Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing

2009-11-27 Thread Christian Löschenkohl
hello

sorry, for my late reply
my core debugging was at info not at debug, now it's changed and i have the log 
needed

i'm sorry but pastebin doesn't work (it seems that my trace was to big)
http://pastebin.freeswitch.org/11305 says Query failure: Got a packet bigger 
than 'max_allowed_packet' bytes

i'll send the logfile personal to you, hope you don't dislike this

br

On 2009-11-24 01:48, Anthony Minessale wrote:
 You forgot to set freeswitch to debug loglevel

 You need both of the following:

 console loglevel debug
 sofia profile internal siptrace on




 2009/11/23 Christian Löschenkohl christian.loeschenk...@xpirio.com
 mailto:christian.loeschenk...@xpirio.com

 sorry about wasting your time (wasn't my intent)

 the log is at http://pastebin.freeswitch.org/11240
 i called 5214448370068 (also other calls are in the log)

 they now have changed 180 to 183 on the sonus, but makes no
 difference here

 br

 On 2009-11-23 22:07, Anthony Minessale wrote:
   do you have the ringback variable set on the channel?
   if so it will cause 180 to attempt to play inband ringback indication
  
   I have nothing left to say because I asked for the whole log with the
   siptrace enables not just 5 lines of it.
   If you still want help, give me the log to examine and I will
 tell you
   what your problem is.
  
  
  
   2009/11/23 Christian Löschenkohl
 christian.loeschenk...@xpirio.com
 mailto:christian.loeschenk...@xpirio.com
   mailto:christian.loeschenk...@xpirio.com
 mailto:christian.loeschenk...@xpirio.com
  
   thany ou for your answer
  
   we use g729 on all our other connections in passthrough mode
 and it
   also doesn't work with alaw.
   so i don't think it's related to this.
  
   br
  
  
   On 2009-11-23 20:48, Brian West wrote:
Well its also G729 so I suspect you don't have G729
   
/b
   
On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote:
   
hi
   
our freeswitch server has to talk to a sonus ip-switch
when we want to setup a call we do get a 100 Trying and then a
180 Ringing
within the 180 Ringing we get a sdp with a=sendonly then our
freeswitch
quits with a CANCEL message.
i simply don't get why our freeswitch aborts the session - i think
it would work
if no a=sendonly would be present in the sdp.
   
my technical contact doesn't want to switch 180 to 183 on the
 sonus
side - this would
also work (i think). in fact he says that 180 ringing is vaild, he
isn't that wrong in
this case.
   
our freeswitch works in proxy mode, we do use trunk 15396
see a ngrep trace under http://pastebin.freeswitch.org/11235
   
92.63.208.36 - freeswitch
38.105.229.100 - sonus
   
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
 mailto:christian.loeschenk...@xpirio.com
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   xpirio
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   Austria
  
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Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX

2009-11-27 Thread Otis
Yes. I ventured to use that  and got some error in connecting to the 
mysql database. Will try with the default sqlite before getting 
adventurous again.

Thanks


Addison Martin wrote:
 Fedora and Centos installation instructions are very similar.  You 
 should be able to compile on Fedora without any problems that I'm 
 aware of.

 Regards,

 Nik



 On Thu, Nov 26, 2009 at 06:24, Otis ab...@greatiam.com 
 mailto:ab...@greatiam.com wrote:

 Thanks. I will try it . I am on Fedora 11



 Mark Crane wrote:
  how about trying Fusionpbx.com  ( GUI) -Ram
 
  I'll second that! I released FusionPBX 1.0 RC5 today. I thought
 it was
  ready to release now but decided to do one more release
 candidate just
  to be sure. This should be the last release candidate before the
  release of version 1.0.
 
  The final release may be by the end of the week as long as no major
  issues are found.
 
  http://fusionpbx.com
 
 
 
 
  --- On *Mon, 11/23/09, ram /talk2...@gmail.com
 mailto:talk2...@gmail.com/* wrote:
 
 
  From: ram talk2...@gmail.com mailto:talk2...@gmail.com
  Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX
  To: freeswitch-users@lists.freeswitch.org
 mailto:freeswitch-users@lists.freeswitch.org
  Date: Monday, November 23, 2009, 10:54 PM
 
 
 
  On Mon, Nov 23, 2009 at 10:37 AM, Otis ab...@greatiam.com
 mailto:ab...@greatiam.com
  /mc/compose?to=ab...@greatiam.com
 mailto:ab...@greatiam.com wrote:
 
  Thanks.
 
  I have to get a centos box I guess.
 
  Much appreciated
 
  Samuel 'Otis'
 
 
 
  how about trying Fusionpbx.com  ( GUI)
 
  Ram
 
  -Inline Attachment Follows-
 
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Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS

2009-11-27 Thread Leon de Rooij

There's a little info here on how to enable it with odbc:

http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2

regards,

Leon


On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote:



On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com  
wrote:
http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- 
news-3-51-18.html


MySQL Connector/ODBC now supports batched statements. In order to  
enable

cached statement support you must switch enable the batched
statement option (FLAG_MULTI_STATEMENTS,
67108864, or Allow multiple statements
within a GUI configuration). Be aware that batched statements
create an increased chance of SQL injection attacks and you  
must

ensure that your application protects against this scenario.
   (Bug#7445)


so, is this the right patch ?

http://bugs.mysql.com/file.php?id=6994


T.

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Re: [Freeswitch-users] Re-routing calls to PSTN

2009-11-27 Thread Otis
Thank you very much . Please what are you calling a hard line ?



Andrew Thompson wrote:
 div class=moz-text-flowed style=font-family: -moz-fixed  On 
 11/26/2009 6:02 AM, Otis wrote:
 Can I get FS to re-route incoming-calls to PSTN. If this has been
 raised  before could someone  direct me to URL  or link please

 Since I don't have a hard line, I do something like:

 include
 extension name=2800
 condition field=destination_number expression=^2800$
 action application=bridge 
 data=sofia/gateway/YOURPROVIDER/18005551212/
 /condition
 /extension
 /include



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Re: [Freeswitch-users] Re-routing calls to PSTN

2009-11-27 Thread Evgeniy Zolotov
Please try this:

action application=bridge data=openzap/1/A/${number}/

Otis ?:
 Thank you very much . Please what are you calling a hard line ?



 Andrew Thompson wrote:
   
 div class=moz-text-flowed style=font-family: -moz-fixed  On 
 11/26/2009 6:02 AM, Otis wrote:
 
 Can I get FS to re-route incoming-calls to PSTN. If this has been
 raised  before could someone  direct me to URL  or link please
   
 Since I don't have a hard line, I do something like:

 include
 extension name=2800
 condition field=destination_number expression=^2800$
 action application=bridge 
 data=sofia/gateway/YOURPROVIDER/18005551212/
 /condition
 /extension
 /include

 


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Re: [Freeswitch-users] Bypass_media and re_invite

2009-11-27 Thread Steve Kurzeja
Is that USD ? :)


On Thu, Nov 26, 2009 at 10:10 AM, Anthony Minessale 
anthony.miness...@gmail.com wrote:

 I can spare you the pain and let you know outright that this sort of
 functionality will cost somewhere in the range of 125,000.00 to 150,000.00
 to properly implement by assembling a team of consultants including members
 of the development team from both FreeSWITCH and Sofia-SIP and even if you
 have the money, finding the time to implement it would also be a factor as
 it's a few thousand man-hours of work.


 On Wed, Nov 25, 2009 at 3:01 PM, Mathieu Rene mrene_li...@avgs.ca wrote:

 You can read all about the sip library at
 http://sofia-sip.sourceforge.net/refdocs/

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




 On 25-Nov-09, at 3:58 PM, srinivasula reddy wrote:

 thanks for your reply mike,
 is there any api in freeswitch or any thing else to update lib
 programatically from pjsua.

 srinivas

 On Thu, Nov 26, 2009 at 2:05 AM, Michael Jerris m...@jerris.com wrote:

 something that is not available in that lib at this time.

 Mike

 On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote

 can please tell me how can i exchange session state into sip library.

 Thanks
 srinivas

 On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris m...@jerris.comwrote:

 For that you would need to fully exchange session state into the sip
 library, *something that is not available in that lib at this time.*


 On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote:

 HI,
 thanks for your reply, my requirement is i am doing failover stuff with
 freeswitch. i dont want cut the calls when freeswitch dies, when failover
 happens mean one freeswitch dies we are going to start the second
 freeswitch, i dont want close call intiated by the  first freeswtich, they
 are communicating with meida(bypass media). when one endpoing try to end 
 the
 call at that time i want to close the call for the other end also.


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 --
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 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
 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
 IRC: irc.freenode.net #freeswitch

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 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] Connecting Multiple domains

2009-11-27 Thread Otis

Could someone please direct me to a link for connecting multiple say 2 
domains each with their own FS server. 

Thanks





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Re: [Freeswitch-users] Bypass_media and re_invite

2009-11-27 Thread Tihomir Culjaga
On Fri, Nov 27, 2009 at 11:00 AM, Steve Kurzeja steve.kurz...@gmail.comwrote:

 Is that USD ? :)


i believe these are not Turkish liras

:P
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Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS

2009-11-27 Thread Frank @ Impact
Thanks.  But when I made these entries in /etc/odbc.ini and rebooted.
 
[freeswitch]
Driver  = MySQL
SERVER  = 127.0.0.1
PORT= 4040
DATABASE= mydb
OPTIONS = 67108864
 
.I still get FS complaining with this.
 
Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744
[WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched
statements!#012If you are using mysql, make sure you are using MYODBC
3.51.18 or higher and enable FLAG_MULTI_STATEMENTS
 
FreeSWITCHversion
FreeSWITCH Version 1.0.trunk (15660)
 
Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 EST
2009 x86_64 x86_64 x86_64 GNU/Linux
 
From /etc/odbcinst.ini
DRIVER = /usr/lib64/libmyodbc5-5.1.5.so
Setup = /usr/lib64/libodbcmyS.so
 
Is this a FS issue ?  or an issue with mysql odbc?  Any insight would be
great.
 
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Leon
de Rooij
Sent: Friday, November 27, 2009 3:37 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS
 
There's a little info here on how to enable it with odbc:
 
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2
 
regards,
 
Leon
 
 
On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote:



 
On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com wrote:
http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html
 
MySQL Connector/ODBC now supports batched statements. In order to enable
cached statement support you must switch enable the batched
statement option (FLAG_MULTI_STATEMENTS,
67108864, or Allow multiple statements
within a GUI configuration). Be aware that batched statements
create an increased chance of SQL injection attacks and you must
ensure that your application protects against this scenario.
   (Bug#7445 http://bugs.mysql.com/7445 )
 

so, is this the right patch ?

http://bugs.mysql.com/file.php?id=6994


T. 

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Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS

2009-11-27 Thread Leon de Rooij

Are you using the myodbc 3.51.18 version or higher ?

I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to  
upgrade from jaunty..


regards,

Leon


On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote:


Thanks.  But when I made these entries in /etc/odbc.ini and rebooted…

[freeswitch]
Driver  = MySQL
SERVER  = 127.0.0.1
PORT= 4040
DATABASE= mydb
OPTIONS = 67108864

…I still get FS complaining with this.

Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744  
[WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched  
statements!#012If you are using mysql, make sure you are using  
MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS


FreeSWITCHversion
FreeSWITCH Version 1.0.trunk (15660)

Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04  
EST 2009 x86_64 x86_64 x86_64 GNU/Linux


From /etc/odbcinst.ini
DRIVER = /usr/lib64/libmyodbc5-5.1.5.so
Setup = /usr/lib64/libodbcmyS.so

Is this a FS issue ?  or an issue with mysql odbc?  Any insight  
would be great.


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org 
] On Behalf Of Leon de Rooij

Sent: Friday, November 27, 2009 3:37 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS

There's a little info here on how to enable it with odbc:

http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2

regards,

Leon


On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote:



On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com  
wrote:
http://dev.mysql.com/doc/refman/5.1/en/connector-odbc- 
news-3-51-18.html


MySQL Connector/ODBC now supports batched statements. In order to  
enable

cached statement support you must switch enable the batched
statement option (FLAG_MULTI_STATEMENTS,
67108864, or Allow multiple statements
within a GUI configuration). Be aware that batched statements
create an increased chance of SQL injection attacks and you  
must

ensure that your application protects against this scenario.
   (Bug#7445)


so, is this the right patch ?

http://bugs.mysql.com/file.php?id=6994


T.

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Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS

2009-11-27 Thread Frank @ Impact
Yes. I am using version 5.1  I am using Fedora 12.
 
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Leon
de Rooij
Sent: Friday, November 27, 2009 10:19 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS
 
Are you using the myodbc 3.51.18 version or higher ?
 
I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to
upgrade from jaunty..
 
regards,
 
Leon
 
 
On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote:



Thanks.  But when I made these entries in /etc/odbc.ini and rebooted.
 
[freeswitch]
Driver  = MySQL
SERVER  = 127.0.0.1
PORT= 4040
DATABASE= mydb
OPTIONS = 67108864
 
.I still get FS complaining with this.
 
Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744
[WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched
statements!#012If you are using mysql, make sure you are using MYODBC
3.51.18 or higher and enable FLAG_MULTI_STATEMENTS
 
FreeSWITCHversion
FreeSWITCH Version 1.0.trunk (15660)
 
Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 EST
2009 x86_64 x86_64 x86_64 GNU/Linux
 
From /etc/odbcinst.ini
DRIVER = /usr/lib64/libmyodbc5-5.1.5.so
Setup = /usr/lib64/libodbcmyS.so
 
Is this a FS issue ?  or an issue with mysql odbc?  Any insight would be
great.
 
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Leon
de Rooij
Sent: Friday, November 27, 2009 3:37 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS
 
There's a little info here on how to enable it with odbc:
 
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2
 
regards,
 
Leon
 
 
On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote:




 
On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com wrote:
http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html
 
MySQL Connector/ODBC now supports batched statements. In order to enable
cached statement support you must switch enable the batched
statement option (FLAG_MULTI_STATEMENTS,
67108864, or Allow multiple statements
within a GUI configuration). Be aware that batched statements
create an increased chance of SQL injection attacks and you must
ensure that your application protects against this scenario.
   (Bug#7445 http://bugs.mysql.com/7445 )
 

so, is this the right patch ?

http://bugs.mysql.com/file.php?id=6994


T.

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Re: [Freeswitch-users] FS missing in action (was: How to run IVR application)

2009-11-27 Thread Andrew Thompson
  On 11/27/2009 12:59 AM, ovvenkat wrote:
 Hi MC,

 I have created won sample application yesterday, It was working fine. 
 Today, I checked that my local ip has changed. so, I changed the 
 domain(IP) name in sip-account settings in my x-lite configuration. 
 After that  x-lite is not able to register with FS. I am getting error 
 like Registration error 405 : Method not Allowed . Could you please 
 tell me ,why its happening ?

Wait, what?

First, don't re-use an existing thread, messages have a tendancy to get 
ignored/lost that way.

Did the IP of your FS change, or of your PC?

I would expect local ip to mean your DHCP'ed address from your 
Internet connection. That should have no bearing on the IP of your FS. 
Go find your FS and make sure *IT* is still on the IP you expect it to 
be on.

If your PC running x-lite is also your FS, you may have other issues 
with IP address changing that I don't know how to handle, as I've only 
used static IPs for FS. (Or, try connecting via localhost, 127.0.0.1 
instead, until you're ready to really start using FS.)

-- 
Andrew Thompson


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[Freeswitch-users] Transfer Problem

2009-11-27 Thread Jonathan Barou
Hi everybody,

I'm actually using the lastest version of Freeswitch, I have a problem. I
have a trunk SIP with my PABX.

There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX)
 2. one Alcatel IpTouch 4028 with number 987 (on
PABX)
 3.  one Siemens Gigaset A580 IP with number
8401 (on Freeswitch)


*The first test* is to say to the phone 2 to transfer all the call to number
8401. So when I dial 987 on the phone 1, all work perfectly, the phone 3 is
 ringing and it's work. I have that in the log :

2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending early
media

2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP
[sofia/internal/3...@10.33.69.246] 10.33.169.92 port 23054 - 10.33.69.246
port 32000 codec: 8 ms: 90

2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer [soft]
720 bytes per 90ms

2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP:

v=0

o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92

s=FreeSWITCH

c=IN IP4 10.33.169.92

t=0 0

m=audio 23054 RTP/AVP 8 106

a=rtpmap:8 PCMA/8000

a=rtpmap:106 telephone-event/8000

a=fmtp:106 0-16

a=silenceSupp:off - - - -

a=ptime:90

a=sendrecv


2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer
sofia/internal/3...@10.33.69.246!

2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/sip:8...@10.33.170.231:5060 [BREAK]

2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/
sip:8...@10.33.170.231:5060 receive message [DISPLAY]

2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/
3...@10.33.69.246 skipping state [early][183]

2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal
sofia/internal/3...@10.33.69.246 [BREAK]

2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec
Activation Success l...@8000hz 1 channel 90ms

2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play Ringback
Tone [%(2000,4000,440.0,480.0)]

2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/
3...@10.33.69.246 receive message [TRANSCODING_NECESSARY]

2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port
confirmed.

2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read Buffer
at 1440 bytes vs 81



*The Second Tes*t is to say to the phone 1 to transfer all the call to
number 8401. So when I dial 368 on the phone 2, the phone 3 is  ringing just
one time and after it hangup. I have that in the log :


2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending early
media

2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP
[sofia/internal/9...@10.33.69.246] 10.33.169.92 port 27732 - 10.33.69.144
port 32000 codec: 8 ms: 90

2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer [soft]
720 bytes per 90ms

2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP:

v=0

o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92

s=FreeSWITCH

c=IN IP4 10.33.169.92

t=0 0

m=audio 27732 RTP/AVP 8 106

a=rtpmap:8 PCMA/8000

a=rtpmap:106 telephone-event/8000

a=fmtp:106 0-16

a=silenceSupp:off - - - -

a=ptime:90

a=sendrecv


2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer
sofia/internal/9...@10.33.69.246!

2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal
sofia/internal/sip:8...@10.33.170.231:5060 [BREAK]

2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/
sip:8...@10.33.170.231:5060 receive message [DISPLAY]

2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/
9...@10.33.69.246 skipping state [early][183]

2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal
sofia/internal/9...@10.33.69.246 [BREAK]

2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec
Activation Success l...@8000hz 1 channel 90ms

2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play Ringback
Tone [%(2000,4000,440.0,480.0)]

2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/
9...@10.33.69.246 receive message [TRANSCODING_NECESSARY]

2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/
9...@10.33.69.246 entering state [terminated][487]

2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/
9...@10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL]

2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/9...@10.33.69.246 [KILL]

2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal
sofia/internal/9...@10.33.69.246 [BREAK]

2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 thread
mismatch skipping state handler.

2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore original
codec.

2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup
sofia/internal/sip:8...@10.33.170.231:5060 [CS_CONSUME_MEDIA]
[ORIGINATOR_CANCEL]

2009-11-27 

Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing

2009-11-27 Thread Anthony Minessale
or you can put it at a url on your web site and just post a link


2009/11/27 Christian Löschenkohl christian.loeschenk...@xpirio.com

 hello

 sorry, for my late reply
 my core debugging was at info not at debug, now it's changed and i have the
 log needed

 i'm sorry but pastebin doesn't work (it seems that my trace was to big)
 http://pastebin.freeswitch.org/11305 says Query failure: Got a packet
 bigger than 'max_allowed_packet' bytes

 i'll send the logfile personal to you, hope you don't dislike this

 br

 On 2009-11-24 01:48, Anthony Minessale wrote:
  You forgot to set freeswitch to debug loglevel
 
  You need both of the following:
 
  console loglevel debug
  sofia profile internal siptrace on
 
 
 
 
  2009/11/23 Christian Löschenkohl christian.loeschenk...@xpirio.com
  mailto:christian.loeschenk...@xpirio.com
 
  sorry about wasting your time (wasn't my intent)
 
  the log is at http://pastebin.freeswitch.org/11240
  i called 5214448370068 (also other calls are in the log)
 
  they now have changed 180 to 183 on the sonus, but makes no
  difference here
 
  br
 
  On 2009-11-23 22:07, Anthony Minessale wrote:
do you have the ringback variable set on the channel?
if so it will cause 180 to attempt to play inband ringback
 indication
   
I have nothing left to say because I asked for the whole log with
 the
siptrace enables not just 5 lines of it.
If you still want help, give me the log to examine and I will
  tell you
what your problem is.
   
   
   
2009/11/23 Christian Löschenkohl
  christian.loeschenk...@xpirio.com
  mailto:christian.loeschenk...@xpirio.com
mailto:christian.loeschenk...@xpirio.com
  mailto:christian.loeschenk...@xpirio.com
   
thany ou for your answer
   
we use g729 on all our other connections in passthrough mode
  and it
also doesn't work with alaw.
so i don't think it's related to this.
   
br
   
   
On 2009-11-23 20:48, Brian West wrote:
 Well its also G729 so I suspect you don't have G729

 /b

 On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote:

 hi

 our freeswitch server has to talk to a sonus ip-switch
 when we want to setup a call we do get a 100 Trying and then
 a
 180 Ringing
 within the 180 Ringing we get a sdp with a=sendonly then
 our
 freeswitch
 quits with a CANCEL message.
 i simply don't get why our freeswitch aborts the session - i
 think
 it would work
 if no a=sendonly would be present in the sdp.

 my technical contact doesn't want to switch 180 to 183 on the
  sonus
 side - this would
 also work (i think). in fact he says that 180 ringing is vaild,
 he
 isn't that wrong in
 this case.

 our freeswitch works in proxy mode, we do use trunk 15396
 see a ngrep trace under http://pastebin.freeswitch.org/11235

 92.63.208.36 - freeswitch
 38.105.229.100 - sonus

 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
  mailto:christian.loeschenk...@xpirio.com
mailto:christian.loeschenk...@xpirio.com
  mailto:christian.loeschenk...@xpirio.com

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Re: [Freeswitch-users] Re-routing calls to PSTN

2009-11-27 Thread Otis
Ok. Thanks


Andrew Thompson wrote:
 div class=moz-text-flowed style=font-family: -moz-fixed  On 
 11/27/2009 3:37 AM, Otis wrote:
 Thank you very much . Please what are you calling a hard line ?
 A real honest to goodness POTS line within 20 feet of and attached to 
 my FS server.

 My calls come in and go out SIP, so If I was sending an inbound call 
 back to the PSTN, I'd just route it back out through my SIP provider.



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Re: [Freeswitch-users] Freeswitch admin GUI

2009-11-27 Thread Otis
Hi

I am no sure but read up on fusionpbx. I asked the same question and 
someone pointed me to that.
check web site http://fusionpbx.com

Regards



Samuel Mukoti wrote:
 div class=moz-text-flowed style=font-family: -moz-fixedHi,

 Any recommendations for apps that can I use ontop of freeswitch as a 
 GUI manager, to manage extensions, queues, ivr, and dialplans?

 Thanks

 Sam


 On 27 Nov,2009, at 5:19 PM, 
 freeswitch-users-requ...@lists.freeswitch.org wrote:

 Send FreeSWITCH-users mailing list submissions to
freeswitch-users@lists.freeswitch.org

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 You can reach the person managing the list at
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 When replying, please edit your Subject line so it is more specific
 than Re: Contents of FreeSWITCH-users digest...


 Today's Topics:

   1. Re: odbc FLAG_MULTI_STATMENTS (Leon de Rooij)


 --

 Message: 1
 Date: Fri, 27 Nov 2009 16:19:03 +0100
 From: Leon de Rooij l...@scarlet-internet.nl
 Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS
 To: freeswitch-users@lists.freeswitch.org
 Message-ID: a8d9107a-f8c9-4705-9cf5-b72fe9083...@scarlet-internet.nl
 Content-Type: text/plain; charset=windows-1252

 Are you using the myodbc 3.51.18 version or higher ?

 I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to
 upgrade from jaunty..

 regards,

 Leon


 On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote:

 Thanks.  But when I made these entries in /etc/odbc.ini and rebooted?

 [freeswitch]
 Driver  = MySQL
 SERVER  = 127.0.0.1
 PORT= 4040
 DATABASE= mydb
 OPTIONS = 67108864

 ?I still get FS complaining with this.

 Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744
 [WARNING] sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched
 statements!#012If you are using mysql, make sure you are using
 MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS

 FreeSWITCHversion
 FreeSWITCH Version 1.0.trunk (15660)

 Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04
 EST 2009 x86_64 x86_64 x86_64 GNU/Linux

 From /etc/odbcinst.ini
 DRIVER = /usr/lib64/libmyodbc5-5.1.5.so
 Setup = /usr/lib64/libodbcmyS.so

 Is this a FS issue ?  or an issue with mysql odbc?  Any insight
 would be great.

 -Original Message-
 From: freeswitch-users-boun...@lists.freeswitch.org 
 [mailto:freeswitch-users-boun...@lists.freeswitch.org
 ] On Behalf Of Leon de Rooij
 Sent: Friday, November 27, 2009 3:37 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS

 There's a little info here on how to enable it with odbc:

 http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2

 regards,

 Leon


 On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote:



 On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com
 wrote:
 http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-
 news-3-51-18.html

 MySQL Connector/ODBC now supports batched statements. In order to
 enable
cached statement support you must switch enable the batched
statement option (FLAG_MULTI_STATEMENTS,
67108864, or Allow multiple statements
within a GUI configuration). Be aware that batched statements
create an increased chance of SQL injection attacks and you
 must
ensure that your application protects against this scenario.
   (Bug#7445)


 so, is this the right patch ?

 http://bugs.mysql.com/file.php?id=6994


 T.

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 End of FreeSWITCH-users Digest, Vol 41, Issue 209
 *


 /div

Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing

2009-11-27 Thread Anthony Minessale
please update to latest trunk 15698 or greater and re-test.
The 183 from the provider had a sendonly attr that tricked the proxy code
into thinking it was a hold/unhold operation.


On Fri, Nov 27, 2009 at 10:57 AM, Anthony Minessale 
anthony.miness...@gmail.com wrote:

 or you can put it at a url on your web site and just post a link


 2009/11/27 Christian Löschenkohl christian.loeschenk...@xpirio.com

 hello

 sorry, for my late reply
 my core debugging was at info not at debug, now it's changed and i have
 the log needed

 i'm sorry but pastebin doesn't work (it seems that my trace was to big)
 http://pastebin.freeswitch.org/11305 says Query failure: Got a packet
 bigger than 'max_allowed_packet' bytes

 i'll send the logfile personal to you, hope you don't dislike this

 br

 On 2009-11-24 01:48, Anthony Minessale wrote:
  You forgot to set freeswitch to debug loglevel
 
  You need both of the following:
 
  console loglevel debug
  sofia profile internal siptrace on
 
 
 
 
  2009/11/23 Christian Löschenkohl christian.loeschenk...@xpirio.com
  mailto:christian.loeschenk...@xpirio.com
 
  sorry about wasting your time (wasn't my intent)
 
  the log is at http://pastebin.freeswitch.org/11240
  i called 5214448370068 (also other calls are in the log)
 
  they now have changed 180 to 183 on the sonus, but makes no
  difference here
 
  br
 
  On 2009-11-23 22:07, Anthony Minessale wrote:
do you have the ringback variable set on the channel?
if so it will cause 180 to attempt to play inband ringback
 indication
   
I have nothing left to say because I asked for the whole log with
 the
siptrace enables not just 5 lines of it.
If you still want help, give me the log to examine and I will
  tell you
what your problem is.
   
   
   
2009/11/23 Christian Löschenkohl
  christian.loeschenk...@xpirio.com
  mailto:christian.loeschenk...@xpirio.com
mailto:christian.loeschenk...@xpirio.com
  mailto:christian.loeschenk...@xpirio.com
   
thany ou for your answer
   
we use g729 on all our other connections in passthrough mode
  and it
also doesn't work with alaw.
so i don't think it's related to this.
   
br
   
   
On 2009-11-23 20:48, Brian West wrote:
 Well its also G729 so I suspect you don't have G729

 /b

 On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote:

 hi

 our freeswitch server has to talk to a sonus ip-switch
 when we want to setup a call we do get a 100 Trying and then
 a
 180 Ringing
 within the 180 Ringing we get a sdp with a=sendonly then
 our
 freeswitch
 quits with a CANCEL message.
 i simply don't get why our freeswitch aborts the session - i
 think
 it would work
 if no a=sendonly would be present in the sdp.

 my technical contact doesn't want to switch 180 to 183 on the
  sonus
 side - this would
 also work (i think). in fact he says that 180 ringing is
 vaild, he
 isn't that wrong in
 this case.

 our freeswitch works in proxy mode, we do use trunk 15396
 see a ngrep trace under http://pastebin.freeswitch.org/11235

 92.63.208.36 - freeswitch
 38.105.229.100 - sonus

 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
  mailto:christian.loeschenk...@xpirio.com
mailto:christian.loeschenk...@xpirio.com
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Re: [Freeswitch-users] Callback to the user in ESL

2009-11-27 Thread Anthony Minessale
I told you to make a new separate inbound connection back to the server from
your script, do not use the same one thta was tethered to the call because
its too late to use that one.

Why do I have to answer you twice?


On Thu, Nov 26, 2009 at 3:27 AM, lakshmanan ganapathy
lakindi...@gmail.comwrote:

 Hi, Any help or suggestion regarding my previous post. Especially


 I also noted that, if I don't receive any events, especially
 SERVER_DISCONNECTED, then the connection is in established state, but once
 I receive the SERVER_DISCONNECTED event, the connection is closed. Is it
 correct??
 Here is the program by which I confirmed the above!


 require ESL;
 use IO::Socket::INET;

 my $ip = 192.168.1.222;
 my $sock = new IO::Socket::INET ( LocalHost = $ip,  LocalPort = '8447',
 Proto = 'tcp',  Listen = 2,  Reuse = 1 );
 die Could not create socket: $!\n unless $sock;
 my $con;
 my $type = user/;

 for(;;) {
 # wait for any client to connect, a new client will get connected
 when a new call comes in the dialplan.

 my $new_sock = $sock-accept();
 # Do fork and let the parent to wait for more clients.

 my $pid = fork();
 if ($pid) {
 close($new_sock);
 next;
 }
 # Extract the host of the client.

 my $host = $new_sock-sockhost();
 # file descriptor for the socket.

 my $fd = fileno($new_sock);
 print Host name is $host\n;
 # Create object for the ESL connection package to access the ESL
 functions.

 $con = new ESL::ESLconnection($fd);
 # Gets the info about this channel.

 my $info = $con-getInfo();
 my $uuid = $info-getHeader(unique-id);
 printf Connected call %s, from %s to %s\n, $uuid,
 $info-getHeader(caller-caller-id-number),
 $info-getHeader(caller-destination-number);

 # Answer the channel.
 $con-execute(answer);
 # Set the event lock to tell the FS to execute the instructions in
 the given order.
 $con-setEventLock(true);
 # Play a file  Get the personal number from the user.

 $con-execute(playback,/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav);
 $con-execute(hangup);

 while($con-connected())
 {
 my $e=$con-recvEvent();
 my $ename=$e-getHeader(Event-Name);
 print $e-serialize();
 print $ename\n;
 print Connection exists\n;
 sleep(1);

 }
 print
 Bye\n--\n;
 close($new_sock);
 }
 I've not registered for any events.
 In the above program I'm receiving the SERVER_DISCONNECTED event.
 Output when receiving event:
 Host name is 192.168.1.222
 Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097
 Event-Name: SERVER_DISCONNECTED

 SERVER_DISCONNECTED
 Connection exists
 Bye

 When I comment the recvEvent line, I got the following output.

 Host name is 192.168.1.222
 Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097
 Connection exists
 Connection exists
 Connection exists
 Connection exists
 Connection exists



 On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy 
 lakindi...@gmail.com wrote:

 I've tried the following program as per the suggestion that you've told.
 But it seems, no success. Once the connection is closed, I created a new
 connection and I send originate to originate a new call. But it is not
 working.

 require ESL;
 use IO::Socket::INET;
 use Data::Dumper;

 my $ip = 192.168.1.222;
 my $sock = new IO::Socket::INET ( LocalHost = $ip,  LocalPort = '8447',
 Proto = 'tcp',  Listen = 2,  Reuse = 1 );
 die Could not create socket: $!\n unless $sock;

 my $make_call;
 my $con;
 my $type = user/;

 for(;;) {
 my $new_sock = $sock-accept();
 my $pid = fork();
 if ($pid) {
 close($new_sock);
 next;
 }
 my $host = $new_sock-sockhost();
 my $fd = fileno($new_sock);
 $con = new ESL::ESLconnection($fd);
 my $info = $con-getInfo();
 my $uuid = $info-getHeader(unique-id);
 printf Connected call %s, from %s to %s\n, $uuid,
 $info-getHeader(caller-caller-id-number),
 $info-getHeader(caller-destination-number);

 $con-filter(Unique-Id, $uuid);
 $con-events(plain, all);
 $con-execute(answer);
 $con-setEventLock(true);
 my $number=$con-execute(read,2 4
 /usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav
 accnt_number 5000 #);
 while($con-connected())
 {
 my $e=$con-recvEvent();
 my $ename=$e-getHeader(Event-Name);
 my $app=$e-getHeader(Application);
 if($ename eq CHANNEL_EXECUTE_COMPLETE and $app eq
 read)
 {

Re: [Freeswitch-users] Connecting Multiple domains

2009-11-27 Thread eman
check out
http://wiki.freeswitch.org/wiki/Multi-tenant

On Fri, Nov 27, 2009 at 6:15 AM, Otis ab...@greatiam.com wrote:


 Could someone please direct me to a link for connecting multiple say 2
 domains each with their own FS server.

 Thanks





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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS

2009-11-27 Thread eman
Check out the Linksys SPA2102

On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith 
mcampbellsm...@gmail.com wrote:

 The only ATA mentioned on the WIKI that supports TLS/SRTP is the
 Grandstream HandyTone 503.  But, again according to the wiki, that
 doesn't seem to behave to well with TLS ...

 On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote:
  Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
  Does the SPA3102 support TLS or only SRTP?
 
  I don't know, but supporting only SRTP would be ridiculous, since the
 keys
  would then be transmitted in the clear and therefore amenable to
 interception.
  SRTP requires the SIP channel to be encrypted by TLS in order to be
 secure.
  ZRTP, on the other hand, doesn't have this limitation: it works entirely
 in
  RTP.
 
  I would be rather surprised were a hardware manufacturer to implement
 SRTP
  without TLS for the SIP traffic. On the other hand, we've seen often in
 this
  forum that some manufacturers are really clueless...
 
 
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Re: [Freeswitch-users] Recording with Native File PCMU

2009-11-27 Thread Anthony Minessale
If you want to go to that much trouble consider http://www.orecx.com/web/
They snoop the RTP from an entirely different box and record your calls for
you.



On Wed, Nov 25, 2009 at 7:35 PM, Seven Du dujinf...@gmail.com wrote:

 Yeah, that's why I had to record to two files(readwrite) and need to mix
 together by using sox.  Do you only try to using PCMU to save CPU power
 matt? As Anthony said, the difference can be ignored. And you also need to
 take extra effort to make sure transcoding will not happen on a
 conversation.

 But it maybe useful for expensive codecs like g729, iLBC, speex etc for
 recording heavy scenarios. I'd like to take a look if there is a 5k bounty
 ;)

 2009/11/26 Anthony Minessale anthony.miness...@gmail.com

 The processor power saved is negligible between PCMU and raw PCM and not
 worth the fuss.
 If you didn't decode the audio first you would not be able to mix the
 stream to produce a single file.
 So if we went to the trouble of making native media bugs to be able to do
 that you could barely use them so it would not be worth the 5k or more
 bounty to develop that functionality.



 On Sun, Nov 22, 2009 at 12:48 PM, Matthew Fong mattdf...@gmail.comwrote:

 I'm trying to conserve processor power by recording in native file
 format, PCMU in my case. It works great with the following line

 session:execute(record,
 /tmp/my_recording...session:getVariable(read_codec));

 however it fails to work with

 session:execute(record_session,
 /tmp/my_recording...session:getVariable(read_codec));
 or
 record = api:execute(sched_api, '+1 none uuid_record
 '..session:getVariable(uuid)..' start
 /tmp/my_recording.'..session:getVariable(read_codec));

 Why is it that it works with record, but not with record_session or
 uuid_record? Is there something I'm over looking? In the latter two the
 consul reports

 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File
 [/tmp/my_recording.PCMU] 8000hz

 as if it's recording, but /tmp/my_recording.PCMU never shows up. However
 if I change it to .wav instead of .PCMU it works. Any ideas?

 --matt

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googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
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Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.

2009-11-27 Thread Anthony Minessale
Did you check the 2 replies that told you you need aliases in your sofia
profile to translate the domain found in your message_waiting to the right
profile?  Both Brian and Mike answered you.




On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX prometheus...@gmx.net wrote:

 I tried now with phones directly attached to the freeswitch (without an
 OpenSIPS in between). I also added the alias. But the behaviour is as
 before:
 No notify message from freeswitch, neither after register nor after a
 voicemail is recorded.

 Best regards
 Peter
 Brian West schrieb:
  Yes an alias will be required for every domain you run on the profile
  so it can find it.
 
  /b
 
  On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
 
 
  Try an alias on the sip profile.
 
  Mike
 
 
 
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iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
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Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.

2009-11-27 Thread Michael Jerris
Does the alias you added match the one that you saw in the event?  The alias is 
100% for sure the fix for this issue, please check again.

Mike

On Nov 26, 2009, at 6:55 PM, Peter P GMX wrote:

 I tried now with phones directly attached to the freeswitch (without an
 OpenSIPS in between). I also added the alias. But the behaviour is as
 before:
 No notify message from freeswitch, neither after register nor after a
 voicemail is recorded.
 
 Best regards
 Peter
 Brian West schrieb:
 Yes an alias will be required for every domain you run on the profile  
 so it can find it.
 
 /b
 
 On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
 
 
 Try an alias on the sip profile.
 
 Mike
 


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

2009-11-27 Thread Anthony Minessale
based on your example past

sip1.mydomain.com is the domain in the packet and thus the profile should
have an alias for this.
Then the user must reside in your sip db with the user 200 and domain
sip1.mydomain.com

if you dont have this consider the force-register-domain and
force-register-db-domain to normalize the host names.


On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale 
anthony.miness...@gmail.com wrote:

 Did you check the 2 replies that told you you need aliases in your sofia
 profile to translate the domain found in your message_waiting to the right
 profile?  Both Brian and Mike answered you.





 On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX prometheus...@gmx.netwrote:

 I tried now with phones directly attached to the freeswitch (without an
 OpenSIPS in between). I also added the alias. But the behaviour is as
 before:
 No notify message from freeswitch, neither after register nor after a
 voicemail is recorded.

 Best regards
 Peter
 Brian West schrieb:
  Yes an alias will be required for every domain you run on the profile
  so it can find it.
 
  /b
 
  On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
 
 
  Try an alias on the sip profile.
 
  Mike
 
 
 
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Re: [Freeswitch-users] Freeswitch admin GUI

2009-11-27 Thread Adam Ford
://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091127/
9c86b324/attachment.html 


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 End of FreeSWITCH-users Digest, Vol 41, Issue 209
 *


 /div



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Re: [Freeswitch-users] Freeswitch admin GUI

2009-11-27 Thread Traun Leyden
-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04
  EST 2009 x86_64 x86_64 x86_64 GNU/Linux
 
  From /etc/odbcinst.ini
  DRIVER = /usr/lib64/libmyodbc5-5.1.5.so
  Setup = /usr/lib64/libodbcmyS.so
 
  Is this a FS issue ?  or an issue with mysql odbc?  Any insight
  would be great.
 
  -Original Message-
  From: freeswitch-users-boun...@lists.freeswitch.org
  [mailto:freeswitch-users-boun...@lists.freeswitch.org
  ] On Behalf Of Leon de Rooij
  Sent: Friday, November 27, 2009 3:37 AM
  To: freeswitch-users@lists.freeswitch.org
  Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS
 
  There's a little info here on how to enable it with odbc:
 
  http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2
 
  regards,
 
  Leon
 
 
  On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote:
 
 
 
  On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com
  wrote:
  http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-
  news-3-51-18.html
 
  MySQL Connector/ODBC now supports batched statements. In order to
  enable
 cached statement support you must switch enable the batched
 statement option (FLAG_MULTI_STATEMENTS,
 67108864, or Allow multiple statements
 within a GUI configuration). Be aware that batched statements
 create an increased chance of SQL injection attacks and you
  must
 ensure that your application protects against this scenario.
(Bug#7445)
 
 
  so, is this the right patch ?
 
  http://bugs.mysql.com/file.php?id=6994
 
 
  T.
 
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[Freeswitch-users] custom call counter

2009-11-27 Thread Juan Backson
Hi,

Instead of using show calls count to obtain the current call count stat, I
am writing some C code to increment a counter during on_answer_hook and
decrement the counter during on_hangup_hook.

It looks like my counter result is very closed to show calls count when
the traffic is low, like 50 -60.  But when traffic is high, like 1000 calls,
my counter is showing 30% less.

When all calls are finished, my counter becomes 0 again, and that proves
that it does not increment/decrement more than it should.

Is this normal?  Does anyone have any idea why there is such discrepancy?

thanks,
jb
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Re: [Freeswitch-users] custom call counter

2009-11-27 Thread Michael Jerris
It depends on the timing of when your increment and decrement are vs when the 
sql calls to push the events into the tables that are used for show calls are.  
Also, the sql calls are batched and queued causing a little delay (less than a 
second).  If your doing a lot of short lived calls there is sure to be timing 
discrepancy.  I am sure there is even more discrepancy if you look at the 
output of status which shows the current number of sessions (those are 
individual call legs) as that information is a little more real time.

Mike

On Nov 28, 2009, at 12:48 AM, Juan Backson wrote:

 Hi,
 
 Instead of using show calls count to obtain the current call count stat, I 
 am writing some C code to increment a counter during on_answer_hook and 
 decrement the counter during on_hangup_hook.
 
 It looks like my counter result is very closed to show calls count when the 
 traffic is low, like 50 -60.  But when traffic is high, like 1000 calls, my 
 counter is showing 30% less.
 
 When all calls are finished, my counter becomes 0 again, and that proves that 
 it does not increment/decrement more than it should.
 
 Is this normal?  Does anyone have any idea why there is such discrepancy?
 
 thanks,
 jb
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