call

 I saw that kokoska rokoska created a jira bounty for 50$: Make RPID SIP 
 header optional

 I'll add 150$ for this if I could manage RPID as described above.
 Sorry to use mailing list for this, I'm unable to add a note on jira for 
 this bounty.

 

 I really don't need to customize RPID (cause it is depricated header),
 but... Well let's add another 50 $ if it will work like you describe :-)

 Best regards,

 kokoska.rokoska


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[Freeswitch-users] FS stopped working when NIC connection bounced.

2009-03-10 Thread Richard Lamkin
Hi All,

 

FS stopped working when NIC connection bounced.

 

Q1- Has anyone an explanation of what happened ? 

Q2 - Is there a way to configure FS not to flip to another IP on a lost
network connection. 

Q3 - Should  an IP changed event be logged at a higher level e.g.
[CRITICAL] ?

 

NIC2 [172.22.240.156] lost connection to LAN then Freeswitch flipped to
NIC1[192.168.1.1]. Then NIC2 restored connection and Freeswitch flipped
back to NIC2.  When it flipped back no SIP connections were restored.
All PBX phones lost registration and my gateways to remote switches did
not recover.   FS is running but not responding to CLI or SIP. There is
nothing logged in the Windows event viewer at the time of the incident.
Sorry but I was not running Wireshark when this happened.

 

== Log extract 

 

2009-03-09 21:00:21 [INFO] mod_sofia.c:2785 general_event_handler() IP
change detected [172.22.240.156]-[192.168.1.1] []-[]

2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write
lock external

2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run()
Waiting for worker thread

2009-03-09 21:00:21 [NOTICE] sofia_glue.c:2923
sofia_glue_restart_all_profiles() Reload XML [Success]

2009-03-09 21:00:21 [INFO] switch_time.c:656 switch_load_timezones()
Timezone reloaded 530 definitions

2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write
lock internal

2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run()
Waiting for worker thread

2009-03-09 21:11:21 [INFO] mod_sofia.c:2785 general_event_handler() IP
change detected [192.168.1.1]-[172.22.240.156] []-[]

2009-03-09 21:11:21 [NOTICE] sofia_glue.c:2923
sofia_glue_restart_all_profiles() Reload XML [Success]

2009-03-09 21:11:21 [INFO] switch_time.c:656 switch_load_timezones()
Timezone reloaded 530 definitions

 

There were no more logged events after 2009-03-09 21:11:21. The log file
was checked at 2009-03-10 08:50 . All phones had lost registration
permanently 

I tried the CLI with sofia status this failed to return anything and
the CLI was no longer responsive. 

 

I did a CTR-C just to see what was alive [ VS2008 trapped the break and
I selected continue] 

 

2009-03-10 08:58:25 [WARNING] switch_scheduler.c:114 task_thread_loop()
Task was executed late by 4 seconds 1 heartbeat (core)

 

Still no CLI.

 

I the killed the FS, restarted and it worked as normal

 

 

Background; I'm testing Freeswitch on Windows XP with Release 1.03 from
the source release tar ball, running in VS2008 debug mode.  

 

The PC has two NIC's 

1 - 192.168.1.1 static 

2 - 172.22.240.156 DHCP

 

I'm using Nortel switches CS1K and CS2K for my upstream gateways and
Linksys SPA942 for PBX clients. 

 

I'm only using SIP - SIP and all sip connections are via
172.22.240.156, There are no SIP devices on 192.168.1.1.   

 

It is unlikely the event was caused by a DHCP lease renew.  

 

DHCP lease details for 172.22.240.156.

Lease Obtained. . . . . . . . . . : 09 March 2009 15:27:28

Lease Expires . . . . . . . . . . : 17 March 2009 15:27:28

 

===

 

Regards 

 

Richard Lamkin

 

PS: I am a new to FS and I am very enthusiastic about it potential. 

 

 


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Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread kokoska rokoska

rod napsal(a):
 ... if we could strip the RPID, we could write a new one, but It
 could be very convenient to get access to the fields in this header for 
 manipulation.
 

Yes, rod, this is exactly why I update bounty to $100 :-)

Thank you very much, rod, for support!

Best regards,

kokoska.rokoska


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Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread Anthony Minessale
Latest SVN:

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

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

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

Send RPID with chosen content
{sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/
u...@example.com
  [ Show » http://jira.freeswitch.org/browse/BOUNTY-2 ]
 Anthony Minessale
IIhttp://jira.freeswitch.org/secure/ViewProfile.jspa?name=anthmadded
a comment - 10/Mar/09
07:59 AM Send no extra caller id info: {sip_cid_type=none}sofia/default/
u...@example.com Send RPID (default) {sip_cid_type=rpid}sofia/default/
u...@example.com Send P-XXX-Identity {sip_cid_type=pid}sofia/default/
u...@example.com Send RPID with chosen content
{sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/
u...@example.com

Send RPID with chosen content and privacy flags (+ delimited, none to clear
all flags)
{sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/
u...@example.com


  [ Show » http://jira.freeswitch.org/browse/BOUNTY-2 ]
 Anthony Minessale
IIhttp://jira.freeswitch.org/secure/ViewProfile.jspa?name=anthmadded
a comment - 10/Mar/09
07:59 AM Send no extra caller id info: {sip_cid_type=none}sofia/default/
u...@example.com Send RPID (default) {sip_cid_type=rpid}sofia/default/
u...@example.com Send P-XXX-Identity {sip_cid_type=pid}sofia/default/
u...@example.com Send RPID with chosen content
{sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/
u...@example.com


Also the privacy app on the inbound leg controls the remaining contents of
the RPID and Privacy headers.


On Tue, Mar 10, 2009 at 7:32 AM, kokoska rokoska kokoska.roko...@post.czwrote:


 rod napsal(a):
  ... if we could strip the RPID, we could write a new one, but It
  could be very convenient to get access to the fields in this header for
  manipulation.
 

 Yes, rod, this is exactly why I update bounty to $100 :-)

 Thank you very much, rod, for support!

 Best regards,

 kokoska.rokoska


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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
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googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread Shelby Ramsey
Anthony,

That is awesome.  This is something that will be a BIG help.

SDR




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Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread kokoska rokoska



Anthony Minessale napsal(a):
 Latest SVN:
 
 Send no extra caller id info:
 {sip_cid_type=none}sofia/default/u...@example.com mailto:u...@example.com
 
 Send RPID (default)
 {sip_cid_type=rpid}sofia/default/u...@example.com mailto:u...@example.com
 
 Send P-XXX-Identity
 {sip_cid_type=pid}sofia/default/u...@example.com mailto:u...@example.com
 
 Send RPID with chosen content
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/u...@example.com
 mailto:u...@example.com
 [ Show » http://jira.freeswitch.org/browse/BOUNTY-2 ]
 Anthony Minessale II
 http://jira.freeswitch.org/secure/ViewProfile.jspa?name=anthm added a
 comment - 10/Mar/09 07:59 AM Send no extra caller id info:
 {sip_cid_type=none}sofia/default/u...@example.com
 mailto:u...@example.com Send RPID (default)
 {sip_cid_type=rpid}sofia/default/u...@example.com
 mailto:u...@example.com Send P-XXX-Identity
 {sip_cid_type=pid}sofia/default/u...@example.com
 mailto:u...@example.com Send RPID with chosen content
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/u...@example.com
 mailto:u...@example.com
 
 Send RPID with chosen content and privacy flags (+ delimited, none to
 clear all flags)
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/u...@example.com
 mailto:u...@example.com
 
 
 [ Show » http://jira.freeswitch.org/browse/BOUNTY-2 ]
 Anthony Minessale II
 http://jira.freeswitch.org/secure/ViewProfile.jspa?name=anthm added a
 comment - 10/Mar/09 07:59 AM Send no extra caller id info:
 {sip_cid_type=none}sofia/default/u...@example.com
 mailto:u...@example.com Send RPID (default)
 {sip_cid_type=rpid}sofia/default/u...@example.com
 mailto:u...@example.com Send P-XXX-Identity
 {sip_cid_type=pid}sofia/default/u...@example.com
 mailto:u...@example.com Send RPID with chosen content
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/u...@example.com
 mailto:u...@example.com
 
 
 Also the privacy app on the inbound leg controls the remaining contents
 of the RPID and Privacy headers.
 

Incredible speed - like usually :-)

Thank you very much, Anthony, for your work! Where should I send my bucks?

Best regards,

kokoska.rokoska


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Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread Anthony Minessale
The paypal button on the hompege will do ;)

On Tue, Mar 10, 2009 at 8:36 AM, kokoska rokoska kokoska.roko...@post.czwrote:




 Anthony Minessale napsal(a):
  Latest SVN:
 
  Send no extra caller id info:
  {sip_cid_type=none}sofia/default/u...@example.com mailto:
 u...@example.com
 
  Send RPID (default)
  {sip_cid_type=rpid}sofia/default/u...@example.com mailto:
 u...@example.com
 
  Send P-XXX-Identity
  {sip_cid_type=pid}sofia/default/u...@example.com mailto:
 u...@example.com
 
  Send RPID with chosen content
 
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/
 u...@example.com
  mailto:u...@example.com
  [ Show » http://jira.freeswitch.org/browse/BOUNTY-2 ]
  Anthony Minessale II
  http://jira.freeswitch.org/secure/ViewProfile.jspa?name=anthm added a
  comment - 10/Mar/09 07:59 AM Send no extra caller id info:
  {sip_cid_type=none}sofia/default/u...@example.com
  mailto:u...@example.com Send RPID (default)
  {sip_cid_type=rpid}sofia/default/u...@example.com
  mailto:u...@example.com Send P-XXX-Identity
  {sip_cid_type=pid}sofia/default/u...@example.com
  mailto:u...@example.com Send RPID with chosen content
 
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/
 u...@example.com
  mailto:u...@example.com
 
  Send RPID with chosen content and privacy flags (+ delimited, none to
  clear all flags)
 
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/
 u...@example.com
  mailto:u...@example.com
 
 
  [ Show » http://jira.freeswitch.org/browse/BOUNTY-2 ]
  Anthony Minessale II
  http://jira.freeswitch.org/secure/ViewProfile.jspa?name=anthm added a
  comment - 10/Mar/09 07:59 AM Send no extra caller id info:
  {sip_cid_type=none}sofia/default/u...@example.com
  mailto:u...@example.com Send RPID (default)
  {sip_cid_type=rpid}sofia/default/u...@example.com
  mailto:u...@example.com Send P-XXX-Identity
  {sip_cid_type=pid}sofia/default/u...@example.com
  mailto:u...@example.com Send RPID with chosen content
 
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/
 u...@example.com
  mailto:u...@example.com
 
 
  Also the privacy app on the inbound leg controls the remaining contents
  of the RPID and Privacy headers.
 

 Incredible speed - like usually :-)

 Thank you very much, Anthony, for your work! Where should I send my bucks?

 Best regards,

 kokoska.rokoska


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Re: [Freeswitch-users] FS stopped working when NIC connection bounced.

2009-03-10 Thread Anthony Minessale
conf/autoload_configs/sofia.conf.xml


 global_settings
param name=log-level value=0/
!-- param name=auto-restart value=false/ --
param name=debug-presence value=0/
  /global_settings



uncomment out the auto-restart.


On Tue, Mar 10, 2009 at 5:26 AM, Richard Lamkin
richard.lam...@mettoni.comwrote:

  Hi All,



 FS stopped working when NIC connection bounced.



 Q1- Has anyone an explanation of what happened ?

 Q2 - Is there a way to configure FS not to flip to another IP on a lost
 network connection.

 Q3 – Should  an IP changed event be logged at a higher level e.g.
 [CRITICAL] ?



 NIC2 [172.22.240.156] lost connection to LAN then Freeswitch flipped to
 NIC1[192.168.1.1]. Then NIC2 restored connection and Freeswitch flipped back
 to NIC2.  When it flipped back no SIP connections were restored.  All PBX
 phones lost registration and my gateways to remote switches did not
 recover.   FS is running but not responding to CLI or SIP. There is nothing
 logged in the Windows event viewer at the time of the incident. Sorry but I
 was not running Wireshark when this happened.



 == Log extract 



 2009-03-09 21:00:21 [INFO] mod_sofia.c:2785 general_event_handler() IP
 change detected [172.22.240.156]-[192.168.1.1] []-[]

 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write
 lock external

 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting
 for worker thread

 2009-03-09 21:00:21 [NOTICE] sofia_glue.c:2923
 sofia_glue_restart_all_profiles() Reload XML [Success]

 2009-03-09 21:00:21 [INFO] switch_time.c:656 switch_load_timezones()
 Timezone reloaded 530 definitions

 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write
 lock internal

 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting
 for worker thread

 2009-03-09 21:11:21 [INFO] mod_sofia.c:2785 general_event_handler() IP
 change detected [192.168.1.1]-[172.22.240.156] []-[]

 2009-03-09 21:11:21 [NOTICE] sofia_glue.c:2923
 sofia_glue_restart_all_profiles() Reload XML [Success]

 2009-03-09 21:11:21 [INFO] switch_time.c:656 switch_load_timezones()
 Timezone reloaded 530 definitions



 There were no more logged events after 2009-03-09 21:11:21. The log file
 was checked at 2009-03-10 08:50 . All phones had lost registration
 permanently

 I tried the CLI with “sofia status” this failed to return anything and the
 CLI was no longer responsive.



 I did a CTR-C just to see what was alive [ VS2008 trapped the break and I
 selected continue]



 2009-03-10 08:58:25 [WARNING] switch_scheduler.c:114 task_thread_loop()
 Task was executed late by 4 seconds 1 heartbeat (core)



 Still no CLI.



 I the killed the FS, restarted and it worked as normal



 

 Background; I’m testing Freeswitch on Windows XP with Release 1.03 from the
 source release tar ball, running in VS2008 debug mode.



 The PC has two NIC’s

 1 - 192.168.1.1 static

 2 - 172.22.240.156 DHCP



 I’m using Nortel switches CS1K and CS2K for my upstream gateways and
 Linksys SPA942 for PBX clients.



 I’m only using SIP - SIP and all sip connections are via 172.22.240.156,
 There are no SIP devices on 192.168.1.1.



 It is unlikely the event was caused by a DHCP lease renew.



 DHCP lease details for 172.22.240.156.

 Lease Obtained. . . . . . . . . . : 09 March 2009 15:27:28

 Lease Expires . . . . . . . . . . : 17 March 2009 15:27:28



 ===



 Regards



 Richard Lamkin



 PS: I am a new to FS and I am very enthusiastic about it potential.





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Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread rod
Hi Anthony,

thanks for this but I'd like to know if it's possible also to change 
only the caller_id_name and caller_id_number without modifying the from 
header.

ex: with the origination variables I get this
From: test sip:1...@172.29.0.5;tag=X4v4Kvt1B2DQF
Remote-Party-ID: test 
sip:1...@172.29.0.5;party=calling;screen=yes;privacy=off

and in the case of an anonymous call where from is like this on A-leg:
From: Anonymous 
sip:anonym...@anonymous.invalid;tag=z07m5db13cb-cbs450547977-o-5273-368294925

it is changed like this on B-leg:
From: test sip:1...@172.29.0.5;tag=X4v4Kvt1B2DQF

Moreover, in case of a bridged call, could you add the possibility to 
pass as is from header for an anonymous call.
Running current, I have this:
on A-leg:
From: 
Anonymoussip:anonym...@anonymous.invalid;user=phone;tag=c0a80101-fed0c
on B-leg:
From: Anonymous sip:anonym...@172.29.0.5;tag=tpjD57U2pyrFN

and I'd like B-leg to match A-leg for anonymous call as stated in RFC, 
anonym...@anonymous.invalid is the proposed way to handle anonymous 
call, I'll add 50$ more for support of this last request.

thanks for your reactivity.
regards,
rod

Anthony Minessale wrote:
 Latest SVN:

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

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

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

 Send RPID with chosen content
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/u...@example.com
  
 mailto:u...@example.com
 [ Show » http://jira.freeswitch.org/browse/BOUNTY-2 ]
 Anthony Minessale II 
 http://jira.freeswitch.org/secure/ViewProfile.jspa?name=anthm added 
 a comment - 10/Mar/09 07:59 AM Send no extra caller id info: 
 {sip_cid_type=none}sofia/default/u...@example.com 
 mailto:u...@example.com Send RPID (default) 
 {sip_cid_type=rpid}sofia/default/u...@example.com 
 mailto:u...@example.com Send P-XXX-Identity 
 {sip_cid_type=pid}sofia/default/u...@example.com 
 mailto:u...@example.com Send RPID with chosen content 
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/u...@example.com
  
 mailto:u...@example.com

 Send RPID with chosen content and privacy flags (+ delimited, none to 
 clear all flags)
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/u...@example.com
  
 mailto:u...@example.com


 [ Show » http://jira.freeswitch.org/browse/BOUNTY-2 ]
 Anthony Minessale II 
 http://jira.freeswitch.org/secure/ViewProfile.jspa?name=anthm added 
 a comment - 10/Mar/09 07:59 AM Send no extra caller id info: 
 {sip_cid_type=none}sofia/default/u...@example.com 
 mailto:u...@example.com Send RPID (default) 
 {sip_cid_type=rpid}sofia/default/u...@example.com 
 mailto:u...@example.com Send P-XXX-Identity 
 {sip_cid_type=pid}sofia/default/u...@example.com 
 mailto:u...@example.com Send RPID with chosen content 
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/u...@example.com
  
 mailto:u...@example.com


 Also the privacy app on the inbound leg controls the remaining 
 contents of the RPID and Privacy headers.


 On Tue, Mar 10, 2009 at 7:32 AM, kokoska rokoska 
 kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz wrote:


 rod napsal(a):
  ... if we could strip the RPID, we could write a new one, but It
  could be very convenient to get access to the fields in this
 header for
  manipulation.
 

 Yes, rod, this is exactly why I update bounty to $100 :-)

 Thank you very much, rod, for support!

 Best regards,

 kokoska.rokoska


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 ClueCon http://www.cluecon.com/

 AIM: anthm
 MSN:anthony_miness...@hotmail.com 
 mailto:msn%3aanthony_miness...@hotmail.com
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 mailto:sip%3a...@conference.freeswitch.org
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 http://iax:gu...@conference.freeswitch.org/888
 googletalk:conf+...@conference.freeswitch.org 
 mailto:googletalk%3aconf%2b...@conference.freeswitch.org
 pstn:213-799-1400
 

 

Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread Anthony Minessale
ok if you are up to date you should be able to add

{sip_from_uri=sip:anonym...@anonymous.invalid} to your dial string.


On Tue, Mar 10, 2009 at 9:30 AM, rod kawa...@laposte.net wrote:

 Hi Anthony,

 thanks for this but I'd like to know if it's possible also to change
 only the caller_id_name and caller_id_number without modifying the from
 header.

 ex: with the origination variables I get this
 From: test sip:1...@172.29.0.5 sip%3a1...@172.29.0.5
 ;tag=X4v4Kvt1B2DQF
 Remote-Party-ID: test
 sip:1...@172.29.0.5 sip%3a1...@172.29.0.5
 ;party=calling;screen=yes;privacy=off

 and in the case of an anonymous call where from is like this on A-leg:
 From: Anonymous
 sip:anonym...@anonymous.invalid
 ;tag=z07m5db13cb-cbs450547977-o-5273-368294925

 it is changed like this on B-leg:
 From: test sip:1...@172.29.0.5 sip%3a1...@172.29.0.5
 ;tag=X4v4Kvt1B2DQF

 Moreover, in case of a bridged call, could you add the possibility to
 pass as is from header for an anonymous call.
 Running current, I have this:
 on A-leg:
 From:
 Anonymoussip:anonym...@anonymous.invalid;user=phone;tag=c0a80101-fed0c
 on B-leg:
 From: Anonymous sip:anonym...@172.29.0.5 sip%3aanonym...@172.29.0.5
 ;tag=tpjD57U2pyrFN

 and I'd like B-leg to match A-leg for anonymous call as stated in RFC,
 anonym...@anonymous.invalid is the proposed way to handle anonymous
 call, I'll add 50$ more for support of this last request.

 thanks for your reactivity.
 regards,
 rod

 Anthony Minessale wrote:
  Latest SVN:
 
  Send no extra caller id info:
  {sip_cid_type=none}sofia/default/u...@example.com
  mailto:u...@example.com
 
  Send RPID (default)
  {sip_cid_type=rpid}sofia/default/u...@example.com
  mailto:u...@example.com
 
  Send P-XXX-Identity
  {sip_cid_type=pid}sofia/default/u...@example.com
  mailto:u...@example.com
 
  Send RPID with chosen content
 
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/
 u...@example.com
  mailto:u...@example.com
  [ Show » http://jira.freeswitch.org/browse/BOUNTY-2 ]
  Anthony Minessale II
  http://jira.freeswitch.org/secure/ViewProfile.jspa?name=anthm added
  a comment - 10/Mar/09 07:59 AM Send no extra caller id info:
  {sip_cid_type=none}sofia/default/u...@example.com
  mailto:u...@example.com Send RPID (default)
  {sip_cid_type=rpid}sofia/default/u...@example.com
  mailto:u...@example.com Send P-XXX-Identity
  {sip_cid_type=pid}sofia/default/u...@example.com
  mailto:u...@example.com Send RPID with chosen content
 
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/
 u...@example.com
  mailto:u...@example.com
 
  Send RPID with chosen content and privacy flags (+ delimited, none to
  clear all flags)
 
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/
 u...@example.com
  mailto:u...@example.com
 
 
  [ Show » http://jira.freeswitch.org/browse/BOUNTY-2 ]
  Anthony Minessale II
  http://jira.freeswitch.org/secure/ViewProfile.jspa?name=anthm added
  a comment - 10/Mar/09 07:59 AM Send no extra caller id info:
  {sip_cid_type=none}sofia/default/u...@example.com
  mailto:u...@example.com Send RPID (default)
  {sip_cid_type=rpid}sofia/default/u...@example.com
  mailto:u...@example.com Send P-XXX-Identity
  {sip_cid_type=pid}sofia/default/u...@example.com
  mailto:u...@example.com Send RPID with chosen content
 
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/
 u...@example.com
  mailto:u...@example.com
 
 
  Also the privacy app on the inbound leg controls the remaining
  contents of the RPID and Privacy headers.
 
 
  On Tue, Mar 10, 2009 at 7:32 AM, kokoska rokoska
  kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz wrote:
 
 
  rod napsal(a):
   ... if we could strip the RPID, we could write a new one, but It
   could be very convenient to get access to the fields in this
  header for
   manipulation.
  
 
  Yes, rod, this is exactly why I update bounty to $100 :-)
 
  Thank you very much, rod, for support!
 
  Best regards,
 
  kokoska.rokoska
 
 
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  --
  Anthony Minessale II
 
  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
 
  AIM: anthm
  MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
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  GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
  

Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread kokoska rokoska



Anthony Minessale napsal(a):
 ok if you are up to date you should be able to add
 
 {sip_from_uri=sip:anonym...@anonymous.invalid} to your dial string.
 
 

Many thanks, Anthony, for that feature!

It makes my life a lot easier :-)

Best regards,

kokoska.rokoska


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Re: [Freeswitch-users] Missing Diversion header in INVITE after 302 reply

2009-03-10 Thread Helmut Kuper
Hello,

has anybody an idea?

regards
Helmut

On 09.03.2009 16:19, Helmut Kuper wrote:
 Hello,

 following scenario:

 -Phone A is redirected unconditionally to phone C
 -Phone B calls A
 -Phone A replys with 302 and Dieversion header
 -FS detects the 302 and sends out a new INVITE to C

 I found that FS doesn't include the received diversion sip header into
 the new INVITE sent to phone C.

 Is there a way to configure FS so that diversion header are included?

 Additionally: Is there a way to inform phone A about the diversion
 header, so that phone A get display a hint to user?

 regards
 Helmut



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[Freeswitch-users] IRC pre release party

2009-03-10 Thread Anthony Minessale
If you have nothing better to do drop by IRC
We are up to 193 users and about to cross 200 for the first time.


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
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Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread Peter P GMX
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:
 On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale
 anthony.miness...@gmail.com wrote:
   
 Latest SVN:

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

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

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

 Send RPID with chosen content
 {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul
  tu...@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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Re: [Freeswitch-users] Rewriting Remote Party ID

2009-03-10 Thread Ken Rice
These should be available any time you are going to process a call thru the
dialplan and call a bridge on the call


 From: Peter P GMX prometheus...@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:
 On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale
 anthony.miness...@gmail.com wrote:
   
 Latest SVN:
 
 Send no extra caller id info:
 {sip_cid_type=none}sofia/default/u...@example.com
 
 Send RPID (default)
 {sip_cid_type=rpid}sofia/default/u...@example.com
 
 Send P-XXX-Identity
 {sip_cid_type=pid}sofia/default/u...@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
 tu...@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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Re: [Freeswitch-users] RPC and web admin panel for conference?

2009-03-10 Thread Ken Rice
Hey, I just implemented something like this and commited it to my contrib
directory (scripts/contrib/swk ) its a mixture of amf-php, ESL, and Flex...
Its not complete by anymeans and you need Flex3 to compile the UI...

Anyone wanting to throw some patches at it for other functionality are
welcome to do so... One thing is severly lacks at this time is ANY sort of
authentication...so you wouldn¹t want it publically open to the world

K



From: Harry FSwitch switchser...@gmail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Mon, 9 Mar 2009 18:17:41 -0400
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] RPC and web admin panel for conference?

Hi all,

I'm looking to implement an admin panel much like the one used at
http://conference.freeswitch.org. Now I obviously cannot login and see the
admin part of the panel but I'm pretty sure whats in there.

I have xml_rpc running and can connect via http and issue commands. I've
searched the forum here and went through the wiki, found nothing that looked
like a panel. I was hoping to find a panel I can just configure and
implement. Does anyone have a php (I guess, seeing as I have a php server)
panel they can share with me? I'm sure I can get it working for my system.
The thought of attempting one on my own at THIS point seems daunting at
best.

Any help would be greatly appreciated!

Thanks


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