[Freeswitch-users] Performance Tuning
Looking at Performance Tune my Freeswitch http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations Is refers to the following: Turn off every module you don't need Turn presence off in the profiles libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles mod_cdr_csv is slower than mod_xml_cdr How do I change each one any references on Wiki? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch
Polycom Firmware matrix (Look at the polycom website) does not allow firmware higher than 2.3.2 (I think) to be loaded on the old 501 phones...So first confirm you are on a supported firmware release... From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Yehavi Bourvine Sent: Sunday, November 29, 2009 8:48 AM To: freeswitch-users Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch Hello, I am trying to set a Polycom 501 phone to do conferencing via the conference room on Freeswitch rather than on the phone (as on the phone it is limited to 3 participants only). Anyone had success with it? I have on the Freeswitch an extension named Conf.* which activates the conference application (it works with other brands). On the Polycom I tried to define voIpProt.SIP.conference.address=sip:conf0...@freeswitch-server. The phone continues to create the conference locally and add the above Conf to it, without REFERing the parties to it. The first phone which called is left on hold... Anyone managed to make this feature work? We use firmware 3.1.3. Thanks! __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Setting up Conference with Moderator
Cool, I will explore that option when I have some time.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Wednesday, November 18, 2009 11:02 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Hi again UK,
IVR is designed to naturally return to previous or top menus. I don't
think there's a way to change this default behavior. Maybe its time
to move to a script-based pin validation system so you have the full
control you need.
http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
Rob
On Nov 18, 2009, at 11:34 PM, Ujjval Karihaloo wrote:
I have used the following setting in ivr.conf.xml to setup
conferencing with moderator.
However, the issue I have is - the user enters 123456 and then say
if it's a moderator they enter wrong Moderator PIN 3 times then it
takes the user back to the main menu...conference_menu and asks
for main conf pin (123456) once again.
I would like the caller to be disconnected if they get into the
Moderator menu and enter wrong Moderator PIN 3 times.
menu name=conference_menu
greet-long=welcome_please_enter_conference_pin.wav
greet-short=check_and_try_again.wav
invalid-sound=passcode_invalid.wav
exit-sound=voicemail/vm-goodbye.wav
timeout=1
inter-digit-timeout=5000
max-failures=3
max-timeouts=3
digit-len=7
entry action=menu-sub digits=123456
param=conference_123456_moderator_menu / !-- conference
moderator menu --
/menu
menu name=conference_123456_moderator_menu
greet-
long
=
conference_confirmed_enter_moderator_pin_or_1_to_join_as_participant
.wav
greet-short=check_moderator_pin_or_1_to_join.wav
invalid-sound=invalid_moderator_pin.wav
exit-sound=voicemail/vm-goodbye.wav
timeout=1
inter-digit-timeout=5000
max-failures=3
max-timeouts=3
digit-len=5
entry action=menu-exec-app digits=1234 param=conference
123...@default+flags{moderator} /
entry action=menu-exec-app digits=1 param=conference
123...@default+flags{} /
/menu
/menus
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Thursday, November 05, 2009 7:52 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Hi UK,
From what I've done and read, the caller-controls (in
conference.conf.xml) can be modified to almost anything you can think
of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a
caller control just for those with the moderator flag. So unless you
want everyone able to mute/kick everyone then you can't do it.
The wiki seems to indicate this as well:
Be aware that the caller-controls are applied across the entire
conference. You cannot enter one member of the conference using
caller-
controls ABC and then enter a second member using caller-controls
XYZ.
http://wiki.freeswitch.org/wiki/Mod_conference
I think this might be a limitation of mod_conference. Perhaps one of
the pros can chime in if I'm off-base or there's some nifty way to
accomplish this.
Cheers,
Rob
On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote:
Any ideas on the below...has anyone implemented the below:
Once I have the Moderator and Participants logged on, how do I
invoke the moderator previlidges, LIk esay muting everyone/someone
or kicking someone out of the Conf and the like?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Ujjval Karihaloo
Sent: Monday, November 02, 2009 12:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Rob:
Once I have the Moderator and Participants logged on, how do I
invoke the moderator previlidges, LIk esay muting everyone/someone
or kicking someone out of the Conf and the like?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 9:34 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Hm, strange. I haven't seen that before. Can you pastebin your logs
at debug level?
On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
It's strange... a tcpdump tells me
Re: [Freeswitch-users] upgrading to latest SVN
I really didn't change anything. I was running 1.0.4 and now built from SVN...I see the oddly placed entry in ivr.conf.xml sscode_invalid.wav/configuration...removed it now its error on somewhere here: configuration name=xml_rpc.conf description=XML RPC settings !-- The port where you want to run the http service (default 8080) -- param name=http-port value=8080/ !-- if all 3 of the following params exist all http traffic will require auth -- param name=auth-realm value=freeswitch/ param name=auth-user value=freeswitch/ param name=auth-pass value=works/ /settings /configuration ERROR is: [r...@ss_freeswitch freeswitch]# freeswitch 2009-11-19 20:48:54.189120 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-19 20:48:54.190970 [DEBUG] switch_event.c:553 Create event dispatch thread 0 Cannot Initialize [[error near line 2840]: unexpected closing tag /section] -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, November 19, 2009 8:30 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN Ujjval Karihaloo ujj...@simplesignal.com wrote: Getting error below..not sure whats wrong..which line number in what file does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] upgrading to latest SVN
Does svn update try to merge the config files..Need some help, I think it has added some entries in my config files that is causing tag mismatches.. Please advise how to get back my orig config? -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval Karihaloo Sent: Thursday, November 19, 2009 8:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN I really didn't change anything. I was running 1.0.4 and now built from SVN...I see the oddly placed entry in ivr.conf.xml sscode_invalid.wav/configuration...removed it now its error on somewhere here: configuration name=xml_rpc.conf description=XML RPC settings !-- The port where you want to run the http service (default 8080) -- param name=http-port value=8080/ !-- if all 3 of the following params exist all http traffic will require auth -- param name=auth-realm value=freeswitch/ param name=auth-user value=freeswitch/ param name=auth-pass value=works/ /settings /configuration ERROR is: [r...@ss_freeswitch freeswitch]# freeswitch 2009-11-19 20:48:54.189120 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-19 20:48:54.190970 [DEBUG] switch_event.c:553 Create event dispatch thread 0 Cannot Initialize [[error near line 2840]: unexpected closing tag /section] -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, November 19, 2009 8:30 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN Ujjval Karihaloo ujj...@simplesignal.com wrote: Getting error below..not sure whats wrong..which line number in what file does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Changing User-Agent String
Not sure I am the only one changing User-Agentbut I just want a way for our Customers to know the purpose of the server when they talk to it. There is FreeSwitch written into the SDP o line as well...which I don't care about, I want to have something in there that identifies the purpose of the server. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Tuesday, November 17, 2009 10:52 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Changing User-Agent String It needs to go in the profile, not in sofia's global config. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.camailto:mr...@avgs.ca On 17-Nov-09, at 9:49 PM, Ujjval Karihaloo wrote: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#User_Agent_.5Buser-agent-string.5D As per the above link, we can change the User Agent String, but I added this param name but does not seem to work. [u...@freeswitch autoload_configs]$ vi sofia.conf.xml configuration name=sofia.conf description=sofia Endpoint global_settings param name=log-level value=0/ !-- param name=auto-restart value=false/ -- param name=debug-presence value=0/ param name=user-agent-string value=Test Server/ /global_settings !-- The rabbit hole goes deep. This includes all the profiles in the sip_profiles directory that is up one level from this directory. -- profiles X-PRE-PROCESS cmd=include data=../sip_profiles/*.xml/ /profiles /configuration ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto:FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Setting up Conference with Moderator
I have used the following setting in ivr.conf.xml to setup conferencing with
moderator.
However, the issue I have is - the user enters 123456 and then say if it's a
moderator they enter wrong Moderator PIN 3 times then it takes the user back to
the main menu...conference_menu and asks for main conf pin (123456) once
again.
I would like the caller to be disconnected if they get into the Moderator menu
and enter wrong Moderator PIN 3 times.
menu name=conference_menu
greet-long=welcome_please_enter_conference_pin.wav
greet-short=check_and_try_again.wav
invalid-sound=passcode_invalid.wav
exit-sound=voicemail/vm-goodbye.wav
timeout=1
inter-digit-timeout=5000
max-failures=3
max-timeouts=3
digit-len=7
entry action=menu-sub digits=123456
param=conference_123456_moderator_menu / !-- conference moderator menu --
/menu
menu name=conference_123456_moderator_menu
greet-long=conference_confirmed_enter_moderator_pin_or_1_to_join_as_participant.wav
greet-short=check_moderator_pin_or_1_to_join.wav
invalid-sound=invalid_moderator_pin.wav
exit-sound=voicemail/vm-goodbye.wav
timeout=1
inter-digit-timeout=5000
max-failures=3
max-timeouts=3
digit-len=5
entry action=menu-exec-app digits=1234 param=conference
123...@default+flags{moderator} /
entry action=menu-exec-app digits=1 param=conference
123...@default+flags{} /
/menu
/menus
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Thursday, November 05, 2009 7:52 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Hi UK,
From what I've done and read, the caller-controls (in
conference.conf.xml) can be modified to almost anything you can think
of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a
caller control just for those with the moderator flag. So unless you
want everyone able to mute/kick everyone then you can't do it.
The wiki seems to indicate this as well:
Be aware that the caller-controls are applied across the entire
conference. You cannot enter one member of the conference using caller-
controls ABC and then enter a second member using caller-controls XYZ.
http://wiki.freeswitch.org/wiki/Mod_conference
I think this might be a limitation of mod_conference. Perhaps one of
the pros can chime in if I'm off-base or there's some nifty way to
accomplish this.
Cheers,
Rob
On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote:
Any ideas on the below...has anyone implemented the below:
Once I have the Moderator and Participants logged on, how do I
invoke the moderator previlidges, LIk esay muting everyone/someone
or kicking someone out of the Conf and the like?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Ujjval Karihaloo
Sent: Monday, November 02, 2009 12:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Rob:
Once I have the Moderator and Participants logged on, how do I
invoke the moderator previlidges, LIk esay muting everyone/someone
or kicking someone out of the Conf and the like?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 9:34 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Hm, strange. I haven't seen that before. Can you pastebin your logs
at debug level?
On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
It's strange... a tcpdump tells me that there is no DTMF from my
provider when using IVR, but when I call into a TN that goes
directly into the Conference App, I see DTMF from the provider.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 7:23 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I've never had any problem with that. Is your logging at debug level
so you can see the RECV DTFM in the log/fs_cli? Are you calling from
a SIP phone on the pbx, or via a PSTN provider? Maybe your provider
isn't passing them through.
Make sure your logging is turned up then try something simpler, like
calling the echo
[Freeswitch-users] Changing User-Agent String
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#User_Agent_.5Buser-agent-string.5D As per the above link, we can change the User Agent String, but I added this param name but does not seem to work. [u...@freeswitch autoload_configs]$ vi sofia.conf.xml configuration name=sofia.conf description=sofia Endpoint global_settings param name=log-level value=0/ !-- param name=auto-restart value=false/ -- param name=debug-presence value=0/ param name=user-agent-string value=Test Server/ /global_settings !-- The rabbit hole goes deep. This includes all the profiles in the sip_profiles directory that is up one level from this directory. -- profiles X-PRE-PROCESS cmd=include data=../sip_profiles/*.xml/ /profiles /configuration ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Simple Conference Setup issue
I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch and when I have 2 people dial in, looks like the Music on Hold never stops. Here is what my public.xml looks like: extension name=test !-- your provider or any name you'd like to call it -- condition field=destination_number expression=xx !-- your DID for this gateway-- action application=conference data=conference.conf+12345/ /condition /extension Help appreciated ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Simple Conference Setup issue
My mistake , it picked the default profile and was waiting for moderator in the conference.cof.xml file that is provided with the install. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, November 10, 2009 3:50 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Simple Conference Setup issue What does your config look like? /b On Nov 10, 2009, at 4:39 PM, Ujjval Karihaloo wrote: I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch and when I have 2 people dial in, looks like the Music on Hold never stops. Here is what my public.xml looks like: extension name=test !-- your provider or any name you'd like to call it -- condition field=destination_number expression=xx !-- your DID for this gateway-- action application=conference data=conference.conf+12345/ /condition /extension ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Setting up Conference with Moderator
OK, I may have solved this mystery, if I use application=answer and answer the call before the IVR which then flows into the Conference app, DTMF works from the ATT phone.. So, if you face issues with Conferencing/IVR, answer the call before you invoke those apps... Problem I have now is that a Polycom old phone like 501s are not doing DTMF 2833 to the Freeswitch server...has anyone seen this..Call is not going thru PSTN...its IP to IP Polycom 501 through our SBC to the Freeswitch Server. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, November 02, 2009 9:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator you know I have heard this before... It seems to ONLY be ATT /b On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote: Yes, I think I did. However here is what furthur testing revelas. If I dial in from ATT cell phone, I do not see any DTMF using Don's IVR.xml.conf to call my conf app. But when I dial the same number using a Verizon Cell, it works. When I dial a number that is provisioned to call the Conf App directly from the public.xml dialplan...it works even with the same ATT cell phone... Strange behaviour ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Setting up Conference with Moderator
Any ideas on the below...has anyone implemented the below:
Once I have the Moderator and Participants logged on, how do I invoke the
moderator previlidges, LIk esay muting everyone/someone or kicking someone out
of the Conf and the like?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval
Karihaloo
Sent: Monday, November 02, 2009 12:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Rob:
Once I have the Moderator and Participants logged on, how do I invoke the
moderator previlidges, LIk esay muting everyone/someone or kicking someone out
of the Conf and the like?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 9:34 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Hm, strange. I haven't seen that before. Can you pastebin your logs
at debug level?
On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
It's strange... a tcpdump tells me that there is no DTMF from my
provider when using IVR, but when I call into a TN that goes
directly into the Conference App, I see DTMF from the provider.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 7:23 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I've never had any problem with that. Is your logging at debug level
so you can see the RECV DTFM in the log/fs_cli? Are you calling from
a SIP phone on the pbx, or via a PSTN provider? Maybe your provider
isn't passing them through.
Make sure your logging is turned up then try something simpler, like
calling the echo application, and see if DTFM comes through.
Rob
On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:
Rob:
For some reason, I don't see the DTMF appear on the fs_CLI when
using the below configurationso it basically timesout.
UK
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 26, 2009 9:21 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra
pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
==
I will try it out.
Again thx a lot for your help. Will keep everyone posted.
From: freeswitch-users-boun...@lists.freeswitch.org
[freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman [rob4manh...@gmail.com]
Sent: Friday, October 23, 2009 12:22 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I just re-tested with the pin in my dial plan:
action application=conference data=conference 123...@default
+flags{}+1234 /
And it doesn't challenge me for the pin. I just drop right in. I
figured this is how it was intended, since the wiki says the pin is
set initially and only challenged in later attempts [by future
callers]:
The first time a conference name (confname) is used, it will be
created on demand, and the pin will be set to what ever is specified
at that time: the pin in the data string if specified, or if not, the
pin setting in the conference profile, and if that is also
unspecified, then there is no pin protection. Any later attempt to
join the conference must specify the same pin number, if one existed
when it was created.
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra
pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
I'm sure there's a scenario where its used and useful, the wiki just
doesn't explain it.
Rob
On Oct 23, 2009, at 12:43 PM, Brian West wrote:
Well first off you're not defining a pine here...
confn...@profilename+flags{mute|deaf|waste|moderator}+[conference
pin
number]
That might be why its not asking for a pin.
/b
On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
entry action=menu-exec-app digits=1 param=conference
123...@default+flags{} /
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Re: [Freeswitch-users] Setting up Conference with Moderator
Was that sarcasm or you really mean it? -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, November 02, 2009 9:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator you know I have heard this before... It seems to ONLY be ATT /b On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote: Yes, I think I did. However here is what furthur testing revelas. If I dial in from ATT cell phone, I do not see any DTMF using Don's IVR.xml.conf to call my conf app. But when I dial the same number using a Verizon Cell, it works. When I dial a number that is provisioned to call the Conf App directly from the public.xml dialplan...it works even with the same ATT cell phone... Strange behaviour ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Setting up Conference with Moderator
Yes, I think I did. However here is what furthur testing revelas. If I dial in from ATT cell phone, I do not see any DTMF using Don's IVR.xml.conf to call my conf app. But when I dial the same number using a Verizon Cell, it works. When I dial a number that is provisioned to call the Conf App directly from the public.xml dialplan...it works even with the same ATT cell phone... Strange behaviour From: freeswitch-users-boun...@lists.freeswitch.org [freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Jerris [m...@jerris.com] Sent: Saturday, October 31, 2009 11:33 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Have you answered the call? On Oct 30, 2009, at 11:34 AM, Rob Forman wrote: Hm, strange. I haven't seen that before. Can you pastebin your logs at debug level? On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: It's strange... a tcpdump tells me that there is no DTMF from my provider when using IVR, but when I call into a TN that goes directly into the Conference App, I see DTMF from the provider. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Setting up Conference with Moderator
Rob:
Once I have the Moderator and Participants logged on, how do I invoke the
moderator previlidges, LIk esay muting everyone/someone or kicking someone out
of the Conf and the like?
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 9:34 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Hm, strange. I haven't seen that before. Can you pastebin your logs
at debug level?
On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
It's strange... a tcpdump tells me that there is no DTMF from my
provider when using IVR, but when I call into a TN that goes
directly into the Conference App, I see DTMF from the provider.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 7:23 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I've never had any problem with that. Is your logging at debug level
so you can see the RECV DTFM in the log/fs_cli? Are you calling from
a SIP phone on the pbx, or via a PSTN provider? Maybe your provider
isn't passing them through.
Make sure your logging is turned up then try something simpler, like
calling the echo application, and see if DTFM comes through.
Rob
On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:
Rob:
For some reason, I don't see the DTMF appear on the fs_CLI when
using the below configurationso it basically timesout.
UK
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 26, 2009 9:21 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra
pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
==
I will try it out.
Again thx a lot for your help. Will keep everyone posted.
From: freeswitch-users-boun...@lists.freeswitch.org
[freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman [rob4manh...@gmail.com]
Sent: Friday, October 23, 2009 12:22 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I just re-tested with the pin in my dial plan:
action application=conference data=conference 123...@default
+flags{}+1234 /
And it doesn't challenge me for the pin. I just drop right in. I
figured this is how it was intended, since the wiki says the pin is
set initially and only challenged in later attempts [by future
callers]:
The first time a conference name (confname) is used, it will be
created on demand, and the pin will be set to what ever is specified
at that time: the pin in the data string if specified, or if not, the
pin setting in the conference profile, and if that is also
unspecified, then there is no pin protection. Any later attempt to
join the conference must specify the same pin number, if one existed
when it was created.
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra
pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
I'm sure there's a scenario where its used and useful, the wiki just
doesn't explain it.
Rob
On Oct 23, 2009, at 12:43 PM, Brian West wrote:
Well first off you're not defining a pine here...
confn...@profilename+flags{mute|deaf|waste|moderator}+[conference
pin
number]
That might be why its not asking for a pin.
/b
On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
entry action=menu-exec-app digits=1 param=conference
123...@default+flags{} /
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Re: [Freeswitch-users] Setting up Conference with Moderator
It's strange... a tcpdump tells me that there is no DTMF from my provider when
using IVR, but when I call into a TN that goes directly into the Conference
App, I see DTMF from the provider.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 7:23 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I've never had any problem with that. Is your logging at debug level
so you can see the RECV DTFM in the log/fs_cli? Are you calling from
a SIP phone on the pbx, or via a PSTN provider? Maybe your provider
isn't passing them through.
Make sure your logging is turned up then try something simpler, like
calling the echo application, and see if DTFM comes through.
Rob
On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:
Rob:
For some reason, I don't see the DTMF appear on the fs_CLI when
using the below configurationso it basically timesout.
UK
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 26, 2009 9:21 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
==
I will try it out.
Again thx a lot for your help. Will keep everyone posted.
From: freeswitch-users-boun...@lists.freeswitch.org
[freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman [rob4manh...@gmail.com]
Sent: Friday, October 23, 2009 12:22 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I just re-tested with the pin in my dial plan:
action application=conference data=conference 123...@default
+flags{}+1234 /
And it doesn't challenge me for the pin. I just drop right in. I
figured this is how it was intended, since the wiki says the pin is
set initially and only challenged in later attempts [by future
callers]:
The first time a conference name (confname) is used, it will be
created on demand, and the pin will be set to what ever is specified
at that time: the pin in the data string if specified, or if not, the
pin setting in the conference profile, and if that is also
unspecified, then there is no pin protection. Any later attempt to
join the conference must specify the same pin number, if one existed
when it was created.
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
I'm sure there's a scenario where its used and useful, the wiki just
doesn't explain it.
Rob
On Oct 23, 2009, at 12:43 PM, Brian West wrote:
Well first off you're not defining a pine here...
confn...@profilename+flags{mute|deaf|waste|moderator}+[conference pin
number]
That might be why its not asking for a pin.
/b
On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
entry action=menu-exec-app digits=1 param=conference
123...@default+flags{} /
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Re: [Freeswitch-users] Setting up Conference with Moderator
Rob:
For some reason, I don't see the DTMF appear on the fs_CLI when using the
below configurationso it basically timesout.
UK
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval
Karihaloo
Sent: Monday, October 26, 2009 9:21 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
==
I will try it out.
Again thx a lot for your help. Will keep everyone posted.
From: freeswitch-users-boun...@lists.freeswitch.org
[freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman
[rob4manh...@gmail.com]
Sent: Friday, October 23, 2009 12:22 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I just re-tested with the pin in my dial plan:
action application=conference data=conference 123...@default
+flags{}+1234 /
And it doesn't challenge me for the pin. I just drop right in. I
figured this is how it was intended, since the wiki says the pin is
set initially and only challenged in later attempts [by future callers]:
The first time a conference name (confname) is used, it will be
created on demand, and the pin will be set to what ever is specified
at that time: the pin in the data string if specified, or if not, the
pin setting in the conference profile, and if that is also
unspecified, then there is no pin protection. Any later attempt to
join the conference must specify the same pin number, if one existed
when it was created.
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
I'm sure there's a scenario where its used and useful, the wiki just
doesn't explain it.
Rob
On Oct 23, 2009, at 12:43 PM, Brian West wrote:
Well first off you're not defining a pine here...
confn...@profilename+flags{mute|deaf|waste|moderator}+[conference pin
number]
That might be why its not asking for a pin.
/b
On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
entry action=menu-exec-app digits=1 param=conference
123...@default+flags{} /
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Re: [Freeswitch-users] Setting up Conference with Moderator
Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
==
I will try it out.
Again thx a lot for your help. Will keep everyone posted.
From: freeswitch-users-boun...@lists.freeswitch.org
[freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman
[rob4manh...@gmail.com]
Sent: Friday, October 23, 2009 12:22 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I just re-tested with the pin in my dial plan:
action application=conference data=conference 123...@default
+flags{}+1234 /
And it doesn't challenge me for the pin. I just drop right in. I
figured this is how it was intended, since the wiki says the pin is
set initially and only challenged in later attempts [by future callers]:
The first time a conference name (confname) is used, it will be
created on demand, and the pin will be set to what ever is specified
at that time: the pin in the data string if specified, or if not, the
pin setting in the conference profile, and if that is also
unspecified, then there is no pin protection. Any later attempt to
join the conference must specify the same pin number, if one existed
when it was created.
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
I'm sure there's a scenario where its used and useful, the wiki just
doesn't explain it.
Rob
On Oct 23, 2009, at 12:43 PM, Brian West wrote:
Well first off you're not defining a pine here...
confn...@profilename+flags{mute|deaf|waste|moderator}+[conference pin
number]
That might be why its not asking for a pin.
/b
On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
entry action=menu-exec-app digits=1 param=conference
123...@default+flags{} /
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[Freeswitch-users] Estimating Call Capacity
With the following spec for CPU and Memory can someone help me guesstimating how many simultaneous calls and Calls/sec a FS server can handle - Used as a Conferencing Server. cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Pentium(R) 4 CPU 2.80GHz stepping: 1 cpu MHz : 2800.386 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 1 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips: 5604.12 processor : 1 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Pentium(R) 4 CPU 2.80GHz stepping: 1 cpu MHz : 2800.386 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 1 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips: 5600.22 # cat /proc/meminfo MemTotal: 1035080 kB MemFree:517972 kB Buffers: 85812 kB Cached: 271264 kB SwapCached: 0 kB Active: 224292 kB Inactive: 223008 kB HighTotal: 130816 kB HighFree:29484 kB LowTotal: 904264 kB LowFree:488488 kB SwapTotal: 2031608 kB SwapFree: 2031520 kB Dirty: 80 kB Writeback: 0 kB AnonPages: 90172 kB Mapped: 39880 kB Slab:60060 kB PageTables: 3232 kB NFS_Unstable:0 kB Bounce: 0 kB CommitLimit: 2549148 kB Committed_AS: 345780 kB VmallocTotal: 114680 kB VmallocUsed: 3584 kB VmallocChunk: 110888 kB HugePages_Total: 0 HugePages_Free: 0 HugePages_Rsvd: 0 Hugepagesize: 4096 kB ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Estimating Call Capacity
Are there any benchmarking test results available publicly? From: freeswitch-users-boun...@lists.freeswitch.org [freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West [br...@freeswitch.org] Sent: Monday, October 26, 2009 11:18 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity You'll have to do your own load testing. Nobody can really tell you exactly how many you'll get. /b On Oct 26, 2009, at 10:39 AM, Ujjval Karihaloo wrote: With the following spec for CPU and Memory can someone help me guesstimating how many simultaneous calls and Calls/sec a FS server can handle - Used as a Conferencing Server. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Setting up Conference with Moderator
One more try for help for COnferencing using moderator,
Anyone successfully accomplished this? Help appreciated!
From: freeswitch-users-boun...@lists.freeswitch.org
[freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval Karihaloo
[ujj...@simplesignal.com]
Sent: Thursday, October 22, 2009 9:01 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Setting up Conference with Moderator
Hi
I have the Basic Conferencing working. Here is my Dial Plan.
I want to be able to setup a Moderator PIN different from other participants,
when I add the moderator flag it logs me in directly w/o asking for a PIN..
action application=conference
data=conference.c...@wideband+flags{moderator}+159753mailto:conference.c...@wideband+flags%7bmoderator%7d+159753/
DialPlan is below for the normal user, and it asks for the PIN with below
settings.
Ujj Inbound from SS - start
--
extension name=simplesignal !-- your provider or any name you'd like
to call it --
condition field=destination_number expression=xx !-- your
DID for this gateway--
action application=conference
data=conference.c...@wideband+159753mailto:conference.c...@wideband+159753/
/condition
/extension
!--
Ujj Inbound from SS – end
And I am using the existing conference.conf.xml file in the auto_loads
directory.
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
[cid:image001.jpg@01CA535A.E43E9870]http://www.simplesignal.com/
inline: image001.jpg___
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[Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has more headers
Hi, I used the downlaoded TAR ball and my calls worked, however, when upgrading to the SVN release...my SBC is rejecting the 200 OK (when the FS answers the call - using Conferencing app).. Here are teh bad and good 200 OKI see a lot of additional headers startin gwith X:FS , can I remove them? Bad one - SIP/2.0 200 OK v: SIP/2.0/UDP x.x.x.210:5060;branch=z9hG4bK1662379c08490b63b0b9be6978d93ebe f: SIMPLE SIGNAL sip:220...@x.x.x.20:5080;user=phone;tag=945d1734da59aac9d071edc798a7 To: trunkgroup user sip:114...@simplesignal.net;ssig=33402ab16f12e359b1f871b80bf7ff1a;tag=gX1Z34952QBpS i: 674babc2dbf2a90da97bd68642865eae-23f...@x.x.x.210mailto:674babc2dbf2a90da97bd68642865eae-23f...@x.x.x.210 CSeq: 3240 INVITE Contact: sip:114...@x.x.x.20:5080;transport=udp User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-FS-Display-Name: +1114847 X-FS-Display-Number: +1114847 X-FS-Support: update_display P-Asserted-Identity: +1114847 +1114847 v=0 o=FreeSWITCH 1256498762 1256498763 IN IP4 x.x.x.20 s=FreeSWITCH c=IN IP4 x.x.x.20 t=0 0 m=audio 29406 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 -- Good One --- SIP/2.0 200 OK v: SIP/2.0/UDP x.x.x.210:5060;branch=z9hG4bKffebde0efa06a6aa7772bae5ca085627 f: SIMPLE SIGNAL sip:220...@x.x.x.20:5080;user=phone;tag=895fd35d4979e6e5b0cdcc795b8c8df2 To: trunkgroup user sip:114...@simplesignal.net;ssig=33402ab16f12e359b1f871b80bf7ff1a;tag=KBXKrBQU99c6D i: 28f0d47f1163e0d478247e6729762d98-23f...@x.x.x.210mailto:28f0d47f1163e0d478247e6729762d98-23f...@x.x.x.210 CSeq: 3278 INVITE Contact: sip:114...@x.x.x.20:5080;transport=udp User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 v=0 o=FreeSWITCH 1256512215 1256512216 IN IP4 x.x.x.20 s=FreeSWITCH c=IN IP4 x.x.x.20 t=0 0 m=audio 17376 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Setting up Conference with Moderator
Any ideas on this one. Look slike only way rite now is to have a different Dest
Phone number for a moderator.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval
Karihaloo
Sent: Thursday, October 22, 2009 9:02 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Setting up Conference with Moderator
Hi
I have the Basic Conferencing working. Here is my Dial Plan.
I want to be able to setup a Moderator PIN different from other participants,
when I add the moderator flag it logs me in directly w/o asking for a PIN..
action application=conference
data=conference.c...@wideband+flags{moderator}+159753mailto:conference.c...@wideband+flags%7bmoderator%7d+159753/
DialPlan is below for the normal user, and it asks for the PIN with below
settings.
Ujj Inbound from SS - start
--
extension name=simplesignal !-- your provider or any name you'd like
to call it --
condition field=destination_number expression=2142349127 !-- your
DID for this gateway--
action application=conference
data=conference.c...@wideband+159753mailto:conference.c...@wideband+159753/
/condition
/extension
!--
Ujj Inbound from SS - end
And I am using the existing conference.conf.xml file in the auto_loads
directory.
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
[cid:image001.jpg@01CA53CF.3D2B8B40]http://www.simplesignal.com/
inline: image001.jpg___
FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
[Freeswitch-users] Connecting to FS CLI...just hangs..
It just hangsand I CTRL-C out of it. [r...@ss]# ./fs_cli -H 127.0.0.1 ^C [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error] Freeswitch process is running: [r...@ss bin]# ps -ef|grep free root 8889 31039 0 12:36 pts/200:00:00 ./freeswitch root 8952 31039 0 12:42 pts/200:00:00 grep free ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Core Dump question!
freeswi...@ss_freeswitch sofia_gateway_data Segmentation fault (core dumped) Just ran the gateway command above w/o any parameters,,, and it core dumped.. I am sure mistakes like that happen...but I not sure if it should core dump and shutdown. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting to FS CLI...just hangs..
If I run .freeswitch , get back to the ROOT prompt and then from same window type in fs_cli...it fails...hangs forever However, if I open another new ssh session and fs_cli from the new session, it works. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of João Mesquita Sent: Thursday, October 22, 2009 1:11 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Connecting to FS CLI...just hangs.. Hangs for how long? Are you sure you are not just waiting on a timeout? JM On Thu, Oct 22, 2009 at 4:42 PM, Ujjval Karihaloo ujj...@simplesignal.commailto:ujj...@simplesignal.com wrote: It just hangsand I CTRL-C out of it. [r...@ss]# ./fs_cli -H 127.0.0.1 ^C [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error] Freeswitch process is running: [r...@ss bin]# ps -ef|grep free root 8889 31039 0 12:36 pts/200:00:00 ./freeswitch root 8952 31039 0 12:42 pts/200:00:00 grep free ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto:FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] FS Registration Contact
Hi All, I have FS registered to an ITSP. The contact is showing as follows.. Contact: sip:gw+i...@1.1.1.1:5080;transport=udp Itsp is the name of the SIP gatway and its IP is changed to 1.1.1.1 I want the Phone number (FromUser)to show in the contact header in the REGISTER msg going to the ITSP. How can I do that? Thx, Ujjval. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Registration Contact
Ok I got this one...just put ext-in-contact setting and then define the extension to be same as FromUser in my provider.xml in /conf/sip-profile/external/ !--/// extension for inbound calls: *optional* same as username, if blank ///-- param name=extension value=xx/ !--extra sip params to send in the contact-- param name=extension-in-contact value=true/ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval Karihaloo Sent: Thursday, October 22, 2009 2:00 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] FS Registration Contact Hi All, I have FS registered to an ITSP. The contact is showing as follows.. Contact: sip:gw+i...@1.1.1.1:5080;transport=udp Itsp is the name of the SIP gatway and its IP is changed to 1.1.1.1 I want the Phone number (FromUser)to show in the contact header in the REGISTER msg going to the ITSP. How can I do that? Thx, Ujjval. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Core Dump question!
I do have the core dump, should I open a ticket. I am running latest Freeswitch 1.0.4 and had done a make current just before it happened. Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [cid:image001.jpg@01CA5342.1F8A8880]http://www.simplesignal.com/ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Thursday, October 22, 2009 2:53 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Core Dump question! I did test this on trunk and it seems to work right: freeswi...@default sofia_gateway_data -ERR Parameter missing Mike On Oct 22, 2009, at 3:58 PM, Michael Collins wrote: What SVN rev of FS? What operating system? If you're not on the latest then do a make current and get to the latest SVN and see if you can replicate the issue. -MC On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo ujj...@simplesignal.commailto:ujj...@simplesignal.com wrote: freeswi...@ss_freeswitch sofia_gateway_data Segmentation fault (core dumped) Just ran the gateway command above w/o any parameters,,, and it core dumped.. I am sure mistakes like that happen...but I not sure if it should core dump and shutdown. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto:FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org inline: image001.jpg___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Core Dump question!
How do I tell if it's the latest...I downloaded is yesterday..and installed it from freeswitch.org Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [cid:image001.jpg@01CA5359.87CB41C0]http://www.simplesignal.com/ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, October 22, 2009 6:26 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Core Dump question! Yes, if this is latest SVN (after a make current) then open a jira. -MC On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo ujj...@simplesignal.commailto:ujj...@simplesignal.com wrote: I do have the core dump, should I open a ticket. I am running latest Freeswitch 1.0.4 and had done a make current just before it happened. inline: image001.jpg___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Setting up Conference with Moderator
Hi
I have the Basic Conferencing working. Here is my Dial Plan.
I want to be able to setup a Moderator PIN different from other participants,
when I add the moderator flag it logs me in directly w/o asking for a PIN..
action application=conference
data=conference.c...@wideband+flags{moderator}+159753mailto:conference.c...@wideband+flags%7bmoderator%7d+159753/
DialPlan is below for the normal user, and it asks for the PIN with below
settings.
Ujj Inbound from SS - start
--
extension name=simplesignal !-- your provider or any name you'd like
to call it --
condition field=destination_number expression=2142349127 !-- your
DID for this gateway--
action application=conference
data=conference.c...@wideband+159753mailto:conference.c...@wideband+159753/
/condition
/extension
!--
Ujj Inbound from SS - end
And I am using the existing conference.conf.xml file in the auto_loads
directory.
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
[cid:image001.jpg@01CA535A.E43E9870]http://www.simplesignal.com/
inline: image001.jpg___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
[Freeswitch-users] Asterisk vs Freeswitch
Is there benchmark test results on how many simultaneous calls Freeswtich can do (with RTP anchored through it) vs the Asterisk. For any hardware/CPU/Mem that anyone may have performed this performance testing. Any numbers on average how much Freeswitch scores over the Asterisk in terms of capacity will help. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Conferencing setup with FS
Hi , I cannot seem to find a Document online for setting up conferencingon FreeSwitch. Can someone point me to one? Thx. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] New install
Hi, I just installed freeswitch as a replacement for our Asterisk Server. I want to untimately do Conferencing with it as I have heard is it pretty good at it. I have it compiled and up and running. However, when I provision a Sofphone/Xlite to register with it to run basic tests, it does not seem to register. Looked at freeswitch.log but doesn't have anything related to the REGISTER requests from Xlite. Not too familiar with CLI or configg files yet. Help is appreciated. Also: If there a howto to setup a conferencing Bridge on it. Thx, Ujjval. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] New install
Would that be firewall on the CentOS machine that FS is installed on? From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, September 04, 2009 5:35 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] New install make sure your firewall is not up /b On Sep 4, 2009, at 5:15 PM, Ujjval Karihaloo wrote: Hi, I just installed freeswitch as a replacement for our Asterisk Server. I want to untimately do Conferencing with it as I have heard is it pretty good at it. I have it compiled and up and running. However, when I provision a Sofphone/Xlite to register with it to run basic tests, it does not seem to register. Looked at freeswitch.log but doesn't have anything related to the REGISTER requests from Xlite. Not too familiar with CLI or configg files yet. Help is appreciated. Also: If there a howto to setup a conferencing Bridge on it. Thx, Ujjval. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
