Re: [Freeswitch-users] opal build error

2009-08-16 Thread Peter Olsson
Make sure to do a complete rebuild. And also read the comments in jira 
MODOPAL-10.

/Peter


On 09-08-16 03.51, Seven Du dujinf...@gmail.com wrote:

Hi,

According to wiki it still in development status, but should compile
right? Any idea about this? thanks.

make

In file included from mod_opal.cpp:25:
mod_opal.h:151: error: conflicting return type specified for 'virtual
OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)'
/usr/include/opal/opal/localep.h:267: error:   overriding 'virtual
ptlib_virtual_function_changed_or_removed**
OpalLocalEndPoint::CreateConnection(OpalCall, void*)'
mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall,
FSEndPoint, switch_caller_profile_t*, switch_core_session_t*,
switch_channel_t*)':
mod_opal.cpp:564: error: no matching function for call to
'OpalLocalConnection::OpalLocalConnection(OpalCall, FSEndPoint, NULL)'
/usr/include/opal/opal/localep.h:290: note: candidates are:
OpalLocalConnection::OpalLocalConnection(OpalCall,
OpalLocalEndPoint, void*, unsigned int,
OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h:
276: note:
OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection)
mod_opal.cpp: In member function 'switch_status_t
FSConnection::receive_message(switch_core_session_message_t*)':
mod_opal.cpp:1037: error: 'SWITCH_CHANNEL_SESSION_LOG' was not
declared in this scope
make[1]: *** [mod_opal.lo] Error 1
make: *** [all] Error 1
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Re: [Freeswitch-users] opal build error

2009-08-16 Thread Seven Du
Thanks, will try later.

On Aug 16, 2009, at 2:20 PM, Peter Olsson wrote:
 Make sure to do a complete rebuild. And also read the comments in  
 jira MODOPAL-10.

 /Peter


 On 09-08-16 03.51, Seven Du dujinf...@gmail.com wrote:

 Hi,

 According to wiki it still in development status, but should compile
 right? Any idea about this? thanks.

 make

 In file included from mod_opal.cpp:25:
 mod_opal.h:151: error: conflicting return type specified for 'virtual
 OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)'
 /usr/include/opal/opal/localep.h:267: error:   overriding 'virtual
 ptlib_virtual_function_changed_or_removed**
 OpalLocalEndPoint::CreateConnection(OpalCall, void*)'
 mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall,
 FSEndPoint, switch_caller_profile_t*, switch_core_session_t*,
 switch_channel_t*)':
 mod_opal.cpp:564: error: no matching function for call to
 'OpalLocalConnection::OpalLocalConnection(OpalCall, FSEndPoint,  
 NULL)'
 /usr/include/opal/opal/localep.h:290: note: candidates are:
 OpalLocalConnection::OpalLocalConnection(OpalCall,
 OpalLocalEndPoint, void*, unsigned int,
 OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h:
 276: note:
 OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection)
 mod_opal.cpp: In member function 'switch_status_t
 FSConnection::receive_message(switch_core_session_message_t*)':
 mod_opal.cpp:1037: error: 'SWITCH_CHANNEL_SESSION_LOG' was not
 declared in this scope
 make[1]: *** [mod_opal.lo] Error 1
 make: *** [all] Error 1
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Re: [Freeswitch-users] Audio only in one direction when calling FS from Skype

2009-08-16 Thread Scott Torr

I have used a MS-Skype client on the local area network to receive Skype
calls and do not have any audio continuity problems, and would conclude
that there is not an issue with the firewall if this is working
correctly.

To the best of my knowledge the external ip addresses and local networks
are configured properly in FS.


When I use the (linux Skpye client)=(mod_skypiax)=(FS) to recieve
Skype calls however I cannot hear the incoming audio.

This problem only occurs with an incoming Skype call (ie cannot hear
their audio, they can hear mine).

If I make an outgoing call from FS to Skype there are no audio problem.

So this is an asymmetrical problem.

I have used wireshark and can see UDP packets going in both directions
on the local area network.
So the incoming audio data is hitting the (linux Skype client).


How can I investigate where the data is being dropped?

Is it between:

(linux Skype client)--(mod_skypiax)
or
(mod_skypiax)--(FS)
 

I need some advice on what debug commands to use to see where the audio
is being dropped or some clues as to how to pinpoint where the problem
is occurring.


For example is (mod_skypiax) getting the audio data from (linux Skype
client)?

How do investigate that?
How do I see that?
Are there channel IO byte counters?

The machine is using snd_dummy, has no vmware sound card.

It is strange the problem is only in one direction?



Thanks,
Scott Torr



On Sat, 15 Aug 2009 17:20 -0400, Michael Jerris m...@jerris.com
wrote:
 Make sure your external ip addresses and local networks are configured  
 properly.
 
 Mike
 
 On Aug 15, 2009, at 1:31 PM, Scott Torr wrote:
 
  Freeswitch-users,
 
  I'm very new to Freeswitch and have installed the following in VMware
  Server 2.0.1
 
 
  ubuntu-8.04.3-server-i386.iso (udate/upgrade)
  skype-debian_2.0.0.72-1_i386.deb
  FreeSWITCH Version 1.0.trunk (14492)
  mod_skypiax
 
 
  Using the FS wiki to install/setup and making small changes to default
  XML configs as required.
 
 
  (phone)--(SIP ATA)--(FS)--(MS-Skype client | Skype 'online number' via
  PSTN)
 
 
  The MS-Skype client is on the same local network as FS.
 
 
  I can make calls from the phone to the Skype Client and this works OK.
  Audio path OK: phone--FS--Skype
 
 
  But ,if I call from the MS-Skype client to the phone I hear no audio
  from the MS-Skype client.
 
  Audio path: phone--FS--Skype
 
  Likewise a call from a Skype 'online number' can hear for example the
  default 5000 ivr but the DTMF tones from the PSTN phone are not  
  detected
  by FS when action application=start_dtmf / is used.
 
 
 
  Any suggestions or pointers in the right direction would be much
  appreciated.
 
  Most likely I have overlook something very obvious to others.
 
 
  Thanks in advance,
  Scott Torr
 
 
 
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Re: [Freeswitch-users] mod nibblebill question

2009-08-16 Thread Manu
Thank you for the reply Michael. Let me ask your opinion on another related
matter also.

I am using a MS SQL Database (which will be there on a remote server). I
wish to keep ...

1- User database in SQL.
2- Dialplans in SQL.
3- CDR logged in SQL.
4- I also require to cut the call in real time when credit is over.

I want to deply this to get good performance for 500 calls simultaneous. I
see in documents that using http responses i can fetch data from my web
server. I figure i can return number of gateways and other dial plan
parameters this way. In the same HTTP request i can also return call Rates
for the called destination (which i can use in nibble)

Is there any other efficiant apprroach you can suggest?

Regards,

Manoj





On Sun, Aug 16, 2009 at 2:50 AM, Michael Jerris m...@jerris.com wrote:

 increments are in seconds, not microseconds.  In IMS for example I
 think it defaults to 20 or 30 second nibbles, depending on your
 tolerances and billing increments something much larger may even make
 sense.  Doing billing in sub second increments doesn't make a lot of
 sense to me.  Remember that this is just keeping track of available
 credit so if there are multiple calls at the same time you won't go
 over balance.  Everything is still reconciled at hang up, so if you
 have a bit too much reserved from your nibble the worst that could
 happen is it could cut off calls a little too early when multiple
 calls are in progress on the same account.

 Mike

 On Aug 15, 2009, at 3:33 PM, Manu wrote:

  Hello,
 
  If we use heartbeat option on in nibblebill.conf.xml does that mean
  ODBC database table will be updated every microsecond or any other
  interval we set?
 
  If this is so and there are many users (Lets say 500 users) are
  connected to FS wouldn't it create locking issues in DB?
 
  Regards,
 
  Manoj

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Re: [Freeswitch-users] mod nibblebill question

2009-08-16 Thread Michael Giagnocavo
Just an in-general performance hint: Get your setup running, then use the SQL 
Profiler to generate a tuning trace, and run the Database Engine Tuning 
Advisor with it. That way it'll tell you which indexes you need (or need to 
drop) to optimize the database side of performance.

500 simultaneous calls isn't much though. Even with 30 second nibblebillings, 
that's only 16 updates/sec - and your calls/sec is probably lower than that. 
That's not too much work for any average server these days, even doing 
dialplan/cdr/billing against remote SQL.

-Michael

From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Manu
Sent: Sunday, August 16, 2009 2:05 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod nibblebill question

Thank you for the reply Michael. Let me ask your opinion on another related 
matter also.

I am using a MS SQL Database (which will be there on a remote server). I wish 
to keep ...

1- User database in SQL.
2- Dialplans in SQL.
3- CDR logged in SQL.
4- I also require to cut the call in real time when credit is over.

I want to deply this to get good performance for 500 calls simultaneous. I see 
in documents that using http responses i can fetch data from my web server. I 
figure i can return number of gateways and other dial plan parameters this way. 
In the same HTTP request i can also return call Rates for the called 
destination (which i can use in nibble)

Is there any other efficiant apprroach you can suggest?

Regards,

Manoj





On Sun, Aug 16, 2009 at 2:50 AM, Michael Jerris 
m...@jerris.commailto:m...@jerris.com wrote:
increments are in seconds, not microseconds.  In IMS for example I
think it defaults to 20 or 30 second nibbles, depending on your
tolerances and billing increments something much larger may even make
sense.  Doing billing in sub second increments doesn't make a lot of
sense to me.  Remember that this is just keeping track of available
credit so if there are multiple calls at the same time you won't go
over balance.  Everything is still reconciled at hang up, so if you
have a bit too much reserved from your nibble the worst that could
happen is it could cut off calls a little too early when multiple
calls are in progress on the same account.

Mike

On Aug 15, 2009, at 3:33 PM, Manu wrote:

 Hello,

 If we use heartbeat option on in nibblebill.conf.xml does that mean
 ODBC database table will be updated every microsecond or any other
 interval we set?

 If this is so and there are many users (Lets say 500 users) are
 connected to FS wouldn't it create locking issues in DB?

 Regards,

 Manoj
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[Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

2009-08-16 Thread Raffaele P. Guidi
I had the sweet surprise to find the installer packaged with FreePBX...
really great! Why it has not been advertised as it deserves? It worked like
a breeze once launched, with the automatic configuration and all of that.,
Only thing: once stopped I cannot get it to load sofia profiles anymore -
issueing sofia status doesn't show anything. I had to copy internal.xml and
default.xml from a previous installation and everything got to work again -
but no FreePBX anymore :( I'm sure I'm missing something important.
Can you give me a hint? Should sofia profiles be served by curl or
something?

Thanks,
   Raffaele
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Re: [Freeswitch-users] Transporting SIP over TCP

2009-08-16 Thread Tzury Bar Yochay
Hi Brian,

Just for your information here is a mail I got from a colleague of
mine which I consider as an experienced freeswitch integrator

 ... TCP used to work, and I had about 20-30 client phones connecting with it.
 About 3 months ago I did an upgrade and TCP no longer worked.  I was in panic 
 mode, so I just ended up walking my
 clients through reconfiguring their phones to use UDP.

 It is definitely a bug in Freeswitch.  I can help you work up a jira bug 
 report if you like.
 The bug happened somewhere between svn version 9800ish and 13223.
 13223 is the version I'm running on my systems now, older, but stable for 
 me...

It would be just great if you can confirm the TCP functionality in the
latest release of FreeSwitch


/t

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[Freeswitch-users] how to set different action for different cause code

2009-08-16 Thread Woody Dickson
Hello,

I find hangup_hook, but I would like to define different actions for
different hangup codes.  Is there anyway to do that?

Woody
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Re: [Freeswitch-users] Transporting SIP over TCP

2009-08-16 Thread bakko
I'm just tryng TCP now and work (latest trunk)

I done this test with eyebeam phone registeres to TCP port and call other ip 
phone (registered over UDB port).

The audio work fine in both directions.

My configuration:

FS -- internet -- NAT -- Phones

In the internal profile:

Call-ID:OTIyNzRlNTQxNTk5NzBhYzMzYzM1YmQyNTgzNTA3MTk.
User:   1...@nydomain.com
Contact:User 
sip:1...@87.12.101.59:13035;transport=TCP;rinstance=f3fd252ea20dd34a
Agent:  eyeBeam release 1102u stamp 52345
Status: Registered(TCP)(unknown) EXP(2009-08-16 08:59:38)
Host:   mydomain.com
IP: XXX:XXX.XXX.XXX
Port:   13035
Auth-User:  1004
Auth-Realm: mydomain.com




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Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

2009-08-16 Thread Carlos Talbot
When you configure FreePBX for the first time it wipes out the sip_profiles
directory. If you follow the FreePBX shortcut on your desktop it'll mention
this on the last screen of the configuration. You might see something such
as the following below. If you plan to use FreePBX you'll need to define
trunk groups, trunks, etc in order to have the sip_profiles directory
populated.
regards,

Carlos


Incompatible ConfigurationWARNING: THE FOLLOWING FILES WILL BE DELETED!

   - D:/FreeSWITCH/conf/sip_profiles/external.xml
   - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
   - D:/FreeSWITCH/conf/sip_profiles/internal.xml


On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi 
raffaele.p.gu...@gmail.com wrote:

 I had the sweet surprise to find the installer packaged with FreePBX...
 really great! Why it has not been advertised as it deserves? It worked like
 a breeze once launched, with the automatic configuration and all of that.,
 Only thing: once stopped I cannot get it to load sofia profiles anymore -
 issueing sofia status doesn't show anything. I had to copy internal.xml and
 default.xml from a previous installation and everything got to work again -
 but no FreePBX anymore :( I'm sure I'm missing something important.
 Can you give me a hint? Should sofia profiles be served by curl or
 something?

 Thanks,
Raffaele

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Re: [Freeswitch-users] Transporting SIP over TCP

2009-08-16 Thread Brian West
Works fine here.

/b

On Aug 16, 2009, at 6:25 AM, Tzury Bar Yochay wrote:

 Hi Brian,

 Just for your information here is a mail I got from a colleague of
 mine which I consider as an experienced freeswitch integrator

 ... TCP used to work, and I had about 20-30 client phones  
 connecting with it.
 About 3 months ago I did an upgrade and TCP no longer worked.  I  
 was in panic mode, so I just ended up walking my
 clients through reconfiguring their phones to use UDP.

 It is definitely a bug in Freeswitch.  I can help you work up a  
 jira bug report if you like.
 The bug happened somewhere between svn version 9800ish and 13223.
 13223 is the version I'm running on my systems now, older, but  
 stable for me...

 It would be just great if you can confirm the TCP functionality in the
 latest release of FreeSwitch


 /t

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Re: [Freeswitch-users] BLF and Openzap

2009-08-16 Thread Brian West
Yes you sure can... setting the presence_id to ex...@domain and have  
the phone sub to that you'll have the light come on when its in use.

/b

On Aug 15, 2009, at 6:29 PM, Terry Moore-Read wrote:

 Is it possible to have a sip phone show blf status for a phone which
 is connected to an openzap port ?


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Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

2009-08-16 Thread Brian West
chances are mod_sofia isn't loaded.

/b

On Aug 16, 2009, at 4:43 AM, Raffaele P. Guidi wrote:

 Can you give me a hint? Should sofia profiles be served by curl or  
 something?

 Thanks,
Raffaele


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Re: [Freeswitch-users] ClueCon2009 Torrents

2009-08-16 Thread Even André Fiskvik

Can we get the torrents added to a mail on this list soon please?

Best regards,
Even André

On 16. aug.. 2009, at 05.28, Mitul Limbani wrote:


I would be glad to offer mirror service to Cluecon 2009 videos :)

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

On 16-Aug-2009, at 3:17 AM, Diego Viola diego.vi...@gmail.com wrote:


Upload the torrent files in http://files.freeswitch.org ;)

On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks jaybi...@gmail.com  
wrote:

I'd also seed such a torrent.

Please send the link :)



On 16/08/2009, at 6:34, João Mesquita jmesqu...@gmail.com wrote:

 I am interested and would also seed to the community

 On 8/15/09, Gabriel Gunderson g...@gundy.org wrote:
 On Sat, Aug 15, 2009 at 12:13 PM, Pederpe...@networkoblivion.com
 wrote:
 If you want the torrents, email me off list.

 Why off list?  Isn't the point of torrents to have more people
 sharing
 in the load?

 Gabe

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 --
 Sent from my mobile device

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[Freeswitch-users] ext sip rtp IP trouble

2009-08-16 Thread kokoska rokoska

Hi all,

I have just upgarded my (a little bit old :-) FreeSWITCH to current
trunk and now i have troubles with IP addresses in SIP  SDP, while
before upgrade everything work fine...

My FreeSWITCH is behind NAT and all relevant sip profiles have set:

param name=ext-rtp-ip value=1.2.3.4/
param name=ext-sip-ip value=1.2.3.4/

But when I send INVITE thru the profile to the gateway associated with
it i see local IP in Via header and in SDP m line too.
BTW: The Contact header is populated with external IP (1.2.3.4)

Could you, please, point me how could I modify IP in Via and in SDP?
Thank you.

Best regards,

kokoska.rokoska


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Re: [Freeswitch-users] ext sip rtp IP trouble

2009-08-16 Thread Brian West
You have to set the local-network-acl on the profile so it knows when  
or not to fix up the VIA
/b

On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:


 Hi all,

 I have just upgarded my (a little bit old :-) FreeSWITCH to current
 trunk and now i have troubles with IP addresses in SIP  SDP, while
 before upgrade everything work fine...

 My FreeSWITCH is behind NAT and all relevant sip profiles have set:

 param name=ext-rtp-ip value=1.2.3.4/
 param name=ext-sip-ip value=1.2.3.4/

 But when I send INVITE thru the profile to the gateway associated with
 it i see local IP in Via header and in SDP m line too.
 BTW: The Contact header is populated with external IP (1.2.3.4)


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Re: [Freeswitch-users] BLF and Openzap

2009-08-16 Thread Terry Moore-Read
On Sun, Aug 16, 2009 at 7:10 AM, Brian Westbr...@freeswitch.org wrote:
 Yes you sure can... setting the presence_id to ex...@domain and have

Thats what I thought ... but where should I be setting it ?

 the phone sub to that you'll have the light come on when its in use.

 /b

 On Aug 15, 2009, at 6:29 PM, Terry Moore-Read wrote:

 Is it possible to have a sip phone show blf status for a phone which
 is connected to an openzap port ?


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Re: [Freeswitch-users] ClueCon2009 Torrents

2009-08-16 Thread Ken Rice
Lets see if this works... There are a few seeders out there...



From: Even André Fiskvik grev...@me.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Sun, 16 Aug 2009 16:55:54 +0200
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] ClueCon2009 Torrents

Can we get the torrents added to a mail on this list soon please?

Best regards,
Even André

On 16. aug.. 2009, at 05.28, Mitul Limbani wrote:

 I would be glad to offer mirror service to Cluecon 2009 videos :)
 
 Thanks  Regards,
 Mitul Limbani,
 Founder  CEO,
 Enterux Solutions Pvt. Ltd.,
 The Enterprise Linux Company (r),
 http://www.enterux.com
 http://www.entVoice.com
 
 On 16-Aug-2009, at 3:17 AM, Diego Viola diego.vi...@gmail.com wrote:
 
 Upload the torrent files in  http://files.freeswitch.org
 http://files.freeswitch.org ;)
 
 On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks  mailto:jaybi...@gmail.com
 jaybi...@gmail.com wrote:
  
 I'd also seed such a torrent.
  
  Please send the link :)
  
 
  
  
  On 16/08/2009, at 6:34, João Mesquita  mailto:jmesqu...@gmail.com
 jmesqu...@gmail.com wrote:
  
   I am interested and would also seed to the community
  
   On 8/15/09, Gabriel Gunderson  mailto:g...@gundy.org g...@gundy.org
 wrote:
   On Sat, Aug 15, 2009 at 12:13 PM, Peder
 mailto:pe...@networkoblivion.com pe...@networkoblivion.com
   wrote:
   If you want the torrents, email me off list.
  
   Why off list?  Isn't the point of torrents to have more people
   sharing
   in the load?
  
   Gabe
  
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ClueCon2009.torrent
Description: Binary data
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Re: [Freeswitch-users] ext sip rtp IP trouble

2009-08-16 Thread kokoska rokoska

Thank you very much, Brian, for your help!

Now are all IPs correct :-) (BTW: In my previous post I mean c line in
SDP, not m line). Thanks once more!

Well, signaling works fine, but I still have troubles with RTPs:
I can see with ngrep RTPs going to FreeSWITCH from both legs (a, b) but
there are not outgoing RTPs from FreeSWITCH to any of the call legs.
And, of course, caller and callee hear nothing :-)

Could you, please, help me to solve this? Thanks.

Best regards,

kokoska.rokoska




Brian West napsal(a):
 You have to set the local-network-acl on the profile so it knows when  
 or not to fix up the VIA
 /b
 
 On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:
 
 Hi all,

 I have just upgarded my (a little bit old :-) FreeSWITCH to current
 trunk and now i have troubles with IP addresses in SIP  SDP, while
 before upgrade everything work fine...

 My FreeSWITCH is behind NAT and all relevant sip profiles have set:

 param name=ext-rtp-ip value=1.2.3.4/
 param name=ext-sip-ip value=1.2.3.4/

 But when I send INVITE thru the profile to the gateway associated with
 it i see local IP in Via header and in SDP m line too.
 BTW: The Contact header is populated with external IP (1.2.3.4)
 
 
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[Freeswitch-users] Is there any freeswitch show version command

2009-08-16 Thread Muhammad Shahzad
Hi,

How can we check the version / svn revision of a running FS instance?

I know, this is kind of a stupid question, but i sometimes run into
situation where i don't know or don't have access to FS source, nor i can
restart it to get its version string.

Thank you.


-- 
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---
CISCO Rich Media Communication Specialist (CRMCS)
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Cell: +92 334 422 40 88
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
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Re: [Freeswitch-users] Is there any freeswitch show version command

2009-08-16 Thread Mathieu Rene

Hi,

type:
version

at the CLI, it'll tell you.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca




On 16-Aug-09, at 4:33 PM, Muhammad Shahzad wrote:


Hi,

How can we check the version / svn revision of a running FS instance?

I know, this is kind of a stupid question, but i sometimes run into  
situation where i don't know or don't have access to FS source, nor  
i can restart it to get its version string.


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] Is there any freeswitch show version command

2009-08-16 Thread Muhammad Shahzad
thanks.

On Mon, Aug 17, 2009 at 2:35 AM, Mathieu Rene mrene_li...@avgs.ca wrote:

 Hi,

 type: version
 at the CLI, it'll tell you.

 Mathieu Rene
 Avant-Garde Solutions Inc
 Office: + 1 (514) 664-1044 x100
 Cell: +1 (514) 664-1044 x200
 mr...@avgs.ca




 On 16-Aug-09, at 4:33 PM, Muhammad Shahzad wrote:

 Hi,

 How can we check the version / svn revision of a running FS instance?

 I know, this is kind of a stupid question, but i sometimes run into
 situation where i don't know or don't have access to FS source, nor i can
 restart it to get its version string.

 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] ext sip rtp IP trouble

2009-08-16 Thread Brian West
Ok update in a few I now recall why we had that  ! 
sofia_test_pflag(tech_pvt-profile, PFLAG_AUTO_NAT) in there.

/b

On Aug 16, 2009, at 2:40 PM, kokoska rokoska wrote:


 Thank you very much, Brian, for your help!

 Now are all IPs correct :-) (BTW: In my previous post I mean c  
 line in
 SDP, not m line). Thanks once more!

 Well, signaling works fine, but I still have troubles with RTPs:
 I can see with ngrep RTPs going to FreeSWITCH from both legs (a, b)  
 but
 there are not outgoing RTPs from FreeSWITCH to any of the call legs.
 And, of course, caller and callee hear nothing :-)

 Could you, please, help me to solve this? Thanks.

 Best regards,

 kokoska.rokoska




 Brian West napsal(a):
 You have to set the local-network-acl on the profile so it knows when
 or not to fix up the VIA
 /b

 On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:

 Hi all,

 I have just upgarded my (a little bit old :-) FreeSWITCH to current
 trunk and now i have troubles with IP addresses in SIP  SDP, while
 before upgrade everything work fine...

 My FreeSWITCH is behind NAT and all relevant sip profiles have set:

 param name=ext-rtp-ip value=1.2.3.4/
 param name=ext-sip-ip value=1.2.3.4/

 But when I send INVITE thru the profile to the gateway associated  
 with
 it i see local IP in Via header and in SDP m line too.
 BTW: The Contact header is populated with external IP (1.2.3.4)


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Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

2009-08-16 Thread Meftah Tayeb

hello
i'm rewriting this executable file in MSI format
i can use:

   * MakeMSI
   * WIX (Windows installer XML)

what you like?
WIX is open source, but MakeMSI i'm not sur
for web server, i'm replacing WAMP with a great web server called 
uniform server
this server have a conventional Unix Path configuration, for example, 
web files can by stored in /www/ and apache is in /usr/local/apache2

this facilitate the deploiment of applications
any suggestion?
thanks
Meftah Tayeb
Carlos Talbot wrote:

http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe

On Sun, Aug 16, 2009 at 1:25 PM, Meftah Tayeb tayeb.mef...@gmail.com 
mailto:tayeb.mef...@gmail.com wrote:


hello,
could you give me the installer Link to test it?
thanks
meftah tayeb
delphiwo...@irc.freenode.net mailto:delphiwo...@irc.freenode.net

* Call Me (SIP)
* Call me (skype)
* Call Me (PSTN) callto:0021348526972

Brian West wrote:

chances are mod_sofia isn't loaded.

/b

On Aug 16, 2009, at 4:43 AM, Raffaele P. Guidi wrote:

  
Can you give me a hint? Should sofia profiles be served by curl or  
something?


Thanks,
   Raffaele


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__ Information from ESET NOD32 Antivirus, version of virus signature 
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Re: [Freeswitch-users] BLF and Openzap

2009-08-16 Thread Brian West
Just as a variable on the session.

/b

On Aug 16, 2009, at 1:47 PM, Terry Moore-Read wrote:

 Thats what I thought ... but where should I be setting it ?


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Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

2009-08-16 Thread Michael Jerris
I think sticking with standard WAMP is preferable.  What is the  
advantage to creating yet another installer over the one that we have  
already done and maintained?


Mike

On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:


hello
i'm rewriting this executable file in MSI format
i can use:
MakeMSI
WIX (Windows installer XML)
what you like?
WIX is open source, but MakeMSI i'm not sur
for web server, i'm replacing WAMP with a great web server called  
uniform server
this server have a conventional Unix Path configuration, for  
example, web files can by stored in /www/ and apache is in /usr/ 
local/apache2

this facilitate the deploiment of applications
any suggestion?
thanks
Meftah Tayeb
Carlos Talbot wrote:


http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe



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[Freeswitch-users] Maxmium Number of Concurrent Sessions on EIDE Drive

2009-08-16 Thread Matthew Fong
I'm interested in doing some testing on the accuracy of mod_vmd (and
mod_amd) but wanted to see if anyone could provide some guidelines on the
maximum number of concurrent sessions I can record audio files to disk with
a typical EIDE drive under 64-bit linux without overloading my system.
Also, since I only need to record the incoming audio, would it be suggested
I use the api command uuid_record or session:record? Is there a way to only
record inbound audio with session:record? Thanks.

--matt
hello hunter corp.
http://www.hellohunter.com
hosted dialer  voice broadcasting
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[Freeswitch-users] Better results from mod_vmd

2009-08-16 Thread Matthew Fong
I tried emailed Eric, seeking advice on this, but his email (the one in the
source code) is bouncing email (invalid user), so thought I would ask here
instead. If anyone has eric's new email address, I'd be interesting in it.

I did some tests with mod_vmd this afternoon, but I'm only finding about 33%
of the voice mail beeps and did have 1 false-positive in my test of 7 voice
mail machines. I've recorded the audio of the session in .wav files that
were both successful and not, as a comparison. I can upload the .wav files
if they would be useful.

mod_vmd works great for voicemails of Skype Users, and kall8.com, but has
issues dealing with mobile phone carriers.

sprint - not successful
tmobile - not successful
verizon - not successful
panasonic home answering machine system - not successful

kall8 - SUCCESS
skype - SUCCESS

I'm wondering if you can recommend a simple fix, like changing some of the
constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source
file, or if better success requires more complex analysis.  Do you have any
recommendations on how this might be done? Listening to the .wav's
its apparent the beeps are not as loud for the mobile phone carriers as they
are with skype and kall8. Any guidance would be greatly appreciated.

--matt
hello hunter
http://www.hellohunter.com
voice broadcasting  hosted dialer
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