date:20090904

On Thu, Sep 3, 2009 at 11:05 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:

 Hi,

 I'm newbie in FS. As far as I know for setting up custom variables in FS we
 use this syntex  in dialplan XML i.e.

 action application=set data=ABC='value'/

 But when I call this variable using eval application i.e.

 action application=eval data=${ABC}/

 the value I get from variable ABC is undefined means no values are passed
 to the variable.

 So kindly do let me know how I can pass values in variables in FS.


Try using the info app instead of eval. Most likely this is just a case of
the dialplan being parsed prior to the variable being assigned. Do this in
your dialplan after you set the variable:
action application=info/

You'll see a whole list of variables and your custom variable(s) should be
shown with their values.
-MC
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Re: [Freeswitch-users] XML Dial Plan vs Language Modules

On Thu, Sep 3, 2009 at 10:59 PM, Muhammad Shahzad 
shaherya...@googlemail.com wrote:

 Hi,

 I couple of my team members are working on translating a very long Asterisk
 Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link below,


Before you go through all the trouble of translating the dialplan be sure to
review the application itself. In many cases just doing a dialplan
translation results in less efficient use of FreeSWITCH's powerful features.
Be sure that you are looking at the way FreeSWITCH handles various
situations and take advantage of its power and ease of use.



 http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables

 The dial plan variables are not getting initialized as expected. I was just
 wondering if we move this variable get and set stuff to any language module
 say mod_perl, will that make any difference performance wise? I mean we will
 be invoking a Perl interpreter for each incoming call, won't that be
 expensive in terms of RAM and CPU usage and thus reducing number of calls
 this FS deployment can handle?

 I have guys with programming skills in Perl, PHP, Python, Java and LUA
 languages. Which language do you recommend for this, again in terms of speed
 and performance?


Lua is very portable and we've done tests with hundreds of concurrent Lua
scripts running. The other languages are heavier but they'll still handle
quite a few concurrent sessions. Just be sure that you don't do the bridge
app right in the script, use transfer instead and have the dialplan process
any bridging that you need to do.

-MC
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Re: [Freeswitch-users] restart when convenient

2009-09-04 Thread João Mesquita

Look at the fsctl api on the wiki. It has what you need.
jmesquita

On 9/4/09, Christian Löschenkohl christian.loeschenk...@xpirio.com wrote:
 hello

 i'm looking for a possibility to restart freeswitch like it is possible with
 asterisk.
 for me i tried to created a script that looks for open channels and if no
 channel
 is open it restarts freeswitch with the init script (not the most efficient
 way).

 i think i would be great if we would have a buildin function for this, i
 think such
 command would help with maintenance and not only for me.

 br

 --
 Ing. Christian Löschenkohl
 Technische Leitung, Forschung  Entwicklung VoIP

 xpirio
 Telekommunikation  Service GmbH
 Lakeside B04
 9020 Klagenfurt
 Austria

 T  +43 (0) 5 77 11 - 1000
 F  +43 (0) 5 77 11 - 1002
 E  christian.loeschenk...@xpirio.com

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Re: [Freeswitch-users] restart when convenient

2009/9/3 Christian Löschenkohl christian.loeschenk...@xpirio.com

 hello

 i'm looking for a possibility to restart freeswitch like it is possible
 with
 asterisk.
 for me i tried to created a script that looks for open channels and if no
 channel
 is open it restarts freeswitch with the init script (not the most efficient
 way).

 i think i would be great if we would have a buildin function for this, i
 think such
 command would help with maintenance and not only for me.

 br

 Like JM said, the fsctl API can help. If you're in Linux you can do a shell
script with a command like ths:

fs_cli -x 'fsctl shutdown restart'

-MC
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Re: [Freeswitch-users] XML Dial Plan vs Language Modules

2009-09-04 Thread Muhammad Shahzad

Thank you so much. Of course we are not doing a blind translation, but at
the very basic we will need to get and set certain variable at different
stage of call processing.

Another question in same context, Can we do post-hangup call processing? I
mean like in Asterisk, we have extension h which is called after hangup.
Can you guide a bit how to do it in FS? Does FS has any such special
extensions?

Thank you.


On Fri, Sep 4, 2009 at 12:06 PM, Michael Collins m...@freeswitch.org wrote:



 On Thu, Sep 3, 2009 at 10:59 PM, Muhammad Shahzad 
 shaherya...@googlemail.com wrote:

 Hi,

 I couple of my team members are working on translating a very long
 Asterisk Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link
 below,


 Before you go through all the trouble of translating the dialplan be sure
 to review the application itself. In many cases just doing a dialplan
 translation results in less efficient use of FreeSWITCH's powerful features.
 Be sure that you are looking at the way FreeSWITCH handles various
 situations and take advantage of its power and ease of use.



 http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables

 The dial plan variables are not getting initialized as expected. I was
 just wondering if we move this variable get and set stuff to any language
 module say mod_perl, will that make any difference performance wise? I mean
 we will be invoking a Perl interpreter for each incoming call, won't that be
 expensive in terms of RAM and CPU usage and thus reducing number of calls
 this FS deployment can handle?

 I have guys with programming skills in Perl, PHP, Python, Java and LUA
 languages. Which language do you recommend for this, again in terms of speed
 and performance?


 Lua is very portable and we've done tests with hundreds of concurrent Lua
 scripts running. The other languages are heavier but they'll still handle
 quite a few concurrent sessions. Just be sure that you don't do the bridge
 app right in the script, use transfer instead and have the dialplan process
 any bridging that you need to do.

 -MC


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-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
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Re: [Freeswitch-users] XML Dial Plan vs Language Modules

On Fri, Sep 4, 2009 at 12:25 AM, Muhammad Shahzad 
shaherya...@googlemail.com wrote:

 Thank you so much. Of course we are not doing a blind translation, but at
 the very basic we will need to get and set certain variable at different
 stage of call processing.


 Another question in same context, Can we do post-hangup call processing? I
 mean like in Asterisk, we have extension h which is called after hangup.
 Can you guide a bit how to do it in FS? Does FS has any such special
 extensions?

 Thank you.


Yes, you can post hangup processing. See the wiki channel_variables page and
look at api_hangup_hook for more information. Just know that it can get
tricky to try and post-process calls from right inside the dialplan. In most
cases we recommend using the event socket and having absolute control over
the call, including what happens at hangup. Lua is especially good at this.

-MC
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Re: [Freeswitch-users] restart when convenient

2009-09-04 Thread Anatoliy Kounitskiy

After some testing (fs_cli -x 'fsctl shutdown restart') I'm seeing
that all active calls are dropped and the freeswitch is restarted



On Fri, Sep 4, 2009 at 10:14 AM, Michael Collinsm...@freeswitch.org wrote:


 2009/9/3 Christian Löschenkohl christian.loeschenk...@xpirio.com

 hello

 i'm looking for a possibility to restart freeswitch like it is possible
 with
 asterisk.
 for me i tried to created a script that looks for open channels and if no
 channel
 is open it restarts freeswitch with the init script (not the most
 efficient way).

 i think i would be great if we would have a buildin function for this, i
 think such
 command would help with maintenance and not only for me.

 br

 Like JM said, the fsctl API can help. If you're in Linux you can do a shell
 script with a command like ths:

 fs_cli -x 'fsctl shutdown restart'

 -MC


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-- 
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-
E-mail: anato...@kounitskiy.com
Mobile: +359898913540

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Re: [Freeswitch-users] restart when convenient

2009-09-04 Thread Seven Du


freeswi...@foosball fsctl
-USAGE: [send_sighup|hupall|pause|resume|shutdown [cancel|elegant|asap| 
restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration  
[num]|loglevel [level]]


On Sep 4, 2009, at 3:35 PM, Anatoliy Kounitskiy wrote:
 After some testing (fs_cli -x 'fsctl shutdown restart') I'm seeing
 that all active calls are dropped and the freeswitch is restarted



 On Fri, Sep 4, 2009 at 10:14 AM, Michael Collinsm...@freeswitch.org  
 wrote:


 2009/9/3 Christian Löschenkohl christian.loeschenk...@xpirio.com

 hello

 i'm looking for a possibility to restart freeswitch like it is  
 possible
 with
 asterisk.
 for me i tried to created a script that looks for open channels  
 and if no
 channel
 is open it restarts freeswitch with the init script (not the most
 efficient way).

 i think i would be great if we would have a buildin function for  
 this, i
 think such
 command would help with maintenance and not only for me.

 br

 Like JM said, the fsctl API can help. If you're in Linux you can do  
 a shell
 script with a command like ths:

 fs_cli -x 'fsctl shutdown restart'

 -MC


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 -- 
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 -
 E-mail: anato...@kounitskiy.com
 Mobile: +359898913540

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[Freeswitch-users] skypiax error

2009-09-04 Thread Jingwei Yang

Hi Folks,

I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine.
After having followed the big help doc from the wiki page (
http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch),
I hit an error when running multi.sh (under
freeswitch/src/mod/endpoints/mod_skypiax/configs/multiple-instance-same-skype-username).

*Couldn't open RGB_DB '/usr/share/X11/rgb'
error opening security policy file /usr/lib64/xserver/SecurityPolicy
*

This error seems not stopping xvfb from getting started. Then I started FS
and loaded mod_skypiax. However, when I initiated a test call (originate
skypiax/ANY/userAAA echo), I saw a bunch of ALSA lib errors popping up:

*ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi*

Again, this error doesn't prohibit the call from reaching me. It's just too
annoying and it keeps popping up after a while. Does anyone know how to get
rid of those errors?

I found a similar post here:
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/013956.html.
Phil, if you happen to see my question, could you please reply and let me
know what the cause was and what you have done to solve it? Thanks!

By the way, I started xvfb and FS using root.

Regards,
-Jingwei
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Re: [Freeswitch-users] skypiax error

Jingwei,

those are normal warnings made by the Skype client (not by
mod_skypiax), you just have to edit /etc/alsa/alsa.conf and comment
out hdmi lines. Is a problem with a lazy implementation of that file,
that supposes you got an hdmi.

The other warning is because there are some files missing from the
Xvfb installation made by centos, but are completely harmless. In the
future I will make the script to redirect them to /dev/null :-)

Bottom line: all is OK.

-giovanni

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

On Fri, Sep 4, 2009 at 10:01 AM, Jingwei Yangjingwei.y...@gmail.com wrote:
Hi Folks,

I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine.
After having followed the big help doc from the wiki page
(http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch),
I hit an error when running multi.sh (under
freeswitch/src/mod/endpoints/mod_skypiax/configs/multiple-instance-same-skype-username).

Couldn't open RGB_DB '/usr/share/X11/rgb'
error opening security policy file /usr/lib64/xserver/SecurityPolicy

ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi

Again, this error doesn't prohibit the call from reaching me. It's just too
annoying and it keeps popping up after a while. Does anyone know how to get
rid of those errors?

By the way, I started xvfb and FS using root.

Regards,
-Jingwei

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Re: [Freeswitch-users] skypiax error

2009-09-04 Thread Jingwei Yang

Hi Giovanni,

That's a big relief. Thanks a lot for the reply :)

Regards,
-Jingwei

On Fri, Sep 4, 2009 at 4:25 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:

 Jingwei,

 those are normal warnings made by the Skype client (not by
 mod_skypiax), you just have to edit /etc/alsa/alsa.conf and comment
 out hdmi lines. Is a problem with a lazy implementation of that file,
 that supposes you got an hdmi.

 The other warning is because there are some files missing from the
 Xvfb installation made by centos, but are completely harmless. In the
 future I will make the script to redirect them to /dev/null :-)

 Bottom line: all is OK.

 -giovanni

 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618




 On Fri, Sep 4, 2009 at 10:01 AM, Jingwei Yangjingwei.y...@gmail.com
 wrote:
  Hi Folks,
 
  I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine.
  After having followed the big help doc from the wiki page
  (
 http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch
 ),
  I hit an error when running multi.sh (under
 
 freeswitch/src/mod/endpoints/mod_skypiax/configs/multiple-instance-same-skype-username).
 
  Couldn't open RGB_DB '/usr/share/X11/rgb'
  error opening security policy file /usr/lib64/xserver/SecurityPolicy
 
  This error seems not stopping xvfb from getting started. Then I started
 FS
  and loaded mod_skypiax. However, when I initiated a test call (originate
  skypiax/ANY/userAAA echo), I saw a bunch of ALSA lib errors popping up:
 
  ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
 
  Again, this error doesn't prohibit the call from reaching me. It's just
 too
  annoying and it keeps popping up after a while. Does anyone know how to
 get
  rid of those errors?
 
  I found a similar post here:
 
 http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/013956.html
 .
  Phil, if you happen to see my question, could you please reply and let me
  know what the cause was and what you have done to solve it? Thanks!
 
  By the way, I started xvfb and FS using root.
 
  Regards,
  -Jingwei
 
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Re: [Freeswitch-users] Proxy authorization

On Fri, Sep 4, 2009 at 1:19 AM, Brian Westbr...@freeswitch.org wrote:
 There will not be an authorization header on the first register attempt...
 it only happens once we are 401/407'ed and the phone comes back and
 registers again.
 /b

Alas, I cannot change the way the provider's gateway works. It
immediately responses with 403... BTW, it's Mera Damos
(http://www.mera-systems.com ?). No workaround possible?

 On Sep 3, 2009, at 3:26 PM, Dmitry Bely wrote:

 Unfortunately even after that there is no Authorization: header in
 the REGISTER message:

   REGISTER sip:1.2.3.4 SIP/2.0
   Via: SIP/2.0/UDP 5.6.7.8:5080;rport;branch=z9hG4bKNBB3ygD85y3eF
   Max-Forwards: 70
   From: sip:...@domain;transport=udp;tag=Nrc6Z9yrNBS3H
   To: sip:...@domain;transport=udp
   Call-ID: a93d949a-98c1-11de-b6b8-8321249ad8d4
   CSeq: 119885384 REGISTER
   Contact: sip:gw+1.2@5.6.7.8:5080;transport=udp
   Expires: 600
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14707M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
 REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0

- Dmitry Bely

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Re: [Freeswitch-users] skypiax error

:-) My fault, I would have to document this.

I'll do pretty soon.

Sorry about that, and thanks for reporting!!!

-gm


Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Fri, Sep 4, 2009 at 10:38 AM, Jingwei Yangjingwei.y...@gmail.com wrote:
 Hi Giovanni,

 That's a big relief. Thanks a lot for the reply :)

 Regards,
 -Jingwei

 On Fri, Sep 4, 2009 at 4:25 PM, Giovanni Maruzzelli gmar...@celliax.org
 wrote:

 Jingwei,

 those are normal warnings made by the Skype client (not by
 mod_skypiax), you just have to edit /etc/alsa/alsa.conf and comment
 out hdmi lines. Is a problem with a lazy implementation of that file,
 that supposes you got an hdmi.

 The other warning is because there are some files missing from the
 Xvfb installation made by centos, but are completely harmless. In the
 future I will make the script to redirect them to /dev/null :-)

 Bottom line: all is OK.

 -giovanni

 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618




 On Fri, Sep 4, 2009 at 10:01 AM, Jingwei Yangjingwei.y...@gmail.com
 wrote:
  Hi Folks,
 
  I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine.
  After having followed the big help doc from the wiki page
 
  (http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch),
  I hit an error when running multi.sh (under
 
  freeswitch/src/mod/endpoints/mod_skypiax/configs/multiple-instance-same-skype-username).
 
  Couldn't open RGB_DB '/usr/share/X11/rgb'
  error opening security policy file /usr/lib64/xserver/SecurityPolicy
 
  This error seems not stopping xvfb from getting started. Then I started
  FS
  and loaded mod_skypiax. However, when I initiated a test call (originate
  skypiax/ANY/userAAA echo), I saw a bunch of ALSA lib errors popping up:
 
  ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
 
  Again, this error doesn't prohibit the call from reaching me. It's just
  too
  annoying and it keeps popping up after a while. Does anyone know how to
  get
  rid of those errors?
 
  I found a similar post here:
 
  http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/013956.html.
  Phil, if you happen to see my question, could you please reply and let
  me
  know what the cause was and what you have done to solve it? Thanks!
 
  By the way, I started xvfb and FS using root.
 
  Regards,
  -Jingwei
 
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Re: [Freeswitch-users] restart when convenient

2009-09-04 Thread Anatoliy Kounitskiy

What you're looking for is :

fs_cli -x 'fsctl shutdown elegant restart'

:)

It will restart the freeswitch after all calls are hanged up.


On Fri, Sep 4, 2009 at 10:46 AM, Seven Dudujinf...@gmail.com wrote:

 freeswi...@foosball fsctl
 -USAGE: [send_sighup|hupall|pause|resume|shutdown [cancel|elegant|asap|
 restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration
 [num]|loglevel [level]]


 On Sep 4, 2009, at 3:35 PM, Anatoliy Kounitskiy wrote:
 After some testing (fs_cli -x 'fsctl shutdown restart') I'm seeing
 that all active calls are dropped and the freeswitch is restarted



 On Fri, Sep 4, 2009 at 10:14 AM, Michael Collinsm...@freeswitch.org
 wrote:


 2009/9/3 Christian Löschenkohl christian.loeschenk...@xpirio.com

 hello

 i'm looking for a possibility to restart freeswitch like it is
 possible
 with
 asterisk.
 for me i tried to created a script that looks for open channels
 and if no
 channel
 is open it restarts freeswitch with the init script (not the most
 efficient way).

 i think i would be great if we would have a buildin function for
 this, i
 think such
 command would help with maintenance and not only for me.

 br

 Like JM said, the fsctl API can help. If you're in Linux you can do
 a shell
 script with a command like ths:

 fs_cli -x 'fsctl shutdown restart'

 -MC


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 --
 Anatoliy Kounitskiy
 -
 E-mail: anato...@kounitskiy.com
 Mobile: +359898913540

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-- 
Anatoliy Kounitskiy
-
E-mail: anato...@kounitskiy.com
Mobile: +359898913540

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Re: [Freeswitch-users] mod_xml_curl.c Oversized file detected [1056100 bytes]

2009-09-04 Thread Peter P GMX

Thanks Anthony,

that did the trick.

Best regards
Peter

Anthony Minessale schrieb:
 you can edit mod_xml_curl.c line 64
 and increase XML_CURL_MAX_BYTES


 On Thu, Sep 3, 2009 at 12:31 PM, Peter P GMX prometheus...@gmx.net
 mailto:prometheus...@gmx.net wrote:

 Hello,

 in a B2BUA scenario we have 2000 defined gateways (defined but not
 registered yet).
 When reloading mod_sofia Freeswitch complains about the XML-Curl File
 size  1MB and deactivates all gateways:
mod_xml_curl.c:121 Oversized file detected [1056100 bytes]

 Is there any way to overcome this? Currently we have 2000 gateways
 defined. Finally we will have about 10.000. And we will not be able to
 reduce the file size below 1 MB. It will become ~ 2-3 MB maybe.

 Best Regards
 Peter

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 -- 
 Anthony Minessale II

 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 Twitter: http://twitter.com/FreeSWITCH_wire

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

 FreeSWITCH Developer Conference
 sip:8...@conference.freeswitch.org
 mailto:sip%3a...@conference.freeswitch.org
 iax:gu...@conference.freeswitch.org/888
 http://iax:gu...@conference.freeswitch.org/888
 googletalk:conf+...@conference.freeswitch.org
 mailto:googletalk%3aconf%2b...@conference.freeswitch.org
 pstn:213-799-1400
 

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Re: [Freeswitch-users] skypiax error

On Fri, Sep 4, 2009 at 10:56 AM, Giovanni Maruzzelligmar...@celliax.org wrote:
 :-) My fault, I would have to document this.

http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#Error_and_warnings_at_the_starting_of_Skype_clients_on_Linux

-giovanni

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Re: [Freeswitch-users] skypiax error

Updated the wiki page with references to other errors/warnings as well :-)

-giovanni

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Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.

2009-09-04 Thread mayamatakeshi

On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi mayamatake...@gmail.comwrote:

 Hello,
 I'm testing FS support for the header Path (FS is behind opensips).
 It pretty much works: I tested calling from one user to the other and calls
 work perfectly.
 However, I've noticed that when I register my terminal directly with FS
 without going thru the proxy, I receive an unsolicited NOTIFY containing
 Message-Waiting information. But when I register via proxy, FS doesn't send
 this NOTIFY.
 What could be causing this difference of behavior? (enabling debug (F8)
 doesn't show anything for registration handling).


I have enabled Sofia debug and I can see NTA is complaining about invalid
URI when building the NOTIFY:

nua: nua_notify: entering
nua(0x9b3c1e8): sent signal r_notify
nua(0x9b3c1e8): recv signal r_notify
nua: nua_stack_set_params: entering
nua(0x9b3c1e8): adding notify usage with event message-summary
nta_leg_tcreate(0x9b74c68)
nta outgoing create: invalid URI
nta: outgoing_free(0x9b74928)
nua(0x9b3c1e8): event r_notify 900 Internal error at nua_client.c:711
nua(0x9b3c1e8): removing notify usage with event message-summary

My REGISTER relayed by opensips is this:

REGISTER sip:test.com SIP/2.0
Record-Route:
sip:192.168.2.100;lr=on;ftag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5
Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKe62c.49e9f6c1.0
Via: SIP/2.0/UDP 192.168.2.121:5060
;received=192.168.2.121;rport=5060;branch=z9hG4bKPj4uAYgDuRbilYy4lCWcjlDKIDAtf-9RdS
Max-Forwards: 69
From: sip:us...@test.com sip%3aus...@test.com
;tag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5
To: sip:us...@test.com sip%3aus...@test.com
Call-ID: JvQ.apMLiJtfHa7z4ShIfgBPi5jIbtBv
CSeq: 14872 REGISTER
Contact: sip:us...@192.168.2.121:5060;nat=yes
Expires: 60
Authorization: Digest username=user1, realm=test.com,
nonce=7d911eef-2c16-4deb-99f6-afcff9968a19, uri=sip:192.168.2.100,
response=df29caeb78790b4527f1176622cbf192, algorithm=MD5,
cnonce=5.EXCbM3RZTx6iOh1cvUzUvEZTs2eheG, qop=auth, nc=0001
Content-Length:  0
Path: sip:opens...@192.168.2.100 sip%3aopens...@192.168.2.100
;lr;received=sip:192.168.2.121:5060

I hope someone can point out a problem.
I'm looking at NTA with gdb but I'm slow on this.
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Re: [Freeswitch-users] skypiax error

2009-09-04 Thread Jingwei Yang

That's efficient :) By the way, do you have any idea about this warning?

ALSA lib pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to open slave


On Fri, Sep 4, 2009 at 5:47 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:

 On Fri, Sep 4, 2009 at 10:56 AM, Giovanni Maruzzelligmar...@celliax.org
 wrote:
  :-) My fault, I would have to document this.


 http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#Error_and_warnings_at_the_starting_of_Skype_clients_on_Linux

 -giovanni

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Re: [Freeswitch-users] Set disable-transcoding in dialplan

I had a similar problem when I needed to talk to a gateway using g729
while g711 was used by default. The following works for me:

vars.xml
(...)
  X-PRE-PROCESS cmd=set
data=global_codec_prefs=PCMU,PCMA,g7...@32000h,g7...@16000h,G722,GSM,G729,G723/
  X-PRE-PROCESS cmd=set data=outbound_codec_prefs=PCMU,PCMA,GSM,G729,G723/

sip_profiles/internal.xml
(...)
param name=inbound-late-negotiation value=true/

dialplan/default/01_example.com.xml
(...)
  action application=set data=absolute_codec_string=G729/
  action application=bridge
data={absolute_codec_string='G729'}sofia/gateway/${default_gateway}/$1/

On Fri, Sep 4, 2009 at 9:54 AM, rodkawa...@laposte.net wrote:
 Hi Nandy,

 yes already tried this, but if I use proxy_media=true, FS makes no
 control on the content of the RTP stream. But the pbm is that I need to
 use this:
 http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate
 This function enables transcoding of SIP_INFO or RFC2833 to Inband DTMF
 in G711

 But this feature doesn't work if I'm using proxy_media=true.

 In fact my setup is the following:

 CPE using G711A, G729 and SIP INFO for DTMF
 PEER_A using G729 only and RFC_2833
 PEER_B using G711 and SIP INFO

 I have been able to make this works, with proxy_media=true for PEER_B
 cause I don't need transcoding of DTMF (SIP INFO to SIP INFO).
 For PEER_A, proxy_media is set to false (default) cause  I need
 transcoding SIP INFO to RFC2833. I'm able to use G729 using
 codec_negotiation=greedy and setting G729 with highest priority on my
 internal profile.

 But the pbm is that I need to add PEER_C.
 PEER_C needs G711 with transcoding DTMF from SIP_INFO to Inband.

 And this is where I'm stuck, cause using greedy settings and G729 with
 priority 1 in my codec list and proxy_media=false force FS to negotiate
 G729 on leg A. But Leg B is willing to use G711 and FS is unable to
 transcode G729 --- G711.

 I was wondering if there is a way for FS to force the codec order on Leg
 A with some knowledge of the preferred codec on Leg B, ie I know that
 Leg B will always use G711 so that I want to biase the SDP answer on Leg
 A based on this fact.

 regards,
 rod

 Nandy Dagondon a écrit :
 rod,

 have you tried this?
 http://lists.freeswitch.org/pipermail/freeswitch-users/2008-March/002199.html

 /nandy


 On Thu, Sep 3, 2009 at 2:50 PM, rod kawa...@laposte.net
 mailto:kawa...@laposte.net wrote:

     Hi Michael,

     I did some tests but I haven't been successful, so there is what I'm
     trying to achieve:

     On A leg, my phone is using: PCMA and G729 (in this priority order)

     With PEER A, I want to use only G729 (thats is the only codec that
     this
     PEER support), so that the RTP flow will be:
        Phone-G729FS-G729-PEER_A

     With PEER B, I want to use only G711, so:
        Phone-G711FS-G711-PEER_B

     In fact, I'd like to force FS announcing the codec list priority based
     on the priority of the codec announced by the PEER, cause FS is unable
     to transcode G729 -- G711.

     Tried a lot of things (greedy for codec-negociation, late_codec,
     disable_transcoding, codec-prefs) without success.

     If you have some clue.

     regards,
     rod

     Michael Collins a écrit :
      Check out this page:
      http://wiki.freeswitch.org/wiki/Codec_negotiation
     
      Late negotiation will probably let you handle all the cases you
     need.
      -MC
     
      On Mon, Aug 31, 2009 at 8:00 AM, rod kawa...@laposte.net
     mailto:kawa...@laposte.net
      mailto:kawa...@laposte.net mailto:kawa...@laposte.net wrote:
     
          Hi all,
     
          I'm wondering if I can do something like this:
             - in my internal profile, I have this because of some PEER
          using G729:
                   - param name=disable-transcoding value=true/
     
          But for a specific PEER, I'd like to activate transcoding:
                   - for this PEER, only G711 is used
                   - I'd like to transcode DTMF SIP INFO or RFC2833 to
     INBAND
     
          So in my dialplan, I tried before bridging:
     
             - action application=set
     data=disable-transcoding=false/
             - action application=start_dtmf_generate data=true/
     
          But I still see RFC2833 events between my FS and PEER and
     the DTMF are
          not working.
     
          So 2 questions:
             - does application start_dtmf_generate requires transcoding
             - if yes, can I set the variable disable-transcoding in
     my dialplan
     
          regards,
          rod

- Dmitry Bely

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[Freeswitch-users] mod_dingaling: dl_login command syntax

2009-09-04 Thread Muhammad Shahzad

Hi,

i am have FS SVN revision 14760, i am trying to use mod_xml_curl against
mod_dingaling. When i call xml_curl url in browser i get mod_dingaling
configuration correctly, also when i do reload mod_dingaling it fetches its
configuration from xml_curl correctly. BUT when i try to use dl_login
command to login a jingle profile it does not work. I have tried both
syntax,

Syntax 1:
===
dl_login profile=abcd

Where abcd is a valid jingle profile fetch-able from xml_curl.

Syntax 2:
===
dl_login name=abcd;login=
x...@gmail.com/talk;pass=YYY;dialplan=XML;context=public;rtp-ip=auto;sasl=plain;tls=true;exten=1001

All these values are correct and work if i reload mod_dingaling but they
don't work with dl_login, and give following output.

USAGE: Existing Profile:
dl_login profile=profile_name
Dynamic Profile:
dl_login var1=val1;var2=val2;varN=valN

I don't think xml_curl has any role in this syntax.

Can you please correct me if i am doing something wrong in here or is it a
bug in mod_dingaling.

Thank you.


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
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Re: [Freeswitch-users] memory leak

2009-09-04 Thread Benedikt Fraunhofer

Hello Anthony,

2009/9/2 Anthony Minessale anthony.miness...@gmail.com:

 yes if you have a version that only has log-file you can use that.

 if you find me on irc and send me the credentials privately I will examine
 your box for you.

thanks for that offer, but the box is pretty deep inside our internal
network with no routing to the outside, several stepping-stones in
between and all that security stuff.

I finally found the right amount of load where the memory leak builds
up quickly enough and was able to stop freeswitch before it started
swapping. The result is available on

  http://ns42.ath.cx/B0GdWh/vg-2.log.bz2

(19k)

Thx in advance
  Beni.

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Re: [Freeswitch-users] Proxy authorization

I cannot change the way SIP Authentication works.  The first register  
is always sent without an authorization header then is challenged.  If  
you're getting an instant 403 then you have something wrong in your  
config and the remote system doesn't like it.  Please contact your  
provider and ask them to troubleshoot it with you.


/b

On Sep 4, 2009, at 3:43 AM, Dmitry Bely wrote:



Alas, I cannot change the way the provider's gateway works. It
immediately responses with 403... BTW, it's Mera Damos
(http://www.mera-systems.com ?). No workaround possible?


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Re: [Freeswitch-users] memory leak

2009-09-04 Thread Rupa Schomaker

Worst offenders (leakers over 100K).  The last one is the worst (672M)
-- looks like a lua script.  What are you doing in lua again?

==28624== 105,725 bytes in 1,804 blocks are still reachable in loss
record 497 of 529
==28624==at 0x4022AB8: malloc (vg_replace_malloc.c:207)
==28624==by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so)
==28624==by 0x50384F2: xmlrpc_strdupnull (asprintf.c:92)
==28624==by 0x503F86D: RequestRead (http.c:57)
==28624==by 0x5044413: ??? (server.c:538)
==28624==by 0x5039FAF: ??? (conn.c:37)
==28624==by 0x50486F1: ??? (thread_pthread.c:48)
==28624==by 0x42114FA: start_thread (in
/lib/tls/i686/cmov/libpthread-2.7.so)
==28624==by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so)
==28624==
==28624==
==28624== 116,772 bytes in 3,156 blocks are definitely lost in loss
record 498 of 529
==28624==at 0x4022AB8: malloc (vg_replace_malloc.c:207)
==28624==by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so)
==28624==by 0x40E4EA9: switch_log_vprintf (switch_log.c:438)
==28624==by 0x40E5130: switch_log_printf (switch_log.c:308)
==28624==by 0x4088664: switch_channel_perform_hangup (switch_channel.c:1674)
==28624==by 0x64A12EA: ??? (mod_dptools.c:633)
==28624==by 0x409AA45: switch_core_session_exec (switch_core_session.c:1476)
==28624==by 0x409AF88: switch_core_session_execute_application
(switch_core_session.c:1398)
==28624==by 0x409E674: switch_core_session_run
(switch_core_state_machine.c:166)
==28624==by 0x409A48E: switch_core_session_thread
(switch_core_session.c:1066)
==28624==by 0x4110E05: dummy_worker (thread.c:138)
==28624==by 0x42114FA: start_thread (in
/lib/tls/i686/cmov/libpthread-2.7.so)
==28624==
==28624==
==28624== 119,658 (119,621 direct, 37 indirect) bytes in 3,233 blocks
are definitely lost in loss record 499 of 529
==28624==at 0x4022AB8: malloc (vg_replace_malloc.c:207)
==28624==by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so)
==28624==by 0x40E4EA9: switch_log_vprintf (switch_log.c:438)
==28624==by 0x40E5130: switch_log_printf (switch_log.c:308)
==28624==by 0x4088664: switch_channel_perform_hangup (switch_channel.c:1674)
==28624==by 0x50B6790: sofia_event_callback (sofia.c:3863)
==28624==by 0x5146787: nua_application_event (nua_stack.c:393)
==28624==by 0x519DB28: su_base_port_execute_msgs (su_base_port.c:280)
==28624==by 0x519D8CF: su_base_port_getmsgs (su_base_port.c:202)
==28624==by 0x519DD75: su_base_port_step (su_base_port.c:454)
==28624==by 0x5190968: su_port_step (su_port.h:340)
==28624==by 0x5190938: su_root_step (su_root.c:858)
==28624==
==28624==
==28624== 124,209 bytes in 3,357 blocks are still reachable in loss
record 500 of 529
==28624==at 0x4022AB8: malloc (vg_replace_malloc.c:207)
==28624==by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so)
==28624==by 0x40E4EA9: switch_log_vprintf (switch_log.c:438)
==28624==by 0x40E5130: switch_log_printf (switch_log.c:308)
==28624==by 0x409A5EC: switch_core_session_thread
(switch_core_session.c:1086)
==28624==by 0x4110E05: dummy_worker (thread.c:138)
==28624==by 0x42114FA: start_thread (in
/lib/tls/i686/cmov/libpthread-2.7.so)
==28624==by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so)
==28624==
==28624==
==28624== 124,290 bytes in 4,143 blocks are still reachable in loss
record 501 of 529
==28624==at 0x4022AB8: malloc (vg_replace_malloc.c:207)
==28624==by 0x443B957: vasprintf (in /lib/tls/i686/cmov/libc-2.7.so)
==28624==by 0x5038532: xmlrpc_vasprintf (asprintf.c:61)
==28624==by 0x5038581: xmlrpc_asprintf (asprintf.c:81)
==28624==by 0x503B881: DateToString (date.c:43)
==28624==by 0x5036D09: handler_hook (mod_xml_rpc.c:733)
==28624==by 0x504456F: ??? (server.c:515)
==28624==by 0x5039FAF: ??? (conn.c:37)
==28624==by 0x50486F1: ??? (thread_pthread.c:48)
==28624==by 0x42114FA: start_thread (in
/lib/tls/i686/cmov/libpthread-2.7.so)
==28624==by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so)
==28624==
==28624==
==28624== 137,085 bytes in 3,705 blocks are still reachable in loss
record 502 of 529
==28624==at 0x4022AB8: malloc (vg_replace_malloc.c:207)
==28624==by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so)
==28624==by 0x40E4EA9: switch_log_vprintf (switch_log.c:438)
==28624==by 0x40E5130: switch_log_printf (switch_log.c:308)
==28624==by 0x409921F: switch_core_session_perform_destroy
(switch_core_session.c:947)
==28624==by 0x409A60D: switch_core_session_thread
(switch_core_session.c:1088)
==28624==by 0x4110E05: dummy_worker (thread.c:138)
==28624==by 0x42114FA: start_thread (in
/lib/tls/i686/cmov/libpthread-2.7.so)
==28624==by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so)
==28624==
==28624==
==28624== 145,589 bytes in 1,837 blocks are possibly lost in loss
record 503 of 529
==28624==at 0x4022AB8: malloc (vg_replace_malloc.c:207)
==28624==

Re: [Freeswitch-users] Run a command on event

2009-09-04 Thread Mathieu Parent

Hi

On Thu, Sep 3, 2009 at 6:52 AM, Mathieu Renemrene_li...@avgs.ca wrote:
 See 
 http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Subscribing_to_events


Thanks.

I have tried this method without success and finally replaced the
voicemail section in dialplan by a spidermonkey script with
session.setHangupHook(). Test passed!

Mathieu Parent

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Re: [Freeswitch-users] memory leak

2009-09-04 Thread Benedikt Fraunhofer

2009/9/4 Rupa Schomaker r...@rupa.com:
 Worst offenders (leakers over 100K).  The last one is the worst (672M)
 -- looks like a lua script.  What are you doing in lua again?

i feel kinda dumb to double post, but here it is again :)

the setup is the same as in http://jira.freeswitch.org/browse/MODSOFIA-22

one is

-

local reason = session:getVariable(originate_disposition);

session:setAutoHangup(false);


if(reason) then
   if(reason == NO_ANSWER) then
 -- nothing
   end
   if(reason == USER_BUSY) then
   -- nothing
   end
end

freeswitch.consoleLog(...

--
anotherone is

local sess = nil;
if(argv[1]) then
   sess=argv[1];
end
freeswitch.consoleLog(...
api = freeswitch.API();
local res = api:execute(sched_api ...
freeswitch.consoleLog(...



and the scheduled script does
---
function log(msg)
   freeswitch.consoleLog(notice, c2c-hangup-timeout.lua:  .. msg .. \n);
end


local sess = argv[1];
if(sess)
then
   freeswitch.consoleLog(INFO, hangup-timeout.lua for uuid  ..
sess .. \n);

   api = freeswitch.API();
   local stillValid = api:execute(uuid_getvar, sess .. 
Dummy-DoesChannelExists);
   if(stillValid:sub(1,4) == -ERR)
   then
   log(session uuid  .. sess ..  disappeared (nothing bad));
   else
   -- this is important!!! Otherwise the aleg get's just hung up!
   api:execute(uuid_media, sess);
   api:execute(uuid_transfer, sess ..  -both timeout);
   end
else -- /if(sess)
   log(called with nil session?);
end -- /if(sess)

---

at least there's no fancy db-connection-thingi which could make
debugging harder :)

Cheers
  Beni.

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Re: [Freeswitch-users] memory leak

2009-09-04 Thread Rupa Schomaker

Doesn't that look like a pool that isn't being destroyed?

On Fri, Sep 4, 2009 at 9:10 AM, Rupa Schomakerr...@rupa.com wrote:
 Worst offenders (leakers over 100K).  The last one is the worst (672M)
 -- looks like a lua script.  What are you doing in lua again?
 ==28624== 672,268,288 bytes in 82,064 blocks are still reachable in
 loss record 529 of 529
 ==28624==    at 0x4022AB8: malloc (vg_replace_malloc.c:207)
 ==28624==    by 0x410BB2F: apr_pool_create_ex (apr_pools.c:300)
 ==28624==    by 0x476: apr_thread_create (thread.c:171)
 ==28624==    by 0x4080878: switch_thread_create (switch_apr.c:631)
 ==28624==    by 0x6C278E9: lua_thread (mod_lua.cpp:372)
 ==28624==    by 0x6C27948: ??? (mod_lua.cpp:407)
 ==28624==    by 0x40AADFC: switch_api_execute (switch_loadable_module.c:1567)
 ==28624==    by 0x583A7FC: ??? (mod_commands.c:2426)
 ==28624==    by 0x40A8881: switch_scheduler_execute (switch_scheduler.c:61)
 ==28624==    by 0x40A8DE0: task_thread_loop (switch_scheduler.c:127)
 ==28624==    by 0x40A8EA3: switch_scheduler_task_thread 
 (switch_scheduler.c:168)
 ==28624==    by 0x4110E05: dummy_worker (thread.c:138)
 ==28624==
 ==28624== LEAK SUMMARY:
 ==28624==    definitely lost: 63,113,740 bytes in 1,690,880 blocks.
 ==28624==    indirectly lost: 35,632 bytes in 491 blocks.
 ==28624==      possibly lost: 645,758 bytes in 9,150 blocks.
 ==28624==    still reachable: 681,849,684 bytes in 113,077 blocks.
 ==28624==         suppressed: 0 bytes in 0 blocks.

-- 
-Rupa

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Re: [Freeswitch-users] memory leak

2009-09-04 Thread Benedikt Fraunhofer

personally i would blame xmlrpc (which is no xml :) for it.

Just my 2cent

  Beni.

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Re: [Freeswitch-users] memory leak

2009-09-04 Thread Rupa Schomaker

There are other smaller leakers.  xmlrpc is leaking, but the leaks are
very small compared to the lua leak.  Same with spidermonkey_curl - it
is leaking but not too terribly much.  I'll hop on #freeswitch in a
bit and see if anyone has an idea.

On Fri, Sep 4, 2009 at 9:35 AM, Benedikt
Fraunhoferfraunhofer.lists.freeswitch-...@traced.net wrote:
 personally i would blame xmlrpc (which is no xml :) for it.

 Just my 2cent

  Beni.

 http://www.freeswitch.org




-- 
-Rupa

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Re: [Freeswitch-users] XML Dial Plan vs Language Modules

2009-09-04 Thread Shameem Shiek

Hi Michael,

Why is it not recommended to do the brdge app right in the script?  The
reason I ask this, I did have lot of trouble using Park/Fifo app in the
script and the whole thing started working after I did the UUID transfer and
have the things I wanted executed as part of the Dial plan.

Also, How many concurrent sessions can one support in ESL using Python/Ruby
compared to using Lua?

Thanks.

On Fri, Sep 4, 2009 at 3:06 AM, Michael Collins m...@freeswitch.org wrote:



 On Thu, Sep 3, 2009 at 10:59 PM, Muhammad Shahzad 
 shaherya...@googlemail.com wrote:

 Hi,

 I couple of my team members are working on translating a very long
 Asterisk Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link
 below,


 Before you go through all the trouble of translating the dialplan be sure
 to review the application itself. In many cases just doing a dialplan
 translation results in less efficient use of FreeSWITCH's powerful features.
 Be sure that you are looking at the way FreeSWITCH handles various
 situations and take advantage of its power and ease of use.



 http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables

 The dial plan variables are not getting initialized as expected. I was
 just wondering if we move this variable get and set stuff to any language
 module say mod_perl, will that make any difference performance wise? I mean
 we will be invoking a Perl interpreter for each incoming call, won't that be
 expensive in terms of RAM and CPU usage and thus reducing number of calls
 this FS deployment can handle?

 I have guys with programming skills in Perl, PHP, Python, Java and LUA
 languages. Which language do you recommend for this, again in terms of speed
 and performance?


 Lua is very portable and we've done tests with hundreds of concurrent Lua
 scripts running. The other languages are heavier but they'll still handle
 quite a few concurrent sessions. Just be sure that you don't do the bridge
 app right in the script, use transfer instead and have the dialplan process
 any bridging that you need to do.

 -MC


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Re: [Freeswitch-users] memory leak

2009-09-04 Thread Anthony Minessale

that looks to me like luarun being called on a script that never terminates.
could your script be ending up caught in an endless loop or blocking on
something?


On Fri, Sep 4, 2009 at 9:42 AM, Rupa Schomaker r...@rupa.com wrote:

 There are other smaller leakers.  xmlrpc is leaking, but the leaks are
 very small compared to the lua leak.  Same with spidermonkey_curl - it
 is leaking but not too terribly much.  I'll hop on #freeswitch in a
 bit and see if anyone has an idea.

 On Fri, Sep 4, 2009 at 9:35 AM, Benedikt
 Fraunhoferfraunhofer.lists.freeswitch-...@traced.net wrote:
  personally i would blame xmlrpc (which is no xml :) for it.
 
  Just my 2cent
 
   Beni.

  http://www.freeswitch.org
 



 --
 -Rupa

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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
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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
pstn:213-799-1400
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[Freeswitch-users] UK English wav files

2009-09-04 Thread Brian Stafford

Hi all

anyone know where I can find UK English recordings for the FS prompts 
(assuming there are any)?  (I've googled to no avail).  Alternatively is 
there a list of the text used so we can record our own?

Regards
Brian

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Re: [Freeswitch-users] UK English wav files

2009-09-04 Thread Michael Jerris

We don't currently have a full set of UK English prompts, the prompts  
list (soon to be updated with some new prompts) is available at:

http://svn.freeswitch.org/svn/freeswitch/trunk/docs/phrase/phrase_en.xml

If you are going to get a set professionally recorded, we would be  
happy to host those files and integrate into the build system like we  
did the russian sounds.

Mike



On Sep 4, 2009, at 7:53 AM, Brian Stafford wrote:

 Hi all

 anyone know where I can find UK English recordings for the FS prompts
 (assuming there are any)?  (I've googled to no avail).   
 Alternatively is
 there a list of the text used so we can record our own?

 Regards
 Brian


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[Freeswitch-users] REMINDER: Weekly call is now happening. Join us!

Hello all,

We are now on line and welcoming callers. Here's the agenda so far:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_04

Come join the conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
1-213-799-1400

-MC
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Re: [Freeswitch-users] REMINDER: Weekly call is now happening. Join us!

2009-09-04 Thread Diego Viola

I'm in, very cool =D

Diego

On Fri, Sep 4, 2009 at 4:18 PM, Michael Collins m...@freeswitch.org wrote:

 Hello all,

 We are now on line and welcoming callers. Here's the agenda so far:
 http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_04

 Come join the conference
 sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
 1-213-799-1400

 -MC

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Re: [Freeswitch-users] REMINDER: Weekly call is now happening. Join us!

For the ones SIP challenged:

call Skype the skypeuser skypiax5 and then press 1

-gm

On Fri, Sep 4, 2009 at 6:43 PM, Diego Violadiego.vi...@gmail.com wrote:
 I'm in, very cool =D

 Diego

 On Fri, Sep 4, 2009 at 4:18 PM, Michael Collins m...@freeswitch.org wrote:

 Hello all,

 We are now on line and welcoming callers. Here's the agenda so far:
 http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_04

 Come join the conference
 sip:8...@conference.freeswitch.org
 1-213-799-1400

 -MC

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[Freeswitch-users] SIP provider directory?

2009-09-04 Thread Lon Baker

Does anyone know of a SIP provider or network directory? A list of all
the public service provider or networks? Gizmo, Google Voice, etc? Or
Vitelity, iCall?

Lon Baker
Kickass Pixels
-
+1-415-894-0184
-
http://kickasspixels.com
http://twitter.com/kickasspixels
http://www.linkedin.com/in/lonbaker
--

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[Freeswitch-users] Minimum/Recommended Freeswitch System Configuration

2009-09-04 Thread Jerry Richards


Under the Minimum/Recommended System Requirements, what is meant by We
recommend you plan for 50% duty cycle?  What is this duty cycle?

Also, I see that the system requirements indicate Freeswitch recommends 1GB
RAM and 50MB disk space.  I guess I'm wondering how the number of extensions
and external interfaces drive size of RAM and disk space?  For example,
would these recommendations support 100 extensions and one external
interface?  500 extensions and 10 external interfaces?  Etc.?

Best Regards,
Jerry



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Re: [Freeswitch-users] Proxy authorization

On Fri, Sep 4, 2009 at 4:45 PM, Brian Westbr...@freeswitch.org wrote:
 I cannot change the way SIP Authentication works.  The first register is
 always sent without an authorization header then is challenged.  If you're
 getting an instant 403 then you have something wrong in your config and the
 remote system doesn't like it.  Please contact your provider and ask them to
 troubleshoot it with you.
 /b

Well, you are right. Looks like the problem is not with authorization
but in the line

  Contact: sip:gw+1.2@5.6.7.8:5080;transport=udp

that the gateway would like to see as

  Contact: sip:usern...@1.2.3.4

I've found (almost undocumented) parameter extension-in-contact, but
it still gives

  Contact: sip:usern...@5.6.7.8:5080;transport=udp

(1.2.3.4 is my IP address, 5.6.7.8 is gateway's one). Any idea how to
overcome this?

 On Sep 4, 2009, at 3:43 AM, Dmitry Bely wrote:

 Alas, I cannot change the way the provider's gateway works. It
 immediately responses with 403... BTW, it's Mera Damos
 (http://www.mera-systems.com ?). No workaround possible?

- Dmitry Bely

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Re: [Freeswitch-users] Proxy authorization

Try filling out contact-host too.  But if the far end gets pissed  
about your contact they are broken.


/b

On Sep 4, 2009, at 2:22 PM, Dmitry Bely wrote:


Well, you are right. Looks like the problem is not with authorization
but in the line

 Contact: sip:gw+1.2@5.6.7.8:5080;transport=udp

that the gateway would like to see as

 Contact: sip:usern...@1.2.3.4

I've found (almost undocumented) parameter extension-in-contact, but
it still gives

 Contact: sip:usern...@5.6.7.8:5080;transport=udp


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Re: [Freeswitch-users] Proxy authorization

I'm started to suspect another thing.. Successful register (SIP phone) contains

REGISTER sip:Domain SIP/2.0

while unsuccessful one is

REGISTER sip:1.2.3.4 SIP/2.0

What parameter is responsible for Request-URI? Note that I need both
IP address for proxy and symbolic name for SIP domain  (which is not
mapped the resolvable DNS name).

On Fri, Sep 4, 2009 at 11:37 PM, Brian Westbr...@freeswitch.org wrote:
 Try filling out contact-host too.  But if the far end gets pissed about your
 contact they are broken.
 /b
 On Sep 4, 2009, at 2:22 PM, Dmitry Bely wrote:

 Well, you are right. Looks like the problem is not with authorization
 but in the line

  Contact: sip:gw+1.2@5.6.7.8:5080;transport=udp

 that the gateway would like to see as

  Contact: sip:usern...@1.2.3.4

 I've found (almost undocumented) parameter extension-in-contact, but
 it still gives

  Contact: sip:usern...@5.6.7.8:5080;transport=udp

- Dmitry Bely

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Re: [Freeswitch-users] Proxy authorization


show me your XML for the gateway please.

/b

On Sep 4, 2009, at 3:43 PM, Dmitry Bely wrote:

I'm started to suspect another thing.. Successful register (SIP  
phone) contains


REGISTER sip:Domain SIP/2.0

while unsuccessful one is

REGISTER sip:1.2.3.4 SIP/2.0

What parameter is responsible for Request-URI? Note that I need both
IP address for proxy and symbolic name for SIP domain  (which is not
mapped the resolvable DNS name).


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Re: [Freeswitch-users] Minimum/Recommended Freeswitch System Configuration

2009-09-04 Thread Mitul Limbani

Jerry,

As far as I understand freeswitch, it using kernel to thread and this  
operation eats good amount of RAM, but since the internal strructure  
of fs is to store all these sip details in runtime sqlite db, which is  
compressed text data earlier written in XML but while fs loads this  
configs it gets it in sqlite and that's what it used instead of  
asterisks astdb.

Although what you see as recommended config for 500 users is true but  
it also depends on which processor you are trying this on. Intel or  
AMD is still ok but if you trying it on arm I don't have any data as  
such, interestingly if you have some test hardware scenario you can  
actually test and let us all know about it, it's quite useful bit of  
info that can be positioned on the FS Wiki, in case you want to take  
this experiment offlist do write to me, im interested to document :)

Look forward to hear from you,

Thanks  Regards,
Mitul Limbani,
Founder  CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

On 05-Sep-2009, at 12:03 AM, Jerry Richards jerry.richa...@teotech.com 
  wrote:


 Under the Minimum/Recommended System Requirements, what is meant by  
 We
 recommend you plan for 50% duty cycle?  What is this duty cycle?

 Also, I see that the system requirements indicate Freeswitch  
 recommends 1GB
 RAM and 50MB disk space.  I guess I'm wondering how the number of  
 extensions
 and external interfaces drive size of RAM and disk space?  For  
 example,
 would these recommendations support 100 extensions and one external
 interface?  500 extensions and 10 external interfaces?  Etc.?

 Best Regards,
 Jerry



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Re: [Freeswitch-users] Proxy authorization

On Sat, Sep 5, 2009 at 1:08 AM, Brian Westbr...@freeswitch.org wrote:
 show me your XML for the gateway please.
 /b

It's fairly standard:

!--
Shell provider account should work with most providers.

--
include
  user id=$${default_provider}
gateways
  gateway name=$${default_provider}
param name=username value=$${default_provider_username}/
param name=password value=$${default_provider_password}/
param name=from-user value=$${default_provider_username}/
param name=from-domain value=$${default_provider_from_domain}/
param name=realm value=$${default_provider_from_domain}/
param name=proxy value=$${default_provider}/
param name=expire-seconds value=600/
param name=register value=$${default_provider_register}/
param name=retry-seconds value=30/
param name=extension value=$${default_provider_contact}/
!--param name=contact-params value=domain_name=$${domain}/--
param name=context value=public/
  /gateway
/gateways
params
  param name=password value=$${default_provider_password}/
/params
  /user
/include

default_provider_register is set to true.

In the meantime I looked into the sources. If I understand them right,
proxy address is always used in REGISTER header:

sofia.c, line 1471
gateway-register_url =
switch_core_sprintf(gateway-pool, sip:%s, proxy);

Probably it's incorrect as RFC 3261 says:

Request-URI: The Request-URI names the domain of the location
   service for which the registration is meant (for example,
   sip:chicago.com).  The userinfo and @ components of the
   SIP URI MUST NOT be present.

So the domain name (from-domain?) should be used there, not the proxy address.

 On Sep 4, 2009, at 3:43 PM, Dmitry Bely wrote:

 I'm started to suspect another thing.. Successful register (SIP phone)
 contains

 REGISTER sip:Domain SIP/2.0

 while unsuccessful one is

 REGISTER sip:1.2.3.4 SIP/2.0

 What parameter is responsible for Request-URI? Note that I need both
 IP address for proxy and symbolic name for SIP domain  (which is not
 mapped the resolvable DNS name).

- Dmitry Bely

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Re: [Freeswitch-users] Proxy authorization

Can you send it to me with the data filled out off list please.

/b

On Sep 4, 2009, at 4:33 PM, Dmitry Bely wrote:

 It's fairly standard:

 !--
Shell provider account should work with most providers.

 --


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[Freeswitch-users] Does FreeSWITCH wiki have update notify?

2009-09-04 Thread Mitul Limbani

Hello,

It may sound a bit stupid but still wanna ask out here, if there is  
any way to replicate FreeSWITCH wiki mirror for local reference or  
mainataining local copy instead of Reading online which costs a lot in  
developing countries like that of ours and Asia/Africa in general.

We tried httrack but that brings in everything back here as static  
HTML so we can't really search coz every search the static content  
takes us back online on wiki.

Any suggestions in this space would be really helpful, in past I have  
asked if one can mirror FS wiki but so far it won't work unless FS  
server allows rsync request as how it works with mirroring php.net  
with all comments etc.

I look forward for more suggestions on the same,

Thanks  Regards,
Mitul Limbani,
Founder  CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

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Re: [Freeswitch-users] Does FreeSWITCH wiki have update notify?

I will look into the MediaWiki docs to see what's available. In the meantime
you will probable need to use the recent changes link on the navigation
bar.
-MC

On Fri, Sep 4, 2009 at 2:39 PM, Mitul Limbani mi...@enterux.com wrote:

 Hello,

 It may sound a bit stupid but still wanna ask out here, if there is
 any way to replicate FreeSWITCH wiki mirror for local reference or
 mainataining local copy instead of Reading online which costs a lot in
 developing countries like that of ours and Asia/Africa in general.

 We tried httrack but that brings in everything back here as static
 HTML so we can't really search coz every search the static content
 takes us back online on wiki.

 Any suggestions in this space would be really helpful, in past I have
 asked if one can mirror FS wiki but so far it won't work unless FS
 server allows rsync request as how it works with mirroring php.net
 with all comments etc.

 I look forward for more suggestions on the same,

 Thanks  Regards,
 Mitul Limbani,
 Founder  CEO,
 Enterux Solutions Pvt. Ltd.,
 The Enterprise Linux Company (r),
 http://www.enterux.com
 http://www.entVoice.com

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[Freeswitch-users] Monitoring FreeSWITCH Events via Telnet socket (Net::Telnet)

2009-09-04 Thread Tina Martinez

Hello,

I have a question, but IÃ¢â¬â¢m not certain whether this is a FreeSWITCH 
issue, or
something specific to Perl.

I have setup a Perl application (Ã¢â¬ÅlistenerÃ¢â¬Â) that monitors the 
events of my
FreeSWITCH server via a Telnet socket.  So far, the application seems to work
very nicely, except that the listener does not consistently capture all of the
events that are streaming through the socket.  I have been able to work around
most of the issues, but one of the more significant pain-points is when a new
member is getting added to a conference (Event-Name: CUSTOM, Action: 
add-member).

The log file that I generate from the Telnet socket shows the event details, so 
I
know that the data is coming across the pipe, but I donÃ¢â¬â¢t consistently 
see the
details in my event listener code.  Also, the fact that I sometimes do see all 
of
the details confuses me (I don't see a pattern to give any clues towards the 
cause).

BTW - I do have the action application=verbose_events data=true/ in the
dialplan.

So, I wonder, is there something else that I can do in FreeSWITCH to increase 
the
reliability of capturing the event details? or is there something I should be
doing in Perl to somehow buffer the data (i.e., why would I see the details in
the socket log file, but not in the data stream within the code?).  I'm not a
socket wizard by any stretch, so I'm hoping that it might be a simple issue
related to the Net::Telnet implementation.  

Any suggestions would be greatly appreciated.

Thank you in advance,


- Tina


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Re: [Freeswitch-users] Monitoring FreeSWITCH Events via Telnet socket (Net::Telnet)

2009-09-04 Thread Anthony Minessale

you should use ESL lib and the supplied perl mod

from FS build root

cd libs/esl
make
make perlmod
cd perl

copy ESL.pm and ESL.so into your INC path
see the examples in that same folder.


On Fri, Sep 4, 2009 at 5:40 PM, Tina Martinez t...@a2unlimited.com wrote:

 Hello,

 I have a question, but IÃ¢â‚¬â„¢m not certain whether this is a FreeSWITCH
 issue, or
 something specific to Perl.

 I have setup a Perl application (Ã¢â‚¬Å“listenerÃ¢â‚¬Â ) that monitors the
 events of my
 FreeSWITCH server via a Telnet socket.  So far, the application seems to
 work
 very nicely, except that the listener does not consistently capture all of
 the
 events that are streaming through the socket.  I have been able to work
 around
 most of the issues, but one of the more significant pain-points is when a
 new
 member is getting added to a conference (Event-Name: CUSTOM, Action:
 add-member).

 The log file that I generate from the Telnet socket shows the event
 details, so I
 know that the data is coming across the pipe, but I donÃ¢â‚¬â„¢t
 consistently see the
 details in my event listener code.  Also, the fact that I sometimes do see
 all of
 the details confuses me (I don't see a pattern to give any clues towards
 the cause).

 BTW - I do have the action application=verbose_events data=true/ in
 the
 dialplan.

 So, I wonder, is there something else that I can do in FreeSWITCH to
 increase the
 reliability of capturing the event details? or is there something I should
 be
 doing in Perl to somehow buffer the data (i.e., why would I see the details
 in
 the socket log file, but not in the data stream within the code?).  I'm not
 a
 socket wizard by any stretch, so I'm hoping that it might be a simple issue
 related to the Net::Telnet implementation.

 Any suggestions would be greatly appreciated.

 Thank you in advance,


 - Tina


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Twitter: http://twitter.com/FreeSWITCH_wire

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IRC: irc.freenode.net #freeswitch

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Re: [Freeswitch-users] Does FreeSWITCH wiki have update notify?

2009-09-04 Thread Mitul Limbani


Mike,

I m not sure if we can program httrack to pickup changes automatically  
(I.e. Looking at recent changes) so this brings us back to square one,  
instead we can setup media wiki here and do rsync with fs wiki box.


Thanks  Regards,
Mitul Limbani,
Founder  CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

On 05-Sep-2009, at 3:33 AM, Michael Collins m...@freeswitch.org wrote:

I will look into the MediaWiki docs to see what's available. In the  
meantime you will probable need to use the recent changes link on  
the navigation bar.

-MC

On Fri, Sep 4, 2009 at 2:39 PM, Mitul Limbani mi...@enterux.com  
wrote:

Hello,

It may sound a bit stupid but still wanna ask out here, if there is
any way to replicate FreeSWITCH wiki mirror for local reference or
mainataining local copy instead of Reading online which costs a lot in
developing countries like that of ours and Asia/Africa in general.

We tried httrack but that brings in everything back here as static
HTML so we can't really search coz every search the static content
takes us back online on wiki.

Any suggestions in this space would be really helpful, in past I have
asked if one can mirror FS wiki but so far it won't work unless FS
server allows rsync request as how it works with mirroring php.net
with all comments etc.

I look forward for more suggestions on the same,

Thanks  Regards,
Mitul Limbani,
Founder  CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

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[Freeswitch-users] Mod_nibblebill for CDR billing

2009-09-04 Thread Rogelio Perez

 From the mod_nibblebill documentation:

At the end of a call, the module sets a variable named  
nibble_total_billed. You can use mod_cdr to record this variable to  
your CDR log.

Is it possible to do the same with mod_xml_cdr?
I'm looking for a simple way of billing my CDRs and this one looks  
like a good solution.
Has anyone tried doing anything similar?

Thanks,
Rogelio 

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Re: [Freeswitch-users] Mod_nibblebill for CDR billing

All the variables are there in XML_CDR too.

/b

On Sep 4, 2009, at 6:28 PM, Rogelio Perez wrote:

 From the mod_nibblebill documentation:

 At the end of a call, the module sets a variable named
 nibble_total_billed. You can use mod_cdr to record this variable to
 your CDR log.

 Is it possible to do the same with mod_xml_cdr?
 I'm looking for a simple way of billing my CDRs and this one looks
 like a good solution.
 Has anyone tried doing anything similar?

 Thanks,
 Rogelio


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[Freeswitch-users] 482 Request merged, in serial forking

2009-09-04 Thread Humberto Quintana


Hello,

I'm a new Freeswitch user. After some reading I put Freeswitch (Version 1.0.4) 
to work as Session Border Controller.  I have only one problem that I dont know 
how to solve it ( or which parameter to set) and I'd appreciate if someone 
could give me a clue about this. 

Kamailio is sitting behind FS and it selects the route or routes in case of 
failure (serial forking) .  Freeswitch is configured to use directly the 
Request-URI sent by Kamailio.

So, when the 1st route fails, Kamailio receives the Reply from FS and sends 
back the ACK to end the transaction.  More than 1 second later, a new INVITE 
from Kamailio with the next route is tried (With the To-header's tag is empty. 
Same Callid, From and Cseq header but different VIA-header's branch parameter) 
and FS is answering back 482 Merged Request.  It happens the same for the 3rd 
route.

It seems that the transaction is still 'alive' in FS even if the ACK was 
received ?


Thanks,

Humberto


===1st route===

U 2009/09/03 17:20:36.069147 kamailio - freeswitch
INVITE sip:514...@gw1 SIP/2.0.
Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.0
Call-ID: 1

U 2009/09/03 17:20:36.169147 freeswitch - gw1
INVITE sip:514...@gw1 SIP/2.0.
Call-ID: 2

U 2009/09/03 17:20:36.170158 gw1 - freeswitch
SIP/2.0 100 Trying.
Call-ID: 2

U 2009/09/03 17:20:36.190457 gw1 - freeswitch
SIP/2.0 503 Service Unavailable.
Call-ID: 2

U 2009/09/03 17:20:36.193296 freeswitch - gw1
ACK sip:5142776...@gw1 SIP/2.0.
Call-ID: 2

U 2009/09/03 17:20:36.227492 freeswitch - kamailio
SIP/2.0 503 Service Unavailable.
Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.0
Call-ID: 1

U 2009/09/03 17:20:36.228122 kamailio - freeswitch
ACK sip:514...@gw1 SIP/2.0.
Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.0
Call-ID: 1


===2nd route===
U 2009/09/03 17:20:37.596885 kamailio - freeswitch
INVITE sip:1514...@gw2:5061 SIP/2.0
Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.1
Call-ID: 1

U 2009/09/03 17:20:37.597590 freeswitch - kamailio
SIP/2.0 482 Request merged.
Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.1
Call-ID: 1

U 2009/09/03 17:20:37.598163 kamailio - freeswitch
ACK sip:1514...@gw2:5061 SIP/2.0.
Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.1
Call-ID: 1


===3rd route===
U 2009/09/03 17:20:37.642098 kamailio - freeswitch
INVITE sip:514...@gw3 SIP/2.0
Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.2
Call-ID: 1

U 2009/09/03 17:20:37.642634 freeswitch - kamailio
SIP/2.0 482 Request merged.
Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.2
Call-ID: 1

U 2009/09/03 17:20:37.643139 kamailio - freeswitch
ACK sip:514...@gw3 SIP/2.0.
Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.2
Call-ID: 1


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[Freeswitch-users] New install

2009-09-04 Thread Ujjval Karihaloo


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.
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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.


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Re: [Freeswitch-users] 482 Request merged, in serial forking