Re: [Freeswitch-users] [Windows] Stable enough for production use?

[Peter Olsson]

Hi,

We've been running FS in win32 in a "semi-production" environment for some time 
now (since version 1.0.1 - following the trunk all the time since then). We use 
it as both a lab environment - to distribute SIP-trunks to different PBX'es, 
and also for "real" endpoints for some (about 10-15) of our internal users. 
Over the time there have been a few issues (now many though), but from 1.0.4 
and later it's been very stable for our use, and no memory leaks etc.

We haven't tried analog PSTN connections though, only SIP, h323 (opal) and some 
(not much) use of Sangoma E1 connected to one of our PBX'es.

/Peter

Från: freeswitch-users-boun...@lists.freeswitch.org 
[freeswitch-users-boun...@lists.freeswitch.org] för Fred-145 
[codecompl...@free.fr]
Skickat: den 16 december 2009 08:38
Till: freeswitch-users@lists.freeswitch.org
Ämne: [Freeswitch-users]  [Windows] Stable enough for production use?

Hello

Since Freeswitch is also available for Windows (and Mac, but I don't
anything about Macintosh), I'd like some feedback from users who routinely
run Freeswitch on that OS.

Is it stable enough to be used in production to handle a single analog line
(ie. SOHO use), or should I warn customers that they really should buy a
dedicated Linux box to run FS?

Thank you.
--
View this message in context: 
http://old.nabble.com/-Windows--Stable-enough-for-production-use--tp26807322p26807322.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

[Peter Olsson]

Thanks Brian, it works now. I'll try to learn to say NO next time :)

/Peter


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Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 26 oktober 2009 19:40
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

At some point we'll have to NO NO NO fix your broken crap.  :P  The  
reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with  
me... "NO!"

/b

On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote:

> I understand your frustration :) We deal with SIP integration with  
> about 10 different PBX vendors today, And it's always something that  
> doesn't work as it should. Right now I don't have anything more  
> connected to FS though.
>
> /Peter


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Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

[Peter Olsson]

I understand your frustration :) We deal with SIP integration with about 10 
different PBX vendors today, And it's always something that doesn't work as it 
should. Right now I don't have anything more connected to FS though.

/Peter


-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 26 oktober 2009 18:46
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Bet your hardware just barfs on those like others have... I mean  
really I HATE SIP. This is stupid.

/b

On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:

> In the non-working one I don't have these, and instead I have these  
> headers;
>   X-FS-Display-Name: 9099
>   X-FS-Display-Number: 9099
>   X-FS-Support: update_display
>   P-Asserted-Identity: "9099" <9099>


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Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

[Peter Olsson]

Yes, I know :) However, now I think this is related to the new headers 
introduced, it's probably not a TCP issue.

Everything seems to work just fine until the 200 OK is sent, the Avaya PBX 
doesn't seem to accept that reply anymore.

The only differences I've found between a working revision, and a non-working 
is this;

In the working 200-OK transaction method UPDATE is listed in the Allow-header, 
and there is a header called "X-Actually-Support: UPDATE".

In the non-working one I don't have these, and instead I have these headers;
   X-FS-Display-Name: 9099
   X-FS-Display-Number: 9099
   X-FS-Support: update_display
   P-Asserted-Identity: "9099" <9099>

So I guess this could be related to the other thread going on right now 
"Downloaded tar vs latest SVN - 200 OK has more headers", and not a TCP issue. 
It's probably the Avaya doing something wrong (not the first time), but still 
it seems these changes affect more systems than mine.

I'm just using it as a lab setup for now, so if anyone want me to test 
something, I can do it immediately.

/Peter


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[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 26 oktober 2009 18:23
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Finding the exact rev that broke it would be helpful.

/b

On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote:

> Hmm... I remembered incorrectly about my setup :) The Avaya PBX  
> talks TLS to the Avaya SES Server, and then UDP to FS, not TCP -  
> sorry, my bad!
>
> However, something that has changed the last 10 days seems to affect  
> my setup so it doesn't work anymore. I'll do some more SIP tracing,  
> and get back when I know more about it.
>
> /Peter


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Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

[Peter Olsson]

Hmm... I remembered incorrectly about my setup :) The Avaya PBX talks TLS to 
the Avaya SES Server, and then UDP to FS, not TCP - sorry, my bad!

However, something that has changed the last 10 days seems to affect my setup 
so it doesn't work anymore. I'll do some more SIP tracing, and get back when I 
know more about it.

/Peter


-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Kristian Kielhofner
Skickat: den 26 oktober 2009 16:47
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Tony,

  It seemed strange to me too (I'm using TCP in other places).

  I'll take another look at this with your suggestions for debugging.

  Thanks!

On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale
 wrote:
> i cant seem to reproduce it.
>
> originate sofia/internal/1...@conference.freeswitch.org;transport=tcp 9998
>
> I get a working call and trace.
>
> Could you possibly have a dns error?  I know it's an ip but it may still
> fail if it has no dns.
>
> try
>
> sofia loglevel all 9
>
> and look for other errors.
>
>
>
>
> On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner
>  wrote:
>>
>> I originally sent this last Friday but I've been unable to confirm it
>> ever made it to the list.
>>
>> Hello everyone,
>>
>>  I'm having some issues with SIP and TCP.  I've used it before with
>> success but I'm seeing some strange behavior...
>>
>>  Level 7 debugs with siptrace on both profiles.  UDP invite from
>> softphone comes in on port 5062, it's supposed to bridge to
>> 10.70.0.62.  When configured to use UDP FS sends an INVITE (nothing
>> currently answers) while TCP doesn't send anything (confirmed with
>> siptrace and packet sniffer).  I confirmed this behavior with a
>> gateway configured for TCP and appending ;transport=tcp to a bridge
>> line.
>>
>>  This is trunk rev 15211 on an Intel Mac running Snow Leopard.  I've
>> also confirmed this behavior on an Intel Linux machine running Ubuntu
>> (not sure of version ATM).
>>
>> TCP:
>>
>> http://pastebin.freeswitch.org/10825
>>
>> UDP:
>>
>> http://pastebin.freeswitch.org/10826
>>
>> dialplan (UDP):
>>
>>   
>>     
>>       
>>       
>>       > data="effective_caller_id_number=19412848354"/>
>>       
>>     
>>   
>>
>> dialplan (TCP):
>>
>>   
>>     
>>       
>>       
>>       > data="effective_caller_id_number=19412848354"/>
>>       > data="sofia/avaya/7...@10.70.0.62;transport=tcp"/>
>>     
>>   
>>
>>  Any thoughts?
>>
>> Thanks!
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> ___
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> 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
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:213-799-1400
>
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>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

[Peter Olsson]

I'm also having problems with this. When running FS compiled about 10 days ago 
it works fine (don't remember exact revision), but when using latest SVN it 
doesn't work anymore. I seems like it's trying to use UDP when it should use 
TCP.

My setup is this: Avaya Communication Manager PBX -> Talks TLS to Avaya SIP SES 
Server -> Talks TCP to FreeSwitch.

The replies from FS seems to be sent using UDP instead of TCP, and when I keep 
the config and revert to the 10 day old version it starts working again, so 
there is definately something wrong.

I'll try to do some more testing, and get back with some SIP-traces as well.

/Peter


-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Kristian Kielhofner
Skickat: den 26 oktober 2009 16:47
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Tony,

  It seemed strange to me too (I'm using TCP in other places).

  I'll take another look at this with your suggestions for debugging.

  Thanks!

On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale
 wrote:
> i cant seem to reproduce it.
>
> originate sofia/internal/1...@conference.freeswitch.org;transport=tcp 9998
>
> I get a working call and trace.
>
> Could you possibly have a dns error?  I know it's an ip but it may still
> fail if it has no dns.
>
> try
>
> sofia loglevel all 9
>
> and look for other errors.
>
>
>
>
> On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner
>  wrote:
>>
>> I originally sent this last Friday but I've been unable to confirm it
>> ever made it to the list.
>>
>> Hello everyone,
>>
>>  I'm having some issues with SIP and TCP.  I've used it before with
>> success but I'm seeing some strange behavior...
>>
>>  Level 7 debugs with siptrace on both profiles.  UDP invite from
>> softphone comes in on port 5062, it's supposed to bridge to
>> 10.70.0.62.  When configured to use UDP FS sends an INVITE (nothing
>> currently answers) while TCP doesn't send anything (confirmed with
>> siptrace and packet sniffer).  I confirmed this behavior with a
>> gateway configured for TCP and appending ;transport=tcp to a bridge
>> line.
>>
>>  This is trunk rev 15211 on an Intel Mac running Snow Leopard.  I've
>> also confirmed this behavior on an Intel Linux machine running Ubuntu
>> (not sure of version ATM).
>>
>> TCP:
>>
>> http://pastebin.freeswitch.org/10825
>>
>> UDP:
>>
>> http://pastebin.freeswitch.org/10826
>>
>> dialplan (UDP):
>>
>>   
>>     
>>       
>>       
>>       > data="effective_caller_id_number=19412848354"/>
>>       
>>     
>>   
>>
>> dialplan (TCP):
>>
>>   
>>     
>>       
>>       
>>       > data="effective_caller_id_number=19412848354"/>
>>       > data="sofia/avaya/7...@10.70.0.62;transport=tcp"/>
>>     
>>   
>>
>>  Any thoughts?
>>
>> Thanks!
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> ___
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> 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
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:213-799-1400
>
> ___
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> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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Re: [Freeswitch-users] mod_opal

[Peter Olsson]

Tihomir,

Yes as I remember it I did get the correct caller id number. I think you need 
to set variable origination_caller_id_number when you originate a call.

/Peter



Från: Tihomir Culjaga 
Skickat: den 1 september 2009 19:20
Till: freeswitch-users@lists.freeswitch.org 

Ämne: Re: [Freeswitch-users] mod_opal

Hi Peter,

i did it on linux... it was enough to use

svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/trunk ptlib
svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/branches/v3_6 
opal

this is something that works well :)

BTW: do you get a correct callingPartyNumber when you place calls through 
opal/h323?

I'm always getting 000 even if i set effective_caller_id_number to some 
value...


T.


On Tue, Sep 1, 2009 at 8:37 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

Please look into MODOPAL-10 in jira. You need to apply a patch if you’re using 
latest opal trunk, ro else you need to use the latest stable version of opal. 
However, I’m not sure how automated this is in the build process in Linux. I’ve 
only done this on Windows builds lately.



/Peter



Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 
[mailto:freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>]
 För Tihomir Culjaga
Skickat: den 1 september 2009 08:09
Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] mod_opal



hhmmm :))

is there any doc following up mod_opal ?
I really don't want to waste your time :)

T.


On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins 
mailto:m...@freeswitch.org>> wrote:



On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga 
mailto:tculj...@gmail.com>> wrote:

hello,

i'm trying to build mod_opal and getting this error:



making all mod_logfile

making all mod_loopback

making all mod_native_file

making all mod_opal
Compiling mod_opal.cpp...
quiet_libtool: compile:  g++ -g -ggdb -I. 
-I/home/tculjaga/freeswitch-trunk/src/include 
-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC 
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 
-D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal 
-DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
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&)
/usr/include/opal/opal/patch.h: In member function ‘switch_status_t 
FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)’:
/usr/include/opal/opal/patch.h:272: error: ‘virtual 
ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ 
is private
mod_opal.cpp:1277: error: within this context
mod_opal.cpp:1277: warning: ‘virtual 
ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ 
is deprecated (declared at /usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1277: warning: ignoring return value of function declared with 
attribute warn_unused_result
/usr/include/opal/opal/patch.h: In member function ‘switch_status_t 
FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)’:
/usr/include/opal/opal/patch.h:272: error: ‘virtual 
ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ 
is private
mod_opal.cpp:1399: error: within this context
mod_opal.cpp:1399: warning: ‘virtual 
ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ 
is deprecated (declared at /usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1399: warning: ignoring return value of function declared with 
attribute warn_unused_result
make[5]: *** [mod_opal.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_opal-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 + FreeSWITCH Build Complete ---+
 + FreeSWITCH has been successfully built.  +
 + Install by running:

Re: [Freeswitch-users] mod_opal

[Peter Olsson]

Please look into MODOPAL-10 in jira. You need to apply a patch if you're using 
latest opal trunk, ro else you need to use the latest stable version of opal. 
However, I'm not sure how automated this is in the build process in Linux. I've 
only done this on Windows builds lately.

/Peter

Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Tihomir Culjaga
Skickat: den 1 september 2009 08:09
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] mod_opal

hhmmm :))

is there any doc following up mod_opal ?
I really don't want to waste your time :)

T.

On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins 
mailto:m...@freeswitch.org>> wrote:

On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga 
mailto:tculj...@gmail.com>> wrote:
hello,

i'm trying to build mod_opal and getting this error:



making all mod_logfile

making all mod_loopback

making all mod_native_file

making all mod_opal
Compiling mod_opal.cpp...
quiet_libtool: compile:  g++ -g -ggdb -I. 
-I/home/tculjaga/freeswitch-trunk/src/include 
-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC 
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 
-D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal 
-DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
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&)
/usr/include/opal/opal/patch.h: In member function 'switch_status_t 
FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)':
/usr/include/opal/opal/patch.h:272: error: 'virtual 
ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()' 
is private
mod_opal.cpp:1277: error: within this context
mod_opal.cpp:1277: warning: 'virtual 
ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()' 
is deprecated (declared at /usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1277: warning: ignoring return value of function declared with 
attribute warn_unused_result
/usr/include/opal/opal/patch.h: In member function 'switch_status_t 
FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)':
/usr/include/opal/opal/patch.h:272: error: 'virtual 
ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()' 
is private
mod_opal.cpp:1399: error: within this context
mod_opal.cpp:1399: warning: 'virtual 
ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()' 
is deprecated (declared at /usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1399: warning: ignoring return value of function declared with 
attribute warn_unused_result
make[5]: *** [mod_opal.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_opal-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 + FreeSWITCH Build Complete ---+
 + FreeSWITCH has been successfully built.  +
 + Install by running:  +
 +  +
 +   make install   +
 +--+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
tculj...@nemesis:~/freeswitch-trunk$
tculj...@nemesis:~/freeswitch-trunk$
tculj...@nemesis:~/freeswitch-trunk$



what ptlib/opal/fs version did you use to build it?


I tried with trunk (ptlib, opal, fs)... and as you can see :)

Did you run the buildopal.sh script in src/build ?
-MC


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Re: [Freeswitch-users] opal build error

[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"  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] Sangoma/FS...

[Peter Olsson]

Of course – no problem!

I’m not using libpri support now, I don’t think it’s ported for Windows (yet)?

I’ll try it out some more, and try to detect what’s going wrong...

/Peter

Från: Moises Silva [mailto:moises.si...@gmail.com]
Skickat: den 12 augusti 2009 16:17
Till: freeswitch-users@lists.freeswitch.org
Kopia: Peter Olsson
Ämne: Re: FW: Sangoma/FS...

Hello Peter,

I'd appreciate if you can keep the discussion going in the freeswitch-users 
mailing list, there are other people there that will benefit of the discussion 
or even can help. Read my comments below.

On Wed, Aug 12, 2009 at 5:59 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

Sorry for spamming you :) But I have some more results now. I’ve tried using 
another lab PBX with Q.SIG enabled, and when using that one I’m able to connect 
calls as I should. At least incoming to FS, outgoing seem to have some problems 
still.. So the problem for the PBX I used yesterday seems to be both related to 
Q.SIG (maybe) and the PBX itself (it does connect to other providers though, so 
I know the trunk works).



Should I take some dumps from the PRI card to try to find out why it didn’t 
work with the first one, or is this “as expected”, since they have Q.SIG 
enabled?
I have no experience with Q.SIG, so I won't be able to help much. One thing 
though, is that if I were you, I'd be using openzap with libpri support, is 
that what you are using, or are you using the ISDN openzap stack?

As of the dumps, they may help, or not, but pastebin them anyways so I can make 
an un-educated guess.

--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com<mailto:m...@sangoma.com>
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[Freeswitch-users] UniMRCP support for Windows...

[Peter Olsson]

Yet another Windows question from me.. :)

I've seen that vcproj-files for mod_unimrcp has been added to SVN, but they are 
not yet included in the main FreeSWITCH solution-file.

What's the curent status of this, should it work if I compile it separately, or 
is it not yet complete for Windows?

/Peter

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[Freeswitch-users] OpenZAP/Sangoma in Windows

[Peter Olsson]

Hi, I'm trying to evaluate the OpenZAP/Sangoma-support in Windows, using PRI E1 
connections.

I'm aware it's not yet to be considered as stable, but I'd still want to try it 
out some, and also help detecting bugs while I'm at it :)

Anyway, I have a couple of questions:

1. Has anyone tested it in Windows at all? I know the build-files for Visual 
Studio has only been checked in for a couple of months, so that's why I'm 
asking.

2. Does anyone have any directions how to configure the driver within Windows? 
Should I use BitStream or HDLC, and how should the channel groups be configured?

I have built everything correctly, and I've succeeded to open (some of) the 
channels when I configure each channel in a separate group, but I don't think 
that's the way it should be done...

Any suggestions welcome!

/Peter

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Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...

[Peter Olsson]

What can I say - you guys provide far much better (and quicker) support then 
any commersial solution :) Thanks for the help!

/Peter



Från: Brian West 
Skickat: den 25 juni 2009 17:53
Till: freeswitch-users@lists.freeswitch.org 

Ämne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...

Fixed revision 13948.

/b

On Jun 25, 2009, at 10:22 AM, Peter Olsson wrote:

Done, added as issue SFSIP-157.

Regards,

Peter Olsson

Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 25 juni 2009 10:16
Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...

Please open a jira and attach sip traces of register and phone calls.

/b

On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote:


I’ve been using FS as a gateway to our OCS server for some time. It’s used just 
for testing, so it’s not really used every day. I don’t know when, but after 
some trunk update (right now I running r13945) of FS it doesn’t send the SIP 
traffic using tcp anymore (OCS only accepts tcp or tls).

My configuration is quite easy, I have a sofia gateway configured to OCS, this 
has the parameter  set in the 
config (nothing in the config has changed for ages). Then in the dialplan I use 
this gateway to connect the calls. When doing a siptrace I can see that the 
headers has transport=tcp set correctly, but according to the trace it’s sent 
using udp instead of tcp.

Has something changed so I need to configure it in another way, or is it just 
simply a bug? I just wanted to check this before issuing a jira case and 
providing more specific information and debug traces etc.

/Peter
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Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...

[Peter Olsson]

Done, added as issue SFSIP-157.

Regards,

Peter Olsson

Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 25 juni 2009 10:16
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...

Please open a jira and attach sip traces of register and phone calls.

/b

On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote:


I've been using FS as a gateway to our OCS server for some time. It's used just 
for testing, so it's not really used every day. I don't know when, but after 
some trunk update (right now I running r13945) of FS it doesn't send the SIP 
traffic using tcp anymore (OCS only accepts tcp or tls).

My configuration is quite easy, I have a sofia gateway configured to OCS, this 
has the parameter  set in the 
config (nothing in the config has changed for ages). Then in the dialplan I use 
this gateway to connect the calls. When doing a siptrace I can see that the 
headers has transport=tcp set correctly, but according to the trace it's sent 
using udp instead of tcp.

Has something changed so I need to configure it in another way, or is it just 
simply a bug? I just wanted to check this before issuing a jira case and 
providing more specific information and debug traces etc.

/Peter
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[Freeswitch-users] Outgoing sofia calls not using tcp anymore...

[Peter Olsson]

I've been using FS as a gateway to our OCS server for some time. It's used just 
for testing, so it's not really used every day. I don't know when, but after 
some trunk update (right now I running r13945) of FS it doesn't send the SIP 
traffic using tcp anymore (OCS only accepts tcp or tls).

My configuration is quite easy, I have a sofia gateway configured to OCS, this 
has the parameter  set in the 
config (nothing in the config has changed for ages). Then in the dialplan I use 
this gateway to connect the calls. When doing a siptrace I can see that the 
headers has transport=tcp set correctly, but according to the trace it's sent 
using udp instead of tcp.

Has something changed so I need to configure it in another way, or is it just 
simply a bug? I just wanted to check this before issuing a jira case and 
providing more specific information and debug traces etc.

/Peter
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Re: [Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL?

[Peter Olsson]

Yes I guess this would probably solve the issue :) But since I stumbled across 
this weird behaviour I just wanted to make sure if this was expected or not, or 
if it might be a bug...

I thought playback was just sending the audio to the caller, but in this case 
it seems that playback sends it to both "parties".

/Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För seven
Skickat: den 18 juni 2009 09:20
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Weird (or expected) behaviour on record/playback 
using ESL?

can you try uuid_record  stop  before playback?


On Jun 18, 2009, at 3:07 PM, Peter Olsson wrote:


I'm not quite sure if this is the expected behaviour, I just wanted to make 
sure.

I've developed a simple IVR application using event socket. I dial in to the 
dialplan and park the call, and then I let the IVR application do whatever it's 
supposed to. I basically listen for DTMF events and play and record files.

Today I just noticed that if I issue a "api uuid_record  start 
", and then do a file playback (using SendMsg, with call-command 
execute and execute-app-name playback), the playback is sent both to the 
caller, and to the recorded file. Is this the way it's supposed to work, or 
should I playback files in another way?

/Peter
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[Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL?

[Peter Olsson]

I'm not quite sure if this is the expected behaviour, I just wanted to make 
sure.

I've developed a simple IVR application using event socket. I dial in to the 
dialplan and park the call, and then I let the IVR application do whatever it's 
supposed to. I basically listen for DTMF events and play and record files.

Today I just noticed that if I issue a "api uuid_record  start 
", and then do a file playback (using SendMsg, with call-command 
execute and execute-app-name playback), the playback is sent both to the 
caller, and to the recorded file. Is this the way it's supposed to work, or 
should I playback files in another way?

/Peter
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Re: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 (Jon DiVita)

[Peter Olsson]

If you use latest stable opal package you will be able to compile without 
patching mod_opal. That is what the guys from Opal recommend. The latest trunk 
has lots of things going on, so I'm not sure the patches I have supplied are 
sufficient anymore..

To load the module you must also enable it as a module in modules.conf.xml.

/Peter


-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Jonathan DiVita
Skickat: den 17 juni 2009 17:20
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 
6-19-09 (Jon DiVita)

I was able to compile mod_opal with the patches you suggested.  Now, once
that is done is there anything more I need to do inorder for mod_opal to
show up when I run freeswitch?

I didn't see that it had started when I ran freeswitch after the mod_opal
compile.

Thanks!

Jon
- Original Message -
From: 
To: 
Sent: Wednesday, June 17, 2009 12:26 AM
Subject: Freeswitch-users Digest, Vol 36, Issue 159


> Send Freeswitch-users mailing list submissions to
> freeswitch-users@lists.freeswitch.org
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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: Compiling Issues: Opal with Latest SVN Builds 6-19-09
>  (Brian West)
>   2. Re: How can I join two freeswitch on two servers? (Edmar Cruz)
>   3. Re: session.getDigits() not working (Brian West)
>   4. Re: How can I join two freeswitch on two servers? (Brian West)
>   5. Re: How can I join two freeswitch on two servers? (Edmar Cruz)
>   6. Re: How can I join two freeswitch on two servers? (Edmar Cruz)
>   7. Re: How can I join two freeswitch on two servers? (Brian West)
>   8. Re: How can I join two freeswitch on two servers? (Edmar Cruz)
>
>
> --
>
> Message: 1
> Date: Tue, 16 Jun 2009 22:12:01 -0500
> From: Brian West 
> Subject: Re: [Freeswitch-users] Compiling Issues: Opal with Latest SVN
> Builds 6-19-09
> To: freeswitch-users@lists.freeswitch.org
> Message-ID: 
> Content-Type: text/plain; charset="iso-8859-1"
>
> please see MODOPAL-10 on jira.
>
> /b
>
> On Jun 16, 2009, at 10:05 PM, Jonathan DiVita wrote:
>
>> Hello, all.  I'm currently playing around with a new install of
>> Freeswitch and wanted to try out mod_opal.  Below are the current
>> SVN builds for  opal, ptlib, and freeswitch.  I end up with the
>> following errors when compiling.
>>
>> making all mod_opal
>> Compiling mod_opal.cpp...
>> Compiling mod_opal.cpp ...
>> 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&)
>> make[4]: *** [mod_opal.lo] Error 1
>> make[3]: *** [all] Error 1
>> make[2]: *** [mod_opal-all] Error 1
>> make[1]: *** [mod_opal] Error 2
>> make: *** [mod_opal] Error 2
>
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/a7bb947f/attachment-0001.html
>
> --
>
> Message: 2
> Date: Tue, 16 Jun 2009 20:03:13 -0700 (PDT)
> From: Edmar Cruz 
> Subject: Re: [Freeswitch-users] How can I join two freeswitch on two
> servers?
> To: freeswitch-users@lists.freeswitch.org
> Message-ID: <929908.28142...@web57310.mail.re1.yahoo.com>
> Content-Type: text/plain; charset="us-ascii"
>
> How can i turn off authentication? This is my acl.conf.xml on
> 192.168.0.105
>
>
> 
> 
> 
>
> On 192.168.0.4
>
> 
> 
> 
>
>
> 
> From: Brian West 
> To: freeswitch-users@lists.freeswitch.org
> Sent: Tuesday, June 16, 2009 10:49:58 PM
> Subject: Re: [Free

[Freeswitch-users] UniMRCP - current status?

[Peter Olsson]

I can see that the UniMRCP libs have been added to FS lately, I was just 
wondering about the current status/stability for this implementation? And will 
this be ported to Windows as well?

Just curious - since you guys add more features all the time :)

/Peter

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Re: [Freeswitch-users] Allow invites from another sip server?

[Peter Olsson]

You need to change the apply-inbound-acl to your list (myip) instead of using 
the domains list.

/Peter


- Ursprungligt meddelande -
Från: Edmar Cruz 
Skickat: den 17 juni 2009 08:20
Till: freeswitch-users@lists.freeswitch.org 

Ämne: [Freeswitch-users]  Allow invites from another sip server?


On my acl.conf.xml

I allow the ip 116.50.110.2

Is this correct?


   


Error sip_invite() ... Error occur rejected by acl domains

param name="apply-inbound-acl" value="domains"/>
param name="apply-register_acl" value="domains"/>

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Re: [Freeswitch-users] Inbound using FS

[Peter Olsson]

If you don't even see it when debug logging is enabled, there is something 
wrong in the other end.

About the IP's. I guess you're just faking IP's in these email,s or are you 
using 2.2.2.2 and 1.1.1.1 for real? Cause in that case you're in trouble. I 
just wanted to make sure... :)

/Peter


-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Rex_Alex
Skickat: den 9 juni 2009 15:55
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Inbound using FS


Hello,

Below are some changes I have made, Post me if any additions required...

acl.conf.xml 

 
   
 

freeswitch.xml 

 

sip_profiles/internal.xml 

< param name="apply-inbound-acl" value="inbound_ac" />  under  tag 

public.xml 

 
 
 
 
 

default.xml 

 
 
 
 
 
 
 

Thanks
Rex

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Re: [Freeswitch-users] Inbound using FS

[Peter Olsson]

Are you able to see anything at all in the console/log?

I'm starting to doubt that the call even gets into the FS box... :)

Try enabling more logs (console loglevel debug), and try again.


On 09-06-08 19.04, "Rex_Alex"  wrote:





Hello,

 Yes you are right. they are the on the same LAN. Inboun-acl added
in internal profile as well and restarted the FS completely. But no luck..
Please help us to resolve the same..

Thanks,
Rex


Peter Olsson wrote:
>
> The PRI/SIP-box probably talks to the internal profile (I guess that they
> are on the same LAN). Try to add the inbound-acl to the internal profile
> as well. Also restart FS completely, just to be 100% sure that config is
> reloaded.
>
> //Peter
>
>
> On 09-06-08 18.34, "Rex_Alex"  wrote:
>
>
>
> Hello,
>
> I am not sure about the profile which I am calling into.
>
> My scenario is this, I am trying to reach extn 1007 registered in FS
> server
> from my mobile through an inbound PRI connected to the audiocode with DID
> 123456.
>
> Thanks,
> Rex
>
>
>
> Peter Olsson wrote:
>>
>> I don't see what you've added. But I guess it's something like .
>>
>> Are you sure you're dialing into the external profile? It's on port 5080
>> by default, and the internal is on 5060.
>>
>> /Peter
>>
>>
>> On 09-06-08 17.53, "Rex_Alex"  wrote:
>>
>>
>>
>> Hello Peter,
>>
>> Yes, I have added
>>
>>
>>
>> under  tag in sip_profiles/external.xml
>>
>> Thanks,
>> Rex
>>
>>
>>
>> Peter Olsson wrote:
>>>
>>> Have you configured the sip profile to use the acl list you have created
>>> (Inbound_Test)?
>>>
>>> /Peter
>>>
>>>
>>> - Ursprungligt meddelande -
>>> Från: Rex_Alex 
>>> Skickat: den 8 juni 2009 17:40
>>> Till: freeswitch-users@lists.freeswitch.org
>>> 
>>> Ämne: Re: [Freeswitch-users] Inbound using FS
>>>
>>>
>>> Hi Brian,
>>>
>>> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1
>>>
>>> Addions made are,
>>>
>>> acl.conf.xml
>>>
>>> 
>>>   
>>> 
>>>
>>> freeswitch.xml
>>>
>>> 
>>>
>>> sip_profiles/external.xml
>>>
>>>  under  tag
>>>
>>> public.xml
>>>
>>> 
>>> 
>>> 
>>> 
>>> 
>>>
>>> default.xml
>>>
>>> 
>>> 
>>> 
>>> 
>>> >> data="sofia/internal/1007%1.1.1.1"/>
>>> 
>>> 
>>>
>>> Still, Inbound is not hitting my FS console itself. Please assist where
>>> am
>>> I
>>> going wrong?
>>>
>>> Thanks,
>>> Rex
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Brian West wrote:
>>>>
>>>>
>>>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote:
>>>>
>>>>>
>>>>> Hello,
>>>>>
>>>>> My public.xml configration is:
>>>>>
>>>>>   
>>>>>  
>>>>>
>>>>>
>>>>>   
>>>>
>>>> $1 will not exist in this case because your regular expression doesn't
>>>> capture anything.  So replace $1 with your target number or use
>>>> ^(123456)$
>>>>
>>>>
>>>>>
>>>>> My default.xml configration is:
>>>>>
>>>>>  
>>>>>
>>>>>  
>>>>>  
>>>>>  
>>>>>
>>>>>  
>>>>
>>>> Can you elaborate how you're registering with your provider?
>>>>
>>>>
>>>>>
>>>>>
>>>>> When I am trying to call 123456 from my mobile no. Not able to see any
>>>>> logging in FS console. Please assist where I am going wrong? Or do I
>>>>> require
>>>>> any extra modules to be installed?
>>>>>
>>>>> Thanks,
>>>>> Rex
>>>>
>>>> Brian West
>>>> br...@freeswitch.org
>>>>
>>>> -- Meet us at ClueCon!  http://www.cluecon.com
&g

Re: [Freeswitch-users] Inbound using FS

[Peter Olsson]

The PRI/SIP-box probably talks to the internal profile (I guess that they are 
on the same LAN). Try to add the inbound-acl to the internal profile as well. 
Also restart FS completely, just to be 100% sure that config is reloaded.

//Peter


On 09-06-08 18.34, "Rex_Alex"  wrote:



Hello,

I am not sure about the profile which I am calling into.

My scenario is this, I am trying to reach extn 1007 registered in FS server
from my mobile through an inbound PRI connected to the audiocode with DID
123456.

Thanks,
Rex



Peter Olsson wrote:
>
> I don't see what you've added. But I guess it's something like .
>
> Are you sure you're dialing into the external profile? It's on port 5080
> by default, and the internal is on 5060.
>
> /Peter
>
>
> On 09-06-08 17.53, "Rex_Alex"  wrote:
>
>
>
> Hello Peter,
>
> Yes, I have added
>
>
>
> under  tag in sip_profiles/external.xml
>
> Thanks,
> Rex
>
>
>
> Peter Olsson wrote:
>>
>> Have you configured the sip profile to use the acl list you have created
>> (Inbound_Test)?
>>
>> /Peter
>>
>>
>> - Ursprungligt meddelande -
>> Från: Rex_Alex 
>> Skickat: den 8 juni 2009 17:40
>> Till: freeswitch-users@lists.freeswitch.org
>> 
>> Ämne: Re: [Freeswitch-users] Inbound using FS
>>
>>
>> Hi Brian,
>>
>> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1
>>
>> Addions made are,
>>
>> acl.conf.xml
>>
>> 
>>   
>> 
>>
>> freeswitch.xml
>>
>> 
>>
>> sip_profiles/external.xml
>>
>>  under  tag
>>
>> public.xml
>>
>> 
>> 
>> 
>> 
>> 
>>
>> default.xml
>>
>> 
>> 
>> 
>> 
>> > data="sofia/internal/1007%1.1.1.1"/>
>> 
>> 
>>
>> Still, Inbound is not hitting my FS console itself. Please assist where
>> am
>> I
>> going wrong?
>>
>> Thanks,
>> Rex
>>
>>
>>
>>
>>
>>
>>
>> Brian West wrote:
>>>
>>>
>>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote:
>>>
>>>>
>>>> Hello,
>>>>
>>>> My public.xml configration is:
>>>>
>>>>   
>>>>  
>>>>
>>>>
>>>>   
>>>
>>> $1 will not exist in this case because your regular expression doesn't
>>> capture anything.  So replace $1 with your target number or use
>>> ^(123456)$
>>>
>>>
>>>>
>>>> My default.xml configration is:
>>>>
>>>>  
>>>>
>>>>  
>>>>  
>>>>  
>>>>
>>>>  
>>>
>>> Can you elaborate how you're registering with your provider?
>>>
>>>
>>>>
>>>>
>>>> When I am trying to call 123456 from my mobile no. Not able to see any
>>>> logging in FS console. Please assist where I am going wrong? Or do I
>>>> require
>>>> any extra modules to be installed?
>>>>
>>>> Thanks,
>>>> Rex
>>>
>>> Brian West
>>> br...@freeswitch.org
>>>
>>> -- Meet us at ClueCon!  http://www.cluecon.com
>>>
>>>
>>>
>>>
>>>
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>>>
>>>
>>
>> --
>> View this message in context:
>> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>>
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Re: [Freeswitch-users] Inbound using FS

[Peter Olsson]

I don't see what you've added. But I guess it's something like .

Are you sure you're dialing into the external profile? It's on port 5080 by 
default, and the internal is on 5060.

/Peter


On 09-06-08 17.53, "Rex_Alex"  wrote:



Hello Peter,

Yes, I have added



under  tag in sip_profiles/external.xml

Thanks,
Rex



Peter Olsson wrote:
>
> Have you configured the sip profile to use the acl list you have created
> (Inbound_Test)?
>
> /Peter
>
>
> - Ursprungligt meddelande -
> Från: Rex_Alex 
> Skickat: den 8 juni 2009 17:40
> Till: freeswitch-users@lists.freeswitch.org
> 
> Ämne: Re: [Freeswitch-users] Inbound using FS
>
>
> Hi Brian,
>
> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1
>
> Addions made are,
>
> acl.conf.xml
>
> 
>   
> 
>
> freeswitch.xml
>
> 
>
> sip_profiles/external.xml
>
>  under  tag
>
> public.xml
>
> 
> 
> 
> 
> 
>
> default.xml
>
> 
> 
> 
> 
>  data="sofia/internal/1007%1.1.1.1"/>
> 
> 
>
> Still, Inbound is not hitting my FS console itself. Please assist where am
> I
> going wrong?
>
> Thanks,
> Rex
>
>
>
>
>
>
>
> Brian West wrote:
>>
>>
>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote:
>>
>>>
>>> Hello,
>>>
>>> My public.xml configration is:
>>>
>>>   
>>>  
>>>
>>>
>>>   
>>
>> $1 will not exist in this case because your regular expression doesn't
>> capture anything.  So replace $1 with your target number or use
>> ^(123456)$
>>
>>
>>>
>>> My default.xml configration is:
>>>
>>>  
>>>
>>>  
>>>  
>>>  
>>>
>>>  
>>
>> Can you elaborate how you're registering with your provider?
>>
>>
>>>
>>>
>>> When I am trying to call 123456 from my mobile no. Not able to see any
>>> logging in FS console. Please assist where I am going wrong? Or do I
>>> require
>>> any extra modules to be installed?
>>>
>>> Thanks,
>>> Rex
>>
>> Brian West
>> br...@freeswitch.org
>>
>> -- Meet us at ClueCon!  http://www.cluecon.com
>>
>>
>>
>>
>>
>> ___
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>>
>>
>
> --
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> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Inbound using FS

[Peter Olsson]

Have you configured the sip profile to use the acl list you have created 
(Inbound_Test)?

/Peter


- Ursprungligt meddelande -
Från: Rex_Alex 
Skickat: den 8 juni 2009 17:40
Till: freeswitch-users@lists.freeswitch.org 

Ämne: Re: [Freeswitch-users] Inbound using FS


Hi Brian,

I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1

Addions made are,

acl.conf.xml


  


freeswitch.xml



sip_profiles/external.xml

 under  tag

public.xml







default.xml









Still, Inbound is not hitting my FS console itself. Please assist where am I
going wrong?

Thanks,
Rex







Brian West wrote:
>
>
> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote:
>
>>
>> Hello,
>>
>> My public.xml configration is:
>>
>>   
>>  
>>
>>
>>   
>
> $1 will not exist in this case because your regular expression doesn't
> capture anything.  So replace $1 with your target number or use
> ^(123456)$
>
>
>>
>> My default.xml configration is:
>>
>>  
>>
>>  
>>  
>>  
>>
>>  
>
> Can you elaborate how you're registering with your provider?
>
>
>>
>>
>> When I am trying to call 123456 from my mobile no. Not able to see any
>> logging in FS console. Please assist where I am going wrong? Or do I
>> require
>> any extra modules to be installed?
>>
>> Thanks,
>> Rex
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
> ___
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Re: [Freeswitch-users] Taking long at startup

[Peter Olsson]

Klaus,

This is probably caused by the new nat features introduced in FreeSWITCH. You 
can start FS with -nonat to skip this detection.

/Peter


-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Klaus Teller
Skickat: den 8 juni 2009 16:28
Till: freeswitch-users@lists.freeswitch.org
Ämne: [Freeswitch-users] Taking long at startup

Hi,

Freeswitch is taking quiet some time to start. Is is normal these days? it 
didn't used to be the case few months ago. Is there anything i can turn off to 
start faster?

Thanks,
Klaus.
-- 
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Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

Brian - you're the man! :)

/Peter


-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 1 juni 2009 16:02
Till: freeswitch-users@lists.freeswitch.org
Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk 
(r13502)?

I made the PAI in the 18X and 200 be present ONLY if you set  
callee_id_name.  Then I fixed the update on transfer to only work on  
snom and polycom along with uuid_display and uuid_hold will only wend  
the display info for snom and polycom till I have others test and  
provide me the confirmation those work with other phones too.

/b

On Jun 1, 2009, at 8:51 AM, Kristian Kielhofner wrote:

> Ahh...  That's what I get for not reading the entire thread!
>
> On Mon, Jun 1, 2009 at 9:32 AM, Brian West   
> wrote:
>> No he's talking about the one from SFSIP-111, Again that shouldn't
>> matter... I'll fix that.
>>
>> /b
>>
>
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
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Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

Thanks for the reply!

Will this really help though? From what I understand of the change that breaks 
the compatibility, it will always send this header in the "200 OK" message (my 
problem is incoming calls to FS). The fix you're describing, isn't it when 
calling from FS to the other end? This way it works either way, it's just when 
the PBX gets this in the 200 OK message with this header that it stops working.

/Peter

-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Kristian Kielhofner
Skickat: den 1 juni 2009 15:20
Till: freeswitch-users@lists.freeswitch.org
Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk 
(r13502)?

Peter,

  Ouch.  Your PBX is broken.  It shouldn't do that.

  Luckily FreeSWITCH provides a way to select RPID/PAI/none:

http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation

On Mon, Jun 1, 2009 at 5:57 AM, Peter Olsson
 wrote:
> Here is an update for this issue (SFSIP-149). I've raised a jira case for 
> this. It was not a RTP problem, the problem is caused by the new 
> "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX 
> not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS 
> still thinks it's answered though, and starts to send RTP data to the other 
> end.
>
> /Peter
>

-- 
Kristian Kielhofner
http://www.astlinux.org
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Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

Here is an update for this issue (SFSIP-149). I've raised a jira case for this. 
It was not a RTP problem, the problem is caused by the new 
"P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not 
to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still 
thinks it's answered though, and starts to send RTP data to the other end.

/Peter


-Ursprungligt meddelande-
Från: Peter Olsson 
Skickat: den 30 maj 2009 09:01
Till: freeswitch-users@lists.freeswitch.org
Ämne: FW: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

And just to be clear, even though media flows in one direction (from FS to 
phone), I get no audio at all.

And by the way, I mean SVN, not SNV :) Sorry for double posting...

/Peter

________
Från: Peter Olsson
Skickat: den 30 maj 2009 08:58
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

I'll try to do this during this weekend.

I've looked through the SNV logs, and I really can't find a good reason for 
this to happen. And when looking into wireshark I can see RTP audio flowing 
from FS to my SIP phone, but not in the other direction. So this still makes me 
wonder if something has happened to sofia (that sets up the media 
incorrectly)... And also when I hangup the call, it takes about a minute for FS 
to detect this, and it reports hangup reason unknown.

But as I said, I'll look into this a bit deeper during this weekend, and file a 
jira case when I have some more information.


//Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[freeswitch-users-boun...@lists.freeswitch.org] för Michael Jerris 
[m...@jerris.com]
Skickat: den 29 maj 2009 19:35
Till: freeswitch-users@lists.freeswitch.org
Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk 
(r13502)?

Can you try to do a binary search and nail down the exact version that caused 
this issue and then file a bug on http://jira.freeswitch.org.

Thanks
Mike

On May 29, 2009, at 9:55 AM, Peter Olsson wrote:

I'm on Windows, so I have everything under my fs directory, but I deleted the 
complete directory and did everything from scratch...

/Peter


Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 29 maj 2009 15:46
Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ?

On Fri, May 29, 2009 at 8:33 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

Nope - it's not :)

Just to make sure I even deleted the source completely, and checked everything 
out again.

Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 
[mailto:freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>]
 För Brian West
Skickat: den 29 maj 2009 15:26

Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Nope its not a sofia issue... its build skew ;)

 On May 29, 2009, at 8:24 AM, Peter Olsson wrote:

I've looked into this a bit more now, and I think it is a sofia issue, I will 
look trough the changes in sofia since I had the last working configuration, 
and see if I find anything.

/Peter


!DSPAM:4a201fa932931035648682!

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Re: [Freeswitch-users] I have problem in compiling freeswitch with mod_opal

[Peter Olsson]

There are some problems with latest trunk of opal, it's not compatible with FS 
anymore. Read more info on jira case MODOPAL-10.

I recommend using an older revision of opal/ptlib (I'm testing r22623 right 
now). It also works better then using the stable versions of opal/ptlib.

/Peter

Från: freeswitch-users-boun...@lists.freeswitch.org 
[freeswitch-users-boun...@lists.freeswitch.org] för Sadjad Seyed-Ahmadian 
[ssa1...@yahoo.com]
Skickat: den 30 maj 2009 09:32
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] I have problem in compiling freeswitch with
mod_opal

Dear Brian,

I had compile it with SVN version and got same error.

Sincerely,
Sadajd

**
Message: 3
Date: Wed, 27 May 2009 08:37:23 -0500
From: Brian West mailto:br...@freeswitch.org>>
Subject: Re: [Freeswitch-users] I have problem in compiling freeswitch
withmod_opal
To: 
freeswitch-users@lists.freeswitch.org
Message-ID: 
mailto:db76f9f4-a60e-4a53-a658-5cdc14d80...@freeswitch.org>>
Content-Type: text/plain; charset="us-ascii"

You have to use the SVN version of both ptlib and OPAL and it will
compile.

/b

On May 27, 2009, at 3:26 AM, Sadjad Seyed-Ahmadian wrote:

> I faced a problem when I want to compile freeswitch with mod_opal.
> It gives me a compilation error like bellow
>
> I used  ptlib-2.6.2 and  opal-3.6.2.
>
> Would someone please help me?
>
> Sincerely,
> Sadjad

Brian West
br...@freeswitch.org

***

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[Freeswitch-users] FW: Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

And just to be clear, even though media flows in one direction (from FS to 
phone), I get no audio at all.

And by the way, I mean SVN, not SNV :) Sorry for double posting...

/Peter


Från: Peter Olsson
Skickat: den 30 maj 2009 08:58
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

I'll try to do this during this weekend.

I've looked through the SNV logs, and I really can't find a good reason for 
this to happen. And when looking into wireshark I can see RTP audio flowing 
from FS to my SIP phone, but not in the other direction. So this still makes me 
wonder if something has happened to sofia (that sets up the media 
incorrectly)... And also when I hangup the call, it takes about a minute for FS 
to detect this, and it reports hangup reason unknown.

But as I said, I'll look into this a bit deeper during this weekend, and file a 
jira case when I have some more information.


//Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[freeswitch-users-boun...@lists.freeswitch.org] för Michael Jerris 
[m...@jerris.com]
Skickat: den 29 maj 2009 19:35
Till: freeswitch-users@lists.freeswitch.org
Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk 
(r13502)?

Can you try to do a binary search and nail down the exact version that caused 
this issue and then file a bug on http://jira.freeswitch.org.

Thanks
Mike

On May 29, 2009, at 9:55 AM, Peter Olsson wrote:

I’m on Windows, so I have everything under my fs directory, but I deleted the 
complete directory and did everything from scratch...

/Peter


Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 29 maj 2009 15:46
Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ?

On Fri, May 29, 2009 at 8:33 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

Nope – it’s not :)

Just to make sure I even deleted the source completely, and checked everything 
out again.

Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 
[mailto:freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>]
 För Brian West
Skickat: den 29 maj 2009 15:26

Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Nope its not a sofia issue... its build skew ;)

 On May 29, 2009, at 8:24 AM, Peter Olsson wrote:

I've looked into this a bit more now, and I think it is a sofia issue, I will 
look trough the changes in sofia since I had the last working configuration, 
and see if I find anything.

/Peter


!DSPAM:4a201fa932931035648682!

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Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

I'll try to do this during this weekend.

I've looked through the SNV logs, and I really can't find a good reason for 
this to happen. And when looking into wireshark I can see RTP audio flowing 
from FS to my SIP phone, but not in the other direction. So this still makes me 
wonder if something has happened to sofia (that sets up the media 
incorrectly)... And also when I hangup the call, it takes about a minute for FS 
to detect this, and it reports hangup reason unknown.

But as I said, I'll look into this a bit deeper during this weekend, and file a 
jira case when I have some more information.


//Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[freeswitch-users-boun...@lists.freeswitch.org] för Michael Jerris 
[m...@jerris.com]
Skickat: den 29 maj 2009 19:35
Till: freeswitch-users@lists.freeswitch.org
Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk 
(r13502)?

Can you try to do a binary search and nail down the exact version that caused 
this issue and then file a bug on http://jira.freeswitch.org.

Thanks
Mike

On May 29, 2009, at 9:55 AM, Peter Olsson wrote:

I’m on Windows, so I have everything under my fs directory, but I deleted the 
complete directory and did everything from scratch...

/Peter


Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 29 maj 2009 15:46
Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ?

On Fri, May 29, 2009 at 8:33 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

Nope – it’s not :)

Just to make sure I even deleted the source completely, and checked everything 
out again.

Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 
[mailto:freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>]
 För Brian West
Skickat: den 29 maj 2009 15:26

Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Nope its not a sofia issue... its build skew ;)

 On May 29, 2009, at 8:24 AM, Peter Olsson wrote:

I've looked into this a bit more now, and I think it is a sofia issue, I will 
look trough the changes in sofia since I had the last working configuration, 
and see if I find anything.

/Peter


!DSPAM:4a201fa932931035648682!

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Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

I'm on Windows, so I have everything under my fs directory, but I deleted the 
complete directory and did everything from scratch...

/Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 29 maj 2009 15:46
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ?

On Fri, May 29, 2009 at 8:33 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

Nope - it's not :)



Just to make sure I even deleted the source completely, and checked everything 
out again.



/Peter





Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 
[mailto:freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>]
 För Brian West
Skickat: den 29 maj 2009 15:26

Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?



Nope its not a sofia issue... its build skew ;)



/b



On May 29, 2009, at 8:24 AM, Peter Olsson wrote:



I've looked into this a bit more now, and I think it is a sofia issue, I will 
look trough the changes in sofia since I had the last working configuration, 
and see if I find anything.

/Peter



Brian West

br...@freeswitch.org<mailto:br...@freeswitch.org>



-- Meet us at ClueCon!  http://www.cluecon.com<http://www.cluecon.com/>









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--
Anthony Minessale II

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ClueCon http://www.cluecon.com/

AIM: anthm
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Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

Actually I deleted everything from disk and downloaded a fresh clean copy from 
SVN and rebuilt it from scratch. I should mention that I'm on windows, so I 
never do "make current". I just do a full clean, get latest from SVN and 
rebuild, that's what I do every time. But for this time I even deleted 
everything from disk and did a fresh rebuild - but nothing helped..

/Peter

Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 29 maj 2009 15:42
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

when you say "i did that"
you typed "make current" to rebuild?

or you are assuming your successful compile is the same effect as cleaning the 
100 object files
that have the wrong data structure in them so the audio data they really seek 
is 8 bytes offset from where they think they are
until they are deleted and recompiled by the make current command?

The coincidental side-effect of this is no audio in any rtp streams.. 
*shrug* or you can continue to update more and more revisions
on top of each other and end up with even worse build skew.

We tried 3 times to tell you that's plenty......


On Fri, May 29, 2009 at 8:28 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

I did that, and it compiles fine. It's just not working :) But as I said in my 
last post, I think it could also be related to sofia, when using h323 it 
works... However - maybe I'm using opal's RTP stream by then..?



I'll get some logs for the scenario, and if I don't find a solution I'll start 
a new issue on jira.



/Peter



Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 
[mailto:freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>]
 För Brian West
Skickat: den 29 maj 2009 15:17

Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?



Its called build skew... we added an extra_data element to the frame struct.  
Please do a fresh checkout and build.



/b



On May 29, 2009, at 5:46 AM, Jason White wrote:



Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

After using the latest trunk revisions I get no audio anymore. The last

working build I have is about 5 days ago. I havn't upgraded until today, so

I don't know exactly when this happened.

You could always check out some intermediate revisions, compile them, and see
if they work. This will enable you to find out exactly which revision
introduced the problem.

The correct way to do this is as a binary search. In the git revision control
system, this is partially automated as the bisect command. Even though
FreeSWITCH uses Subversion rather than Git, you can still perform the
bisect manually by checking out particular revisions by number.
svn update -r 
should do it.


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Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

Nope - it's not :)

Just to make sure I even deleted the source completely, and checked everything 
out again.

/Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 29 maj 2009 15:26
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Nope its not a sofia issue... its build skew ;)

/b

On May 29, 2009, at 8:24 AM, Peter Olsson wrote:


I've looked into this a bit more now, and I think it is a sofia issue, I will 
look trough the changes in sofia since I had the last working configuration, 
and see if I find anything.

/Peter

Brian West
br...@freeswitch.org<mailto:br...@freeswitch.org>

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Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

I did that, and it compiles fine. It's just not working :) But as I said in my 
last post, I think it could also be related to sofia, when using h323 it 
works... However - maybe I'm using opal's RTP stream by then..?

I'll get some logs for the scenario, and if I don't find a solution I'll start 
a new issue on jira.

/Peter

Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 29 maj 2009 15:17
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Its called build skew... we added an extra_data element to the frame struct.  
Please do a fresh checkout and build.

/b

On May 29, 2009, at 5:46 AM, Jason White wrote:


Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

After using the latest trunk revisions I get no audio anymore. The last
working build I have is about 5 days ago. I havn't upgraded until today, so
I don't know exactly when this happened.

You could always check out some intermediate revisions, compile them, and see
if they work. This will enable you to find out exactly which revision
introduced the problem.

The correct way to do this is as a binary search. In the git revision control
system, this is partially automated as the bisect command. Even though
FreeSWITCH uses Subversion rather than Git, you can still perform the
bisect manually by checking out particular revisions by number.
svn update -r 
should do it.


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Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

I've looked into this a bit more now, and I think it is a sofia issue, I will 
look trough the changes in sofia since I had the last working configuration, 
and see if I find anything.

/Peter

-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Jason White
Skickat: den 29 maj 2009 12:47
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Peter Olsson  wrote:
> After using the latest trunk revisions I get no audio anymore. The last
> working build I have is about 5 days ago. I havn't upgraded until today, so
> I don't know exactly when this happened.

You could always check out some intermediate revisions, compile them, and see
if they work. This will enable you to find out exactly which revision
introduced the problem.

The correct way to do this is as a binary search. In the git revision control
system, this is partially automated as the bisect command. Even though
FreeSWITCH uses Subversion rather than Git, you can still perform the
bisect manually by checking out particular revisions by number.
svn update -r 
should do it.


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[Freeswitch-users] FW: Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

Sorry for missing this in my last post, but I'm using sofia for all calls.

/Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Peter Olsson
Skickat: den 29 maj 2009 12:31
Till: 'freeswitch-users@lists.freeswitch.org'
Ämne: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

After using the latest trunk revisions I get no audio anymore. The last working 
build I have is about 5 days ago. I havn't upgraded until today, so I don't 
know exactly when this happened.

I've noticed quite a few changes on the RTP stack, beacuse of the 
implementation om ZRTP, and I guess it's somewhere around this time when it 
happened. How to continue debugging on this issue? I have both a working 
version of FS (compiled 5 days ago), and a broken one (compiled today), so I 
can test this very easily, and everything is on a non live server.

The conf-dir is the same between the revisions.

The calls I'm trying to do is both directly to FS (voicemail or similar 
applications), and aslo calls to another SIP-trunk, to PSTN (media stream is 
sent through FS).

Regards,

Peter

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[Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

After using the latest trunk revisions I get no audio anymore. The last working 
build I have is about 5 days ago. I havn't upgraded until today, so I don't 
know exactly when this happened.

I've noticed quite a few changes on the RTP stack, beacuse of the 
implementation om ZRTP, and I guess it's somewhere around this time when it 
happened. How to continue debugging on this issue? I have both a working 
version of FS (compiled 5 days ago), and a broken one (compiled today), so I 
can test this very easily, and everything is on a non live server.

The conf-dir is the same between the revisions.

The calls I'm trying to do is both directly to FS (voicemail or similar 
applications), and aslo calls to another SIP-trunk, to PSTN (media stream is 
sent through FS).

Regards,

Peter

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Re: [Freeswitch-users] Audio "clicks" between playback of audio files

[Peter Olsson]

Brian,

After converting all the files to .wav it works great. And it also solved my 
issue for recording messages.

This is good enough for me - even though I guess there might be some issues 
with native files i FreeSWITCH.

Thanks for all your help!

/Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 11 maj 2009 16:24
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Audio "clicks" between playback of audio files

I'm not sure.  Can you provide me a test file and a known case that you can 
produce this issue with?

/b

On May 11, 2009, at 9:20 AM, Peter Olsson wrote:


Brian,

Thanks for the response. No, I didn't try wav files - and I'd prefer to keep 
the current codec if that's possible. But I could give it a try and see what 
happens.

Do you think it might only be related to the native files in FS?

//Peter

Brian West
br...@freeswitch.org<mailto:br...@freeswitch.org>

-- Meet us at ClueCon!  http://www.cluecon.com<http://www.cluecon.com/>




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Re: [Freeswitch-users] Audio "clicks" between playback of audio files

[Peter Olsson]

In this case I'm using Avaya Phones connected to a Avaya CM PBX, which talks 
SIP to FreeSWITCH. But I'll try to connect a SIP phone directly as well - to 
see if it makes any difference. I Have a Polycom IP550 I can use for some 
testing.

I've used the exact same setup using yate, and that worked fine, but I think 
one difference is that it sends rtp even when not playing files, so that's why 
I thought that could be an issue.

/Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 11 maj 2009 16:43
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Audio "clicks" between playback of audio files

What phone are you using?

/b

On May 11, 2009, at 9:40 AM, Peter Olsson wrote:


The problem is that it's not 100% reproducable. Sometimes I can play 2-3-4 
files, and it sounds great, sometimes it "clicks" louder, somtimes not so much. 
I could get a Wireshark dump for you, could that help?

Regards,

Peter


Brian West
br...@freeswitch.org<mailto:br...@freeswitch.org>

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Re: [Freeswitch-users] Audi record using uuid_record

[Peter Olsson]

I've confirmed that recording to wav files works just fine. So it seems to be 
somthing strange with the native files in FS?

/Peter


Från: Peter Olsson
Skickat: den 11 maj 2009 16:16
Till: 'freeswitch-users@lists.freeswitch.org'
Ämne: RE: [Freeswitch-users] Audi record using uuid_record

In this case it is pure PCMA, all the way from the phone. I just dial in to 
number 2100 (using SIP, codec PCMA), and then I have a event socket connected 
that sees the ivr-test flag. I then play some files (PCMA), and then start a 
recording.


  
  
  
  
  


I understand that it works with wav, however, the application I'm working on 
already exists, and it makes lots of trouble for me to change the file format 
during the recording - I have lots of other parts of the software that needs to 
be changed as well. I've used yate for this application before this - so 
everything does exist, I'm just porting it to FreeSWITCH.

//Peter

Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 11 maj 2009 15:58
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Audi record using uuid_record

Chances are if both legs are NOT already alaw you'll need to record it with 
.wav or .al files.  .PCMU or .PCMA are native file formats if any transcoding 
is taking place you probably can't get way with .PCMA.

/b

On May 11, 2009, at 8:39 AM, Peter Olsson wrote:

Hello again,

I also have a problem when I try to record messages. I record to .PCMA-files, 
and the file is created perfectly. But it's just distorted audio in it. It 
sounds to me that there might be a codec issue. The media stream is PCMA all 
the way from the phone to FreeSWITCH, and to start recording I simply call 
"uuid_record UUID start c:\test.PCMA".

According to the docs the file should automatically be recorded as PCMA when 
the file is named .PCMA.

Any ideas what I can be doing wrong?

Regards,

Peter
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Re: [Freeswitch-users] Audio "clicks" between playback of audio files

[Peter Olsson]

The problem is that it's not 100% reproducable. Sometimes I can play 2-3-4 
files, and it sounds great, sometimes it "clicks" louder, somtimes not so much. 
I could get a Wireshark dump for you, could that help?

Regards,

Peter

Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 11 maj 2009 16:24
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Audio "clicks" between playback of audio files

I'm not sure.  Can you provide me a test file and a known case that you can 
produce this issue with?

/b

On May 11, 2009, at 9:20 AM, Peter Olsson wrote:


Brian,

Thanks for the response. No, I didn't try wav files - and I'd prefer to keep 
the current codec if that's possible. But I could give it a try and see what 
happens.

Do you think it might only be related to the native files in FS?

//Peter

Brian West
br...@freeswitch.org<mailto:br...@freeswitch.org>

-- Meet us at ClueCon!  http://www.cluecon.com<http://www.cluecon.com/>




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Re: [Freeswitch-users] Audio "clicks" between playback of audio files

[Peter Olsson]

Brian,

Thanks for the response. No, I didn't try wav files - and I'd prefer to keep 
the current codec if that's possible. But I could give it a try and see what 
happens.

Do you think it might only be related to the native files in FS?

//Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 11 maj 2009 16:09
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Audio "clicks" between playback of audio files

Have you tried wav files?

/b

On May 11, 2009, at 8:34 AM, Peter Olsson wrote:


I'm implementing an IVR solution for FreeSWITCH, but I have a little problem 
with audio playback. I'm just calling into application park, and then handle 
the flow using the event socket.

All my audio phrases are .PCMA (8KHz a-law), and I play lots  of files after 
eachother. Between each file I can sometimes here a little "click", even though 
I'm 100% sure that this is not from my files. My guess is that it might be 
caused of the fact that no RTP is sent at all when the phrase is not playing. 
If I merge the files together before playing them it sounds just fine.

Is it possible to make FreeSWITCH send silence frames, even when not needed? I 
know this is a waste of resources, but it will still make the solution sound 
much better.

Regards

Peter Olsson

Brian West
br...@freeswitch.org<mailto:br...@freeswitch.org>

-- Meet us at ClueCon!  http://www.cluecon.com<http://www.cluecon.com/>




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Re: [Freeswitch-users] Audi record using uuid_record

[Peter Olsson]

In this case it is pure PCMA, all the way from the phone. I just dial in to 
number 2100 (using SIP, codec PCMA), and then I have a event socket connected 
that sees the ivr-test flag. I then play some files (PCMA), and then start a 
recording.


  
  
  
  
  


I understand that it works with wav, however, the application I'm working on 
already exists, and it makes lots of trouble for me to change the file format 
during the recording - I have lots of other parts of the software that needs to 
be changed as well. I've used yate for this application before this - so 
everything does exist, I'm just porting it to FreeSWITCH.

//Peter

Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 11 maj 2009 15:58
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Audi record using uuid_record

Chances are if both legs are NOT already alaw you'll need to record it with 
.wav or .al files.  .PCMU or .PCMA are native file formats if any transcoding 
is taking place you probably can't get way with .PCMA.

/b

On May 11, 2009, at 8:39 AM, Peter Olsson wrote:


Hello again,

I also have a problem when I try to record messages. I record to .PCMA-files, 
and the file is created perfectly. But it's just distorted audio in it. It 
sounds to me that there might be a codec issue. The media stream is PCMA all 
the way from the phone to FreeSWITCH, and to start recording I simply call 
"uuid_record UUID start c:\test.PCMA".

According to the docs the file should automatically be recorded as PCMA when 
the file is named .PCMA.

Any ideas what I can be doing wrong?

Regards,

Peter
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[Freeswitch-users] Audi record using uuid_record

[Peter Olsson]

Hello again,

I also have a problem when I try to record messages. I record to .PCMA-files, 
and the file is created perfectly. But it's just distorted audio in it. It 
sounds to me that there might be a codec issue. The media stream is PCMA all 
the way from the phone to FreeSWITCH, and to start recording I simply call 
"uuid_record UUID start c:\test.PCMA".

According to the docs the file should automatically be recorded as PCMA when 
the file is named .PCMA.

Any ideas what I can be doing wrong?

Regards,

Peter
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[Freeswitch-users] Audio "clicks" between playback of audio files

[Peter Olsson]

I'm implementing an IVR solution for FreeSWITCH, but I have a little problem 
with audio playback. I'm just calling into application park, and then handle 
the flow using the event socket.

All my audio phrases are .PCMA (8KHz a-law), and I play lots  of files after 
eachother. Between each file I can sometimes here a little "click", even though 
I'm 100% sure that this is not from my files. My guess is that it might be 
caused of the fact that no RTP is sent at all when the phrase is not playing. 
If I merge the files together before playing them it sounds just fine.

Is it possible to make FreeSWITCH send silence frames, even when not needed? I 
know this is a waste of resources, but it will still make the solution sound 
much better.

Regards

Peter Olsson

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Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

[Peter Olsson]

I've added this as jira case http://jira.freeswitch.org/browse/MODSOFIA-4

I wasn't sure if it should be under mod_sofia or sofia-sip.

The report has a full debug log attached.

Regards,

Peter Olsson

Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 16 april 2009 14:23
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the 
call?

yes open a jira http://jira.freeswitch.org

*attach* the following (do not paste it inline into the comments and give all 
trace files a .txt extension)

repeat the trace you did earlier with more debugging enabled.
 type these 3 cli commands before you call
 sofia profile internal siptrace on
 sofia loglevel all 9
 console loglevel debug




On Thu, Apr 16, 2009 at 2:13 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

Allright, I tried this again now, with revision 13042 - it's the same result as 
before.. Should I file a jira case for this?



If you want any more information, or more traces, please get back to me, and 
I'll try to help out as much as possible.





Peter





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 För Brian West
Skickat: den 15 april 2009 23:21

Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds 
the call?



What port are you hitting?  Make sure you turn sip tracing on external and 
internal just in case you're using either or both.



/b



On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote:



I've built using latest trunk now, but I won't be able to test again until 
tomorrow - I'll get back to you after that.

Just to make the scenario a bit more clear;
The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP 
Enablement Services), this one talks UDP to FreeSWITCH. Could this be something 
that causes the problem? I also tried to dial into the dialplan, answer the 
call, and then try to deflect the call using REFER. This didn't create any SIP 
messages either (and nothing happened with the call), so it seems there might 
be a bigger issue than just BYE.

Peter



Brian West

br...@freeswitch.org<mailto:br...@freeswitch.org>



-- Meet us at ClueCon!  http://www.cluecon.com<http://www.cluecon.com/>









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Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

[Peter Olsson]

Allright, I tried this again now, with revision 13042 - it's the same result as 
before.. Should I file a jira case for this?

If you want any more information, or more traces, please get back to me, and 
I'll try to help out as much as possible.


Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 15 april 2009 23:21
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds 
the call?

What port are you hitting?  Make sure you turn sip tracing on external and 
internal just in case you're using either or both.

/b

On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote:


I've built using latest trunk now, but I won't be able to test again until 
tomorrow - I'll get back to you after that.

Just to make the scenario a bit more clear;
The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP 
Enablement Services), this one talks UDP to FreeSWITCH. Could this be something 
that causes the problem? I also tried to dial into the dialplan, answer the 
call, and then try to deflect the call using REFER. This didn't create any SIP 
messages either (and nothing happened with the call), so it seems there might 
be a bigger issue than just BYE.

Peter

Brian West
br...@freeswitch.org<mailto:br...@freeswitch.org>

-- Meet us at ClueCon!  http://www.cluecon.com<http://www.cluecon.com/>




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Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call?

[Peter Olsson]

It's standard port 5060 UDP at both ends. I'll enable tracing on the external 
profile as well - but I'm quite sure it's not used at all in this case.

I'm on my way to the lab now, so I'll soon get back with my results after using 
latest SVN revision (my 'old' revision was just about 2 days old though).

Peter



Från: Brian West 
Skickat: den 15 april 2009 23:29
Till: freeswitch-users@lists.freeswitch.org 

Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds 
the call?

What port are you hitting?  Make sure you turn sip tracing on external and 
internal just in case you're using either or both.

/b

On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote:

I've built using latest trunk now, but I won't be able to test again until 
tomorrow - I'll get back to you after that.

Just to make the scenario a bit more clear;
The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP 
Enablement Services), this one talks UDP to FreeSWITCH. Could this be something 
that causes the problem? I also tried to dial into the dialplan, answer the 
call, and then try to deflect the call using REFER. This didn't create any SIP 
messages either (and nothing happened with the call), so it seems there might 
be a bigger issue than just BYE.

Peter

Brian West
br...@freeswitch.org<mailto:br...@freeswitch.org>

-- Meet us at ClueCon!  http://www.cluecon.com<http://www.cluecon.com/>




!DSPAM:49e651b332933023977319!

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Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call?

[Peter Olsson]

I've built using latest trunk now, but I won't be able to test again until 
tomorrow - I'll get back to you after that.

Just to make the scenario a bit more clear;
The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP 
Enablement Services), this one talks UDP to FreeSWITCH. Could this be something 
that causes the problem? I also tried to dial into the dialplan, answer the 
call, and then try to deflect the call using REFER. This didn't create any SIP 
messages either (and nothing happened with the call), so it seems there might 
be a bigger issue than just BYE.

Peter


On 09-04-15 20.04, "Brian West"  wrote:

I recall this and transport=tls in this case... maybe MikeJ can chime in on 
this one..I thought we already fixed this.

On Apr 15, 2009, at 12:50 PM, Peter Olsson wrote:

 Record-Route: 


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com 
<http://www.cluecon.com/><http://www.cluecon.com/>





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[Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call?

[Peter Olsson]

My current revision is r13015. I will do an update as soon as possible and see 
if that solves the issue.

Thanks!

//Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[freeswitch-users-boun...@lists.freeswitch.org] för Anthony Minessale 
[anthony.miness...@gmail.com]
Skickat: den 15 april 2009 18:46
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the 
call?

This sounds familiar:

What revision of the code is this?
Can you confirm you have this problem with SVN trunk (r13034 at the time of 
this writing).


On Wed, Apr 15, 2009 at 11:24 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

This is the full SIP-trace for the call. It’s not sending a BYE at all, and I 
can’t see one in Wireshark either. As you can see in the end there is a call to 
hangup_function(), but no SIP messages after that. When I manually hangup the 
phone I can see it sends BYE to FreeSWITCH (which is quite expected, since it 
thinks the call still exists), and FreeSWITCH just answers ”481 Call Does Not 
Exist” – which of course is also expected, since the call was dropped.



recv 1255 bytes from udp/[192.168.94.53]:32769 at 16:17:57.853727:

   

   INVITE sip:2...@192.168.1.155:5060;lr SIP/2.0

   Accept-Language: en

   Call-ID: 80948a675733de14449f79df00

   CSeq: 1 INVITE

   From: "Peter Olsson" 
http://sip:1...@sip.se:6001>>;tag=80948a675733de13449f79df00

   Record-Route: 
,

   To: "2100" mailto:sip%3a2...@192.168.94.53>>

   Via: SIP/2.0/UDP 
192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 
192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00

   Content-Length: 165

   Content-Type: application/sdp

   Contact: "Peter Olsson" 

   Max-Forwards: 67

   User-Agent: Avaya CM/R015x.01.1.415.1

   Allow: 
INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH

   Supported: 100rel,timer,replaces,join,histinfo

   Alert-Info: 
;avaya-cm-alert-type=internal

   Min-SE: 1200

   Session-Expires: 1200;refresher=uac

   P-Asserted-Identity: "Peter Olsson" 
http://sip:1...@sip.se:6001>>

   History-Info: 
mailto:sip%3a2...@192.168.94.53>>;index=1,"2100" 
mailto:sip%3a2...@192.168.94.53>>;index=1.1



   v=0

   o=- 1 1 IN IP4 192.168.94.53

   s=-

   c=IN IP4 192.168.94.59

   b=AS:64

   t=0 0

   m=audio 2062 RTP/AVP 8 127

   a=rtpmap:8 PCMA/8000

   a=rtpmap:127 telephone-event/8000

   

send 541 bytes to udp/[192.168.94.53]:5060 at 16:17:57.854727:

   

   SIP/2.0 100 Trying

   Via: SIP/2.0/UDP 
192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 
192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00

   Record-Route: 

   Record-Route: 

   From: "Peter Olsson" 
http://sip:1...@sip.se:6001>>;tag=80948a675733de13449f79df00

   To: "2100" mailto:sip%3a2...@192.168.94.53>>

   Call-ID: 80948a675733de14449f79df00

   CSeq: 1 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN

   Content-Length: 0



   

2009-04-15 18:17:57 [NOTICE] switch_channel.c:597 switch_channel_set_name() 
NewChannel sofia/internal/1...@sip.se:6001<http://1...@sip.se:6001> 
[fa1c328e-bdfe-7d49-ab6f-dc9ec791c455]

2009-04-15 18:17:57 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 
Peter Olsson->2100 in context public

2009-04-15 18:17:57 [NOTICE] mod_dptools.c:649 answer_function() Channel 
[sofia/internal/1...@sip.se:6001<http://1...@sip.se:6001>] has been answered

send 1322 bytes to udp/[192.168.94.53]:5060 at 16:17:57.871727:

   

   SIP/2.0 200 OK

   Via: SIP/2.0/UDP 
192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 
192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00

   Record-Route: 

   Record-Route: 

   From: "Peter Olsson" 
http://sip:1...@sip.se:6001>>;tag=80948a675733de13449f79df00

   To: "2100" 
mailto:sip%3a2...@192.168.94.53>>;tag=Sv6KrDv9vQrer

   Call-ID: 80948a675733de14449f79df00

   CSeq: 1 INVITE

   Contact: 

   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN

   Accept: application/sdp

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, 
REFER, UPDATE, REGISTER, INFO, PUBLISH

   Require: timer

   Supported: timer, precondition, path, replaces

   Allow-Events: talk, presence, dialog, call-info, sla, 
include-session-descriptio

Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

[Peter Olsson]

This is the full SIP-trace for the call. It's not sending a BYE at all, and I 
can't see one in Wireshark either. As you can see in the end there is a call to 
hangup_function(), but no SIP messages after that. When I manually hangup the 
phone I can see it sends BYE to FreeSWITCH (which is quite expected, since it 
thinks the call still exists), and FreeSWITCH just answers "481 Call Does Not 
Exist" - which of course is also expected, since the call was dropped.

recv 1255 bytes from udp/[192.168.94.53]:32769 at 16:17:57.853727:
   
   INVITE sip:2...@192.168.1.155:5060;lr SIP/2.0
   Accept-Language: en
   Call-ID: 80948a675733de14449f79df00
   CSeq: 1 INVITE
   From: "Peter Olsson" ;tag=80948a675733de13449f79df00
   Record-Route: 
,
   To: "2100" 
   Via: SIP/2.0/UDP 
192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 
192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00
   Content-Length: 165
   Content-Type: application/sdp
   Contact: "Peter Olsson" 
   Max-Forwards: 67
   User-Agent: Avaya CM/R015x.01.1.415.1
   Allow: 
INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH
   Supported: 100rel,timer,replaces,join,histinfo
   Alert-Info: 
;avaya-cm-alert-type=internal
   Min-SE: 1200
   Session-Expires: 1200;refresher=uac
   P-Asserted-Identity: "Peter Olsson" 
   History-Info: ;index=1,"2100" 
;index=1.1

   v=0
   o=- 1 1 IN IP4 192.168.94.53
   s=-
   c=IN IP4 192.168.94.59
   b=AS:64
   t=0 0
   m=audio 2062 RTP/AVP 8 127
   a=rtpmap:8 PCMA/8000
   a=rtpmap:127 telephone-event/8000
   
send 541 bytes to udp/[192.168.94.53]:5060 at 16:17:57.854727:
   
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 
192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 
192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00
   Record-Route: 
   Record-Route: 
   From: "Peter Olsson" ;tag=80948a675733de13449f79df00
   To: "2100" 
   Call-ID: 80948a675733de14449f79df00
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Content-Length: 0

   
2009-04-15 18:17:57 [NOTICE] switch_channel.c:597 switch_channel_set_name() 
NewChannel sofia/internal/1...@sip.se:6001 
[fa1c328e-bdfe-7d49-ab6f-dc9ec791c455]
2009-04-15 18:17:57 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 
Peter Olsson->2100 in context public
2009-04-15 18:17:57 [NOTICE] mod_dptools.c:649 answer_function() Channel 
[sofia/internal/1...@sip.se:6001] has been answered
send 1322 bytes to udp/[192.168.94.53]:5060 at 16:17:57.871727:
   
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 
192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 
192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00
   Record-Route: 
   Record-Route: 
   From: "Peter Olsson" ;tag=80948a675733de13449f79df00
   To: "2100" ;tag=Sv6KrDv9vQrer
   Call-ID: 80948a675733de14449f79df00
   CSeq: 1 INVITE
   Contact: 
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, 
REFER, UPDATE, REGISTER, INFO, PUBLISH
   Require: timer
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, 
include-session-description, presence.winfo, message-summary, refer
   Session-Expires: 1200;refresher=uac
   Min-SE: 1200
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 265

   v=0
   o=FreeSWITCH 484797194364394181 220756314446402535 IN IP4 192.168.1.155
   s=FreeSWITCH
   c=IN IP4 192.168.1.155
   t=0 0
   m=audio 23574 RTP/AVP 8 127
   a=rtpmap:8 PCMA/8000
   a=rtpmap:127 telephone-event/8000
   a=fmtp:127 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   
recv 521 bytes from udp/[192.168.94.53]:32769 at 16:17:57.880727:
   ----
   ACK sip:mod_so...@192.168.1.155:5060;transport=udp SIP/2.0
   From: "Peter Olsson" ;tag=80948a675733de13449f79df00
   To: "2100" ;tag=Sv6KrDv9vQrer
   Call-ID: 80948a675733de14449f79df00
   CSeq: 1 ACK
   Max-Forwards: 69
   Via: SIP/2.0/UDP 
192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.1,SIP/2.0/TLS 
192.168.94.53:6001;psrrposn=1;branch=z9hG4bK80948a675733de16449f79df00

   User-Agent: Avaya CM/R015x.01.1.415.1
   Content-Length: 0
   Record-Route: 

   --

[Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

[Peter Olsson]

When I do a call from my Avaya SIP Server to FreeSWITCH. And then let 
FreeSWITCH do a hangup of the call, FreeSWITCH doesn't seem to send a "BYE" 
back to the Avaya PBX. I've narrowed it down to this simple example in the 
dialplan;

  
  
  

In this case no BYE is issued, and the phone still thinks the call is alive. If 
you want to I could send the SIP headers as well for this scenario..

Regards,

Peter Olsson
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Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...

[Peter Olsson]

Allright - last question :) I'll try to be a little more specific. Lets say I 
whant  to do the following;


1.   Dial into FreeSWITCH, to some kind of application (javascript or 
whatever).

2.   Answer that call, and let the user choose what to do; 1: record 
message, 2: transfer to XXX etc. The user presses 2.

3.   I don't want to release the first call leg yet, since I need to be 
really sure that 2 is reachable (or else I will give the user choices again, 
with som kind of "the call could not be transferred"). So lets say I play some 
music for the user while trying to connect the call.

4.   I originate another call - now I understand I have two choices, either 
I originate directly to a SIP phone (sofia/internal...), or I let the dialplan 
do the work - and if I want the dialplan to be the one to transfer the call 
somewhere (maybe to the same extension), I must use loopback - right?

5.   If the new call answers, bridge the two calls, if it fails, start over 
again, after reading an error message.

Whould this also be possible with transfer? If I understand everything right I 
loose control of the call, and won't be able to handle the failed transfer? Or 
is it possible to solve in a better way?

What I guess I'd really want to do is to ask the dialplan "hey, I want to dial 
 - give me the full sofia profile string" so I can originate the call 
directly, and I won't need a loopback. I could of course connect to the sofia 
string directly, but it would be nice to leave that kind of lookup logic to the 
dialplan.

Thanks for staying with me - I hope you understand my problem :)

Regards,

Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 14 april 2009 18:26
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...

The bridge application will let you bridge right to a destination on *another* 
box.
If you want to connect to a local extension like 5000 you can use the transfer 
application or method.

session.transfer("5000");
exit();

or

session.execute("transfer", "5000");
exit();


On Tue, Apr 14, 2009 at 10:59 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

Yes, I'm starting to realize that... :) but you to get everything right - if I 
want to bridge a call, using the dialplan, then the only way is to use 
loopback, right? If I don't want a loopback I'm able to bridge to the 
destination directly?



//Peter





Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 
[mailto:freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>]
 För Anthony Minessale
Skickat: den 14 april 2009 17:27

Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...



yes,

But if you plan is to bridge the call, the loopback channel is completely 
unnecessary.
Be careful how much control you want =D getting a phone call up and running is 
more work
than you think (see switch_ivr_originate.c)

On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

Anthony,



Yes, it seems to work correct now. I did a couple of test calls, and tha audio 
was good - thanks!



Another question about this scenario...



When doing a session.transfer("5000"), this will transfer the call directly 
into the dialplan without the use of loopback-channels. But that way it's not 
possible to do it in a controlled way. Shouldn't it be possible to do the same 
thing with a bridge? As soon as the call is bridged, it gets "rid of" 
unneccecary loopback channels, and connecting the two endpoints directly - 
cause by then it should be two "normal" endpoints talking?



Regards,



Peter



Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 
[mailto:freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>]
 För Anthony Minessale
Skickat: den 13 april 2009 20:38
Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...



see how it works in latest trunk 13011

nontheless you can just say

session.execute("bridge", "loopback/5000");

and get the same result without touching that other channel.

when the call fails, you will have an originate_disposition variable in session 
you can check.

On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
w

Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...

[Peter Olsson]

Yes, I'm starting to realize that... :) but you to get everything right - if I 
want to bridge a call, using the dialplan, then the only way is to use 
loopback, right? If I don't want a loopback I'm able to bridge to the 
destination directly?

//Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 14 april 2009 17:27
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...

yes,

But if you plan is to bridge the call, the loopback channel is completely 
unnecessary.
Be careful how much control you want =D getting a phone call up and running is 
more work
than you think (see switch_ivr_originate.c)

On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

Anthony,



Yes, it seems to work correct now. I did a couple of test calls, and tha audio 
was good - thanks!



Another question about this scenario...



When doing a session.transfer("5000"), this will transfer the call directly 
into the dialplan without the use of loopback-channels. But that way it's not 
possible to do it in a controlled way. Shouldn't it be possible to do the same 
thing with a bridge? As soon as the call is bridged, it gets "rid of" 
unneccecary loopback channels, and connecting the two endpoints directly - 
cause by then it should be two "normal" endpoints talking?



Regards,



Peter



Från: 
freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>
 
[mailto:freeswitch-users-boun...@lists.freeswitch.org<mailto:freeswitch-users-boun...@lists.freeswitch.org>]
 För Anthony Minessale
Skickat: den 13 april 2009 20:38
Till: 
freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...



see how it works in latest trunk 13011

nontheless you can just say

session.execute("bridge", "loopback/5000");

and get the same result without touching that other channel.

when the call fails, you will have an originate_disposition variable in session 
you can check.


On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

 1.  The latest trunk I've tried with is 13008. Since I'm not doing anything 
for production yet (just testing/evaluating), so I tend to update as soon as 
there is new version available..
 2.  Yep, you will find it below. In javascript - my sample for .NET does 
basically the same thing, with the same result, except that it also won't drop 
the loopback-a call leg.
 3.  Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm 
not 100% sure what I'm doing.. :) What I want to be able to do is to dial into 
a script, let the script dial another extension, and bridge them together when 
the other party answers the call. I also need to take care of call setup 
problems - if the other part doesn't respond, is unavailable or busy in the 
phone - so I though this was the only way? If I use the 
session.execute("bridge"..), will I be able to control the call if it couldn't 
be connected?

---

if (session.ready()) {

  session.answer();

  new_session = new Session("loopback/5000", session);
  new_session.waitForAnswer();

  bridge(session, new_session);

  // Not sure if this is needed - I've tried with it both enabled and disabled
  session.hangup();
  new_session.hangup();
}

Peter


On 09-04-13 17.54, "Anthony Minessale" 
mailto:anthony.miness...@gmail.com>> wrote:

1) When you say latest, which rev does that mean? we change revs pretty often.
2) Do you have a minimal script that reproduces your issue.
3) is there a reason you cannot just session.execute("bridge", dest);
   instead of doing it manually (which is a process not for the faint at heart)?



On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:
I have two problems that I haven't been able to solve. I've done the same tests 
in both javascript, and in .NET.

The two scripts are pretty simple, they just answer an incomming call, creates 
a new session, wait for an answer on the second call leg, and then bridge the 
two channels together.

In both cases everything works just fine, but the audio is distorted. The 
destination I'm calling is "loopback/5000" - the sample IVR application 
included in FreeSWITCH. I first thought it was a codec issue, but even after 
trying to switch to different codecs the problem was the same. It more sounds 
like it's a timestamping issue - the voice is not distorted enough to be a bad 
codec, but it reads way to fast (mayby twice the "normal" speed

Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...

[Peter Olsson]

Anthony,

Yes, it seems to work correct now. I did a couple of test calls, and tha audio 
was good - thanks!

Another question about this scenario...

When doing a session.transfer("5000"), this will transfer the call directly 
into the dialplan without the use of loopback-channels. But that way it's not 
possible to do it in a controlled way. Shouldn't it be possible to do the same 
thing with a bridge? As soon as the call is bridged, it gets "rid of" 
unneccecary loopback channels, and connecting the two endpoints directly - 
cause by then it should be two "normal" endpoints talking?

Regards,

Peter

Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 13 april 2009 20:38
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...

see how it works in latest trunk 13011

nontheless you can just say

session.execute("bridge", "loopback/5000");

and get the same result without touching that other channel.

when the call fails, you will have an originate_disposition variable in session 
you can check.


On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:
 1.  The latest trunk I've tried with is 13008. Since I'm not doing anything 
for production yet (just testing/evaluating), so I tend to update as soon as 
there is new version available..
 2.  Yep, you will find it below. In javascript - my sample for .NET does 
basically the same thing, with the same result, except that it also won't drop 
the loopback-a call leg.
 3.  Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm 
not 100% sure what I'm doing.. :) What I want to be able to do is to dial into 
a script, let the script dial another extension, and bridge them together when 
the other party answers the call. I also need to take care of call setup 
problems - if the other part doesn't respond, is unavailable or busy in the 
phone - so I though this was the only way? If I use the 
session.execute("bridge"..), will I be able to control the call if it couldn't 
be connected?

---

if (session.ready()) {

  session.answer();

  new_session = new Session("loopback/5000", session);
  new_session.waitForAnswer();

  bridge(session, new_session);

  // Not sure if this is needed - I've tried with it both enabled and disabled
  session.hangup();
  new_session.hangup();
}

Peter


On 09-04-13 17.54, "Anthony Minessale" 
mailto:anthony.miness...@gmail.com>> wrote:

1) When you say latest, which rev does that mean? we change revs pretty often.
2) Do you have a minimal script that reproduces your issue.
3) is there a reason you cannot just session.execute("bridge", dest);
   instead of doing it manually (which is a process not for the faint at heart)?



On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:
I have two problems that I haven't been able to solve. I've done the same tests 
in both javascript, and in .NET.

The two scripts are pretty simple, they just answer an incomming call, creates 
a new session, wait for an answer on the second call leg, and then bridge the 
two channels together.

In both cases everything works just fine, but the audio is distorted. The 
destination I'm calling is "loopback/5000" - the sample IVR application 
included in FreeSWITCH. I first thought it was a codec issue, but even after 
trying to switch to different codecs the problem was the same. It more sounds 
like it's a timestamping issue - the voice is not distorted enough to be a bad 
codec, but it reads way to fast (mayby twice the "normal" speed). When doing a 
direct transfer() to the other destination this works just fine, but I need to 
be able to have some extra logic to tell if the destination is available or not.

The second problem occurs only in .NET. After doing this sample there is as 
loopback channel still hanging around. It seems like the call creates a 
loopback-a and loopback-b, the loopback-b dissapears as it should (when the 
call has been disconnected), but the other one stays there. When doing the same 
in javascript this doesn't seem to occur.

I'm using the latest SVN trunk, and my OS is Windows XP.

I found bug FSCORE-349 in Jira, which seems to point in to the direction that 
there might be a bug with the loopback channels in some cases, but I could not 
find anything about the audio which plays too fast.

Has anyone else experienced this?

Regards,

Peter Olsson

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Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...

[Peter Olsson]

 1.  The latest trunk I've tried with is 13008. Since I'm not doing anything 
for production yet (just testing/evaluating), so I tend to update as soon as 
there is new version available..
 2.  Yep, you will find it below. In javascript - my sample for .NET does 
basically the same thing, with the same result, except that it also won't drop 
the loopback-a call leg.
 3.  Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm 
not 100% sure what I'm doing.. :) What I want to be able to do is to dial into 
a script, let the script dial another extension, and bridge them together when 
the other party answers the call. I also need to take care of call setup 
problems - if the other part doesn't respond, is unavailable or busy in the 
phone - so I though this was the only way? If I use the 
session.execute("bridge"..), will I be able to control the call if it couldn't 
be connected?

---

if (session.ready()) {

   session.answer();

   new_session = new Session("loopback/5000", session);
   new_session.waitForAnswer();

   bridge(session, new_session);

   // Not sure if this is needed - I've tried with it both enabled and disabled
   session.hangup();
   new_session.hangup();
}

Peter


On 09-04-13 17.54, "Anthony Minessale"  wrote:

1) When you say latest, which rev does that mean? we change revs pretty often.
2) Do you have a minimal script that reproduces your issue.
3) is there a reason you cannot just session.execute("bridge", dest);
instead of doing it manually (which is a process not for the faint at 
heart)?



On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson 
 wrote:
I have two problems that I haven't been able to solve. I've done the same tests 
in both javascript, and in .NET.

The two scripts are pretty simple, they just answer an incomming call, creates 
a new session, wait for an answer on the second call leg, and then bridge the 
two channels together.

In both cases everything works just fine, but the audio is distorted. The 
destination I'm calling is "loopback/5000" - the sample IVR application 
included in FreeSWITCH. I first thought it was a codec issue, but even after 
trying to switch to different codecs the problem was the same. It more sounds 
like it's a timestamping issue - the voice is not distorted enough to be a bad 
codec, but it reads way to fast (mayby twice the "normal" speed). When doing a 
direct transfer() to the other destination this works just fine, but I need to 
be able to have some extra logic to tell if the destination is available or not.

The second problem occurs only in .NET. After doing this sample there is as 
loopback channel still hanging around. It seems like the call creates a 
loopback-a and loopback-b, the loopback-b dissapears as it should (when the 
call has been disconnected), but the other one stays there. When doing the same 
in javascript this doesn't seem to occur.

I'm using the latest SVN trunk, and my OS is Windows XP.

I found bug FSCORE-349 in Jira, which seems to point in to the direction that 
there might be a bug with the loopback channels in some cases, but I could not 
find anything about the audio which plays too fast.

Has anyone else experienced this?

Regards,

Peter Olsson

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Re: [Freeswitch-users] mod_opal and DTMF...

[Peter Olsson]

Anthony,

Please read my comments on Jira cases MODOPAL-3 and MODOPAL-5.

In MODOPAL-3 I've also attached a patch to get DTMF working better (handle both 
Tone and String input), and also increasing the stability of mod_opal.

But the problem also is connected to MODOPAL-5, which I created just now. It 
seems to be a codec issue when using A-Law, which I didn't found out until now 
- that's why the in-band DTMF detection didn't work.

But I have DTMF working right now, but it's a bit improved with my patch :)

Regards,

Peter


On 09-04-13 17.25, "Anthony Minessale"  wrote:

which rev was "latest" for you?
I have confirmation that it is indeed working.


On Sun, Apr 12, 2009 at 6:28 AM, Peter Olsson 
 wrote:
Thanks,

I tried latest trunk, but still no success.. :(

Peter


On 09-04-11 18.30, "Anthony Minessale"  wrote:

see if it works in latest trunk please.


On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson 
 wrote:
When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem to 
detect DTMF. I'm not sure if this is a setting somewhere in the config files, 
or if it's a bug. The test scenario is simple - use the default FreeSWITCH 
config, dial in to voicemail (4000) and try to log in. It doesn't detect any 
DTMF tones.

Regards,

Peter Olsson

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[Freeswitch-users] Use of loopback channels and bridge() in scripts...

[Peter Olsson]

I have two problems that I haven't been able to solve. I've done the same tests 
in both javascript, and in .NET.

The two scripts are pretty simple, they just answer an incomming call, creates 
a new session, wait for an answer on the second call leg, and then bridge the 
two channels together.

In both cases everything works just fine, but the audio is distorted. The 
destination I'm calling is "loopback/5000" - the sample IVR application 
included in FreeSWITCH. I first thought it was a codec issue, but even after 
trying to switch to different codecs the problem was the same. It more sounds 
like it's a timestamping issue - the voice is not distorted enough to be a bad 
codec, but it reads way to fast (mayby twice the "normal" speed). When doing a 
direct transfer() to the other destination this works just fine, but I need to 
be able to have some extra logic to tell if the destination is available or not.

The second problem occurs only in .NET. After doing this sample there is as 
loopback channel still hanging around. It seems like the call creates a 
loopback-a and loopback-b, the loopback-b dissapears as it should (when the 
call has been disconnected), but the other one stays there. When doing the same 
in javascript this doesn't seem to occur.

I'm using the latest SVN trunk, and my OS is Windows XP.

I found bug FSCORE-349 in Jira, which seems to point in to the direction that 
there might be a bug with the loopback channels in some cases, but I could not 
find anything about the audio which plays too fast.

Has anyone else experienced this?

Regards,

Peter Olsson

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Re: [Freeswitch-users] mod_opal and DTMF...

[Peter Olsson]

Thanks,

I tried latest trunk, but still no success.. :(

Peter


On 09-04-11 18.30, "Anthony Minessale"  wrote:

see if it works in latest trunk please.


On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson 
 wrote:
When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem to 
detect DTMF. I'm not sure if this is a setting somewhere in the config files, 
or if it's a bug. The test scenario is simple - use the default FreeSWITCH 
config, dial in to voicemail (4000) and try to log in. It doesn't detect any 
DTMF tones.

Regards,

Peter Olsson

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Re: [Freeswitch-users] Some spidermonkey modules fails to load in Windows

[Peter Olsson]

I looked into the SVN logs early this morning, and found out that something was 
changed for this.

However, the problem still exists for me, even though I make a clean and full 
rebuild of FreeSWITCH.

Peter


On 09-04-10 14.29, "Michael Jerris"  wrote:



On Apr 10, 2009, at 4:59 AM, Peter Olsson wrote:

> The spidermonkey modules core_db/odbc, curl, socket and teletone
> fails to load in Windows. They just return error 1271 (Sym Error).
> I'm not sure if this is a known issue, or if just doesn't work in
> Windows :) I've been using the latest SVN when trying this.
>
> Any ideas anyone? :)

This was fixed in svn yesterday.

Mike


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[Freeswitch-users] How to find/build FreeSWITCH.Managed.dll?

[Peter Olsson]

Hello again!

When trying to load the mod_managed module it get an error that it can't find 
FreeSWITCH.Managed.dll.

So my question is simply - where do I find this file, or how do I build it? I'm 
using the VC++ Express edition when building, so I guess I also have to install 
the C# edition - will this solve my problem?

Sorry for asking stupid questions here - but I've just been playing around with 
FreeSWITCH for a day or so :)

Regards,

Peter Olsson

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[Freeswitch-users] Some spidermonkey modules fails to load in Windows

[Peter Olsson]

The spidermonkey modules core_db/odbc, curl, socket and teletone fails to load 
in Windows. They just return error 1271 (Sym Error). I'm not sure if this is a 
known issue, or if just doesn't work in Windows :) I've been using the latest 
SVN when trying this.

Any ideas anyone? :)

Regards,

Peter Olsson

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[Freeswitch-users] mod_opal and DTMF...

[Peter Olsson]

When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem to 
detect DTMF. I'm not sure if this is a setting somewhere in the config files, 
or if it's a bug. The test scenario is simple - use the default FreeSWITCH 
config, dial in to voicemail (4000) and try to log in. It doesn't detect any 
DTMF tones.

Regards,

Peter Olsson

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Re: [Freeswitch-users] Crash in mod_opal when hanging up call...

[Peter Olsson]

Thanks for the reply. This update got rid of the error in the log, but 
FreeSWITCH still crashes. I've files a jira cace about it (MODOPAL-3).


On 09-04-10 00.03, "Anthony Minessale"  wrote:

can you try the latest trunk again r12975
if it's still a problem please open a jira on the issue
http://jira.freeswitch.org


On Thu, Apr 9, 2009 at 8:34 AM, Peter Olsson  
wrote:
Hello everyone. I've been following this project for quite some time now, but I 
never got the time to test it. But today I finally had a day off from work, so 
I could sit and play around with it for some time :) Everything wen't really 
smooth - even though I built everything from scratch on a Windows machine, 
including mod_opal (linked against the opal library). And with the docs I found 
I didn't even have to search Google to get it up and running :) So first of all 
- what a great job, guys - I'm really impressed, and the code seems really 
stable as well!

My setup right now is FreeSWITCH in Windows XP, with both SIP and H323 enabled. 
And most of the stuff works just fine. However, I think I've found a bug in 
mod_opal - it sometimes causes FreeSWITCH to crash when hanging up a call. I 
think that mod_opal is considered to be in beta stage still, so I'm not all 
that surprised. :) Check the error found in the log below. Does anyone have any 
ideas? It's pretty easy to reproduce, just dial in to FreeSWITCH using H323 and 
hang up the call, for me it happens maybe 2 out of 5 times.

I'm using latest SVN trunk versions (checked out today), for both FreeSWITCH 
and for opal/ptlib.

If you need further information, or if I should file a jira case, please get 
back to me, and I'll try to help out as much as possible.

2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:657 FSConnection::OnAlerting() 
Ring-Ready !
2009-04-09 14:37:05 [NOTICE] switch_channel.c:597 switch_channel_set_name() New 
Channel opal/in: [c7441e16-394c-d843-9ce4-760786dcecbf]
2009-04-09 14:37:05 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 
Peter Olsson [172.18.96.100]-> in context default
2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in: initialise 
opal/in:write audio codec G.711-uLaw-64k for connection 
FSMediaStream-Source-G.711-uLaw-64k
2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in: initialise 
opal/in:read audio codec G.711-uLaw-64k for connection 
FSMediaStream-Sink-G.711-uLaw-64k
2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:795 FSConnection::OnOpenMediaStream() 
Channel [opal/in:] has been answered Assertion failed: (*frame)->codec != 
((void *)0), file ..\..\src\switch_core_io.c, line 202
2009-04-09 14:37:09 [NOTICE] mod_opal.cpp:650 FSConnection::OnReleased() Hangup 
opal/in: [CS_EXECUTE] [NORMAL_CLEARING]
2009-04-09 14:37:09 [INFO] h323pdu.cxx:1005 H225() Read error (0):

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Re: [Freeswitch-users] Status of Sangoma support in Windows?

[Peter Olsson]

That sounds great - we'll just have to hope for the best :)

Thanks for your quick response.

//Peter


On 09-04-09 16.05, "Michael Jerris"  wrote:

The code that "should" work for this is on a box at Sangmoa under
testing right now on linux.  It should be committed as soon as the new
driver is released (which the new module will require) at which point
it will just need build integration completed and proper testing on
windows.

Mike

On Apr 9, 2009, at 9:37 AM, Peter Olsson wrote:

>> From what I've found in the docs and lists, the support for Sangoma
>> (PRI) cards is still not avaiable in the Windows port. Is this
>> planned to be implemented in the future, or will it never be
>> included in Win32? Just a curious thought, since I might need to
>> use some PRI stuff in the future...
>
> Regards,
>
> Peter Olsson


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[Freeswitch-users] Crash in mod_opal when hanging up call...

[Peter Olsson]

Hello everyone. I've been following this project for quite some time now, but I 
never got the time to test it. But today I finally had a day off from work, so 
I could sit and play around with it for some time :) Everything wen't really 
smooth - even though I built everything from scratch on a Windows machine, 
including mod_opal (linked against the opal library). And with the docs I found 
I didn't even have to search Google to get it up and running :) So first of all 
- what a great job, guys - I'm really impressed, and the code seems really 
stable as well!

My setup right now is FreeSWITCH in Windows XP, with both SIP and H323 enabled. 
And most of the stuff works just fine. However, I think I've found a bug in 
mod_opal - it sometimes causes FreeSWITCH to crash when hanging up a call. I 
think that mod_opal is considered to be in beta stage still, so I'm not all 
that surprised. :) Check the error found in the log below. Does anyone have any 
ideas? It's pretty easy to reproduce, just dial in to FreeSWITCH using H323 and 
hang up the call, for me it happens maybe 2 out of 5 times.

I'm using latest SVN trunk versions (checked out today), for both FreeSWITCH 
and for opal/ptlib.

If you need further information, or if I should file a jira case, please get 
back to me, and I'll try to help out as much as possible.

2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:657 FSConnection::OnAlerting() 
Ring-Ready !
2009-04-09 14:37:05 [NOTICE] switch_channel.c:597 switch_channel_set_name() New 
Channel opal/in: [c7441e16-394c-d843-9ce4-760786dcecbf]
2009-04-09 14:37:05 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 
Peter Olsson [172.18.96.100]-> in context default
2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in: initialise 
opal/in:write audio codec G.711-uLaw-64k for connection 
FSMediaStream-Source-G.711-uLaw-64k
2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in: initialise 
opal/in:read audio codec G.711-uLaw-64k for connection 
FSMediaStream-Sink-G.711-uLaw-64k
2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:795 FSConnection::OnOpenMediaStream() 
Channel [opal/in:] has been answered Assertion failed: (*frame)->codec != 
((void *)0), file ..\..\src\switch_core_io.c, line 202
2009-04-09 14:37:09 [NOTICE] mod_opal.cpp:650 FSConnection::OnReleased() Hangup 
opal/in: [CS_EXECUTE] [NORMAL_CLEARING]
2009-04-09 14:37:09 [INFO] h323pdu.cxx:1005 H225() Read error (0):

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[Freeswitch-users] Status of Sangoma support in Windows?

[Peter Olsson]

>From what I've found in the docs and lists, the support for Sangoma (PRI) 
>cards is still not avaiable in the Windows port. Is this planned to be 
>implemented in the future, or will it never be included in Win32? Just a 
>curious thought, since I might need to use some PRI stuff in the future...

Regards,

Peter Olsson

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[Freeswitch-users] FreeSWITCH, Sangoma cards and Win32

[Peter Olsson]

Hello everyone,

First of all - I've been following this project for some time now, and I've 
tried it out on both Linux and Windows. And I must say - I'm very impressed :) 
Good and stable code, and the community seems very good as well. I've been 
using both Asterisk and Yate before, but now we're looking more and more into 
FreeSWITCH.

I have a couple of questions though - it's about Sangoma support in Win32. 
First of all, when looking through the information on Sangoma's website I found 
out that the driver for Win32 still is in beta state - I'm not sure about how 
stable/unstable it is, since I havn't tried it at all yet. But how about 
FreeSWITCH and Sangoma in Win32, will this setup be supported in a near future, 
or how is the plans for this? And when/if this support will be added - how 
about support for Q.SIG, will this be implemented as well, or do you have any 
plans at all for this? I'm not talking about the basic Q.SIG handlig that 
Asterisk does (we've done pretty much patching for Q.SIG in Asterisk to get 
things to work properly), I'm interested in a more complete implementation of 
Q.SIG, to be able to do path replacement etc.

I've looked through the archives, but I couldn't find any information about 
this - so all answers are welcome :)

Regards,

Peter Olsson
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