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gmaruzz at celliax.org
Guest

Posted: Wed Nov 04, 2009 12:34 pm Post subject: [Freeswitch-users] Skypiax load error

2009/11/4 ´óÄàÈË <qinglan_zeng@hotmail.com>:

Quote:

Newbie to FS and I installed FS using Windows precompiled binaries. I want
to set up some skype trunks with FS and so I followed the instructions while
get some errors:
(1). Launch Skype by clicking the skype.exe.
(2). Launch FS
(3) In FS I enter the cmd as: load mod_skypiax and then come to error:
module load file routine retured an error. I had saved a screenshot for your
referrence.

Please Daniel, do not send mail both to me personally and to the
mailing list. Send only to the mailing list.

As you can see in the screenshot you attach, mod_skypiax cannot find
its configuration file.

For what I can understand, you have not the skills needed for
administering FS, so it would be better if you find someone (a friend,
etc) that can help you.

-giovanni

Quote:

Any idea on this?

Thanks
Daniel Zeng

From: freeswitch-users-request@lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 41, Issue 27
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Nov 2009 07:46:45 -0800

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

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
freeswitch-users-request@lists.freeswitch.org

You can reach the person managing the list at
freeswitch-users-owner@lists.freeswitch.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."

--¸½×ª·¢µÄÓÊ¼þ--
From: diego.viola@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date : Wed, 4 Nov 2009 15:25:55 +0000
Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress

Hello,

I tried to help Roy with this issue yesterday, I saw that calls couldn't go
through and then I made a sofia profile internal siptrace on.

Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" and saw
he had like 800 sessions.

I thought it was an ACL issue but it wasn't, it seems like he reached a
session limit, when I restarted his FS the problem went away.

Best Regards,

Diego

On Wed, Nov 4, 2009 at 5:41 AM, RA Cohen <roy@net-vantage.com> wrote:

Here's what's in switch.conf.xml:



<param name="max-sessions" value="1000"/>



<param name="sessions-per-second" value="30"/>

Yet this message: SIP/2.0 503 Maximum Calls In Progress

This is a small medical practice, 5-6 extensions, 3000 outbound minutes

per month and at least the same inbound. We did fsctl shutdown restart

and it flushed the sessions. What is going on?

Thank you for your help!

--

Roy A Cohen

Network Advantage LLC

www.net-vantage.com

413.223.9007 option 1

--------------------------------------------------

"Bringing Cost-Saving, State-of-the-Art Technology

Solutions to Small and M id-Size Organizations"

_______________________________________________

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

--¸½×ª·¢µÄÓÊ¼þ--
From: anthony.minessale@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Nov 2009 09:39:21 -0600
Subject: Re: [Freeswitch-users] SIP Overlap support?

You cannot.
This is how the sip spec works.
Every new invite is a new call and a new trip to the dialplan.

You will probabl y need to design your code to send the appropriate 484 and
be prepared to exit and be called again with the new digits.

On Wed, Nov 4, 2009 at 2:23 AM, Dennis <odermann@googlemail.com> wrote:

is there a way to send something like 484 (or something else), which

does not make it a final answer and keep the call/socket alive?

so we can ask the cirpack for further digits and decide what to do, if

the cirpack does not send any digits.

2009/11/3 Anthony Minessale <anthony.minessale@gmail.com>:

Quote:

The patch was it's ability to accept subsequent invites.

Quote:

Your problem is that in sip each new attempt to send an invite is another

Quote:

call.

Quote:

484 is a final response so the call with too few digits is terminated.

Quote:

Hello,

I tried to help Roy with this issue yesterday, I saw that ca

lls

Quote:

couldn't go through and then I made a sofia profile internal
siptrace on.

Then I found a message like "SIP/2.0 503 Maximum Calls In Progress"
and saw he had like 800 sessions.

I thought it was an ACL issue but it wasn't, it seems like he
reached a session limit, when I restarted his FS the problem went
away.

Best Regards,

Diego

--¸½×ª·¢µÄÓÊ¼þ--
From: jlenk@frontiernet.net
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Nov 2009 07:43:54 -0800
Subject: Re: [Freeswitch-users] Precompiled Windows Binaries

Hi Carlos,

very cool that the x64 version is included now! Hopefully this will get more
people using the x64 version under Windows!

Regards,
Jeff

Carlos Talbot wrote:

Quote:

On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson
<stevendt@primros

ebank.net>wrote:

Quote:

Jeff,

thanks a lot for the reply. I was a little confused by the fact that the
"SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried
that I might lose something. As you say though, think that I'll cross my
fingers and try the updated release. I am running FreeSwitch on a test
machine at the moment until the target hardware arrives - hopefully
tomorrow, so I can afford to have a little play.

I usually try to update the svn file at least once a month. I have a new
version ready that was compiled last night but am ironing out login issues
with the FS dudes for upload access. Also, the SVN snapshot now includes
binaries for 32 and 64 bit. It no longer includes flite though as the
install file was approaching 80MB in size. I will revisit this later if
others feel it impo

rtant to include flite.

Quote:

You mentioned FreePBX V3. I had been fumbling around trying to work out
what
this is and from what I've read, it seems to provide a GUI Front End for
configuring FreeSwitch ?

Yes, it's still in development phase and as such not ready for production
use.

Quote:

I am guessing that while it has been installed with FreeSwitch, I then
need
to run the FreePBX Installer to update the FreePBX/FreeSwitch
configuration
on my hardware ?

When I start FreeSwitch, it does not automatically load the WAMPServer.

Freeswitch and WAMPServer are independant of each other. WAMPServer is

bundled in this install for the purpose of FreePBX as MySQL, Apache and
PHP
are all required components of FreePBX.

When

I start WAMPServer manually, and open up localhost (127.0.0.1) in a

Quote:

web

Quote:

browser, I can see the WampServer logo and various tools such as
phpinfo()
and phpmyadmin. FreePBX is there under Your Projects.

If you want to configure FreePBX you need to click on the FreePBX.url

shortcut that gets created on your desktop.

Quote:

When I opened this up the first time, it appeared to want to install
FreePBX
over FreeSwitch, I tried to abort this when it was going to overwrite
some
FreeSwitch conf files and I thought I'd better not go on until I had a
better idea what was happening. I backed out of the FreePBX install and
now
I can't get the FreePBX or phpmyadmin pages up again (missing files) so
it
looks like I'm going to have to reinstall anyway.

So, for nex

t time,am I right in thinking that I should proceed with

Quote:

running
the FreePBX install from the WAMPServer menu ?

No, launch it from the shortcut as stated above. Unfortunately, at this
time
there is very little user documentation on configuring FreePBX. Here is
the
link to the developer's info: http://www.freepbx.org/v3

regards,

Carlos

Quote:

----- Original Message -----
From: "Jeff Lenk" <jlenk@frontiernet.net>
To: <freeswitch-users@lists.freeswitch.org>
Sent: Tuesday, November 03, 2009 2:48 PM
Subject: Re: [Freeswitch-users] Precompiled Windows Binaries

Quote:

Hi Dave,

These are supported by "Carlos Talbot" . They also include Freepbx v3<

BR>>> >

Quote:

Just as you said freeswitch-1.0.4.exe is the tagged release and
freeswitch.exe is a newer svn snapshot.

There should be no problems installing the new version allthough best

Quote:

just try and see!

Not sure why the newest one is from October 7th.

Jeff

Dave Stevenson wrote:

Quote:

Hi,

I have read the Docs on the Wiki
(

http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries)

Quote:

but am still not sure of what the different Windows install files are.
Currently, the Windows Installer directory contains :-

LATEST_SVN_15106 - 6 Bytes

freeswitch-1.0.4.exe - 42 Megabytes

freeswitch.exe - 32 Megabytes

I have installed the freeswitch-1.0.4.exe file which is dated 3rd
September. The freeswitch.exe file is dated 7th October and think that

Quote:

contains the minor updates since 3rd September ?

Could someone who knows FreeSwitch under windows help me understand

the

Quote:

two files please ?

I chickened out of running the later exe in case it did something to

the

Quote:

running install of FreeSwitch 1.0.4, is it safe to run the newer exe

with

Quote:

the old one already installed ?
What will it

actually do ?

Quote:

regards
Dave
_______________________________________________
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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

Quote:

http://www.freeswitch.org

--
View this message in context:

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937943p3938887.html

Quote:

Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

Quote:

http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http

://lists.freeswitch.org/mailman/listinfo/freeswitch-users

Quote:

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

--
View this message in context: http
://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3946039.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

--¸½×ª·¢µÄÓÊ¼þ--
From: tculjaga@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Nov 2009 16:44:08 +0100
Subject: Re: [Freeswitch-users] SIP Overlap support?

Brian is right,

pls, lets stop with exceptions and get stick to RFCs... otherwise it will be
a big mess ...

T.

On Wed, Nov 4, 2009 at 3:03 PM, Brian West <brian@freeswitch.org> wrote:

I'm going to say No!

/b

On Nov 4, 2009, at 2:23 AM, Dennis wrote:

Quote:

is there a way to send something like 484 (or something else), which

Quote:

does not make it a final answer and keep the call/socket alive?

Quote:

so we can ask the cirpack for further digits and decide what to do, if

Quote:

the cirpack does not send any digits.

_______________________________________________

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

--¸½×ª·¢µÄÓÊ¼þ--
From: anthony.minessale@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Nov 2009 09:46:11 -0600
Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress

Which revision of FreeSWITCH are you using?

On Tue, Nov 3, 2009 at 11:41 PM, RA Cohen <roy@net-vantage.com> wrote:

Here's what's in switch.conf.xml:



<param name="max-sessions" value="1000"/>



<param name="sessions-per-second" value="30"/>

Yet this message: SIP/2.0 503 Maximum Calls In Progress

This is a small medical practice, 5-6 extensions, 3000 outbound minutes

per month and at least the same inbound. We did fsctl shutdown restart

and it flushed the sessions. What is going on?

Thank you for your help!

--

Roy A Cohen

Network Advantage LLC

www.net-vantage.com

413.223.9007 option 1

--------------------------------------------------

"Bringing Cost-Saving, State-of-the-Art Technology

Solutions to Small and M id-Size Organizations"

_______________________________________________

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_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888

googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400

________________________________
¸ü¶àÈÈÀ±×ÊÑ¶¾¡ÔÚÐÂ°æMSNÊ×Ò³£¡ Á¢¿Ì·ÃÎÊ£¡
_______________________________________________
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

--
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

_______________________________________________
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jlenk at frontiernet.net
Guest

Posted: Wed Nov 04, 2009 12:38 pm Post subject: [Freeswitch-users] Skypiax load error

follow these intructions ->
http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances

å¤§æ³¥äºº wrote:

Quote:

Hello All,

Newbie to FS and I installed FS using Windows precompiled binaries. I want
to set up some skype trunks with FS and so I followed the instructions
while get some errors:

(1). Launch Skype by clicking the skype.exe.

(2). Launch FS

(3) In FS I enter the cmd as: load mod_skypiax and then come to error:
module load file routine retured an error. I had saved a screenshot for
your referrence.

Any idea on this?

Thanks

Daniel Zeng

From: freeswitch-users-request@lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 41, Issue 27
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Nov 2009 07:46:45 -0800

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

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
freeswitch-users-request@lists.freeswitch.org

You can reach the person managing the list at
freeswitch-users-owner@lists.freeswitch.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."

--é™„è½¬å‘çš„é‚®ä»¶--
From: diego.viola@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Nov 2009 15:25:55 +0000
Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress

Hello,

I tried to help Roy with this issue yesterday, I saw that calls couldn't
go through and then I made a sofia profile internal siptrace on.

Then I found a message like "SIP/2.0 503 Maximum Calls In Progress" and
saw he had like 800 sessions.

I thought it was an ACL issue but it wasn't, it seems like he reached a
session limit, when I restarted his FS the problem went away.

Best Regards,

Diego

On Wed, Nov 4, 2009 at 5:41 AM, RA Cohen <roy@net-vantage.com> wrote:

Here's what's in switch.conf.xml:





Yet this message: SIP/2.0 503 Maximum Calls In Progress

This is a small medical practice, 5-6 extensions, 3000 outbound minutes

per month and at least the same inbound. We did fsctl shutdown restart

and it flushed the sessions. What is going on?

Thank you for your help!

--

Roy A Cohen

Network Advantage LLC

www.net-vantage.com

413.223.9007 option 1

--------------------------------------------------

"Bringing Cost-Saving, State-of-the-Art Technology

Solutions to Small and Mid-Size Organizations"

_______________________________________________

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

--é™„è½¬å‘çš„é‚®ä»¶--
From: anthony.minessale@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Nov 2009 09:39:21 -0600
Subject: Re: [Freeswitch-users] SIP Overlap support?

You cannot.
This is how the sip spec works.
Every new invite is a new call and a new trip to the dialplan.

You will probably need to design your code to send the appropriate 484 and
be prepared to exit and be called again with the new digits.

On Wed, Nov 4, 2009 at 2:23 AM, Dennis <odermann@googlemail.com> wrote:

is there a way to send something like 484 (or something else), which

does not make it a final answer and keep the call/socket alive?

so we can ask the cirpack for further digits and decide what to do, if

the cirpack does not send any digits.

2009/11/3 Anthony Minessale <anthony.minessale@gmail.com>:

Quote:

The patch was it's ability to accept subsequent invites.

Quote:

Your problem is that in sip each new attempt to send an invite is another

Quote:

call.

Quote:

484 is a final response so the call with too few digits is terminated.

Quote:

Hello,

I tried to help Roy with this issue yesterday, I saw that calls
couldn't go through and then I made a sofia profile internal
siptrace on.

Then I found a message like "SIP/2.0 503 Maximum Calls In Progress"
and saw he had like 800 sessions.

I thought it was an ACL issue but it wasn't, it seems like he
reached a session limit, when I restarted his FS the problem went
away.

Best Regards,

Diego

--é™„è½¬å‘çš„é‚®ä»¶--
From: jlenk@frontiernet.net
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Nov 2009 07:43:54 -0800
Subject: Re: [Freeswitch-users] Precompiled Windows Binaries

Hi Carlos,

very cool that the x64 version is included now! Hopefully this will get
more
people using the x64 version under Windows!

Regards,
Jeff

Carlos Talbot wrote:

Quote:

On Tue, Nov 3, 2009 at 11:27 AM, Dave Stevenson
<stevendt@primrosebank.net>wrote:

Quote:

I usually try to update the svn file at least once a month. I have a new
version ready that was compiled last night but am ironing out login
issues
with the FS dudes for upload access. Also, the SVN snapshot now includes
binaries for 32 and 64 bit. It no longer includes flite though as the
install file was approaching 80MB in size. I will revisit this later if
others feel it important to include flite.

Quote:

You mentioned FreePBX V3. I had been fumbling around trying to work out
what
this is and from what I've read, it seems to provide a GUI Front End for
configuring FreeSwitch ?

Yes, it's still in development phase and as such not ready for production
use.

Quote:

bundled in this install for the purpose of FreePBX as MySQL, Apache and
PHP
are all required components of FreePBX.

When I start WAMPServer manually, and open up localhost (127.0.0.1) in a
web

Quote:

browser, I can see the WampServer logo and various tools such as
phpinfo()
and phpmyadmin. FreePBX is there under Your Projects.

If you want to configure FreePBX you need to click on the FreePBX.url

shortcut that gets created on your desktop.

Quote:

Hi Dave,

These are supported by "Carlos Talbot" . They also include Freepbx v3

Just as you said freeswitch-1.0.4.exe is the tagged release and
freeswitch.exe is a newer svn snapshot.

There should be no problems installing the new version allthough best

Quote:

just try and see!

Not sure why the newest one is from October 7th.

Jeff

Dave Stevenson wrote:

Quote:

Hi,

I have read the Docs on the Wiki
(

http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries)

Quote:

but am still not sure of what the different Windows install files

are.

Quote:

Currently, the Windows Installer directory contains :-

LATEST_SVN_15106 - 6 Bytes

freeswitch-1.0.4.exe - 42 Megabytes

freeswitch.exe - 32 Megabytes

I have installed the freeswitch-1.0.4.exe file which is dated 3rd
September. The freeswitch.exe file is dated 7th October and think

that
it

Quote:

contains the minor updates since 3rd September ?

Could someone who knows FreeSwitch under windows help me understand

the

Quote:

two files please ?

I chickened out of running the later exe in case it did something to

the

Quote:

running install of FreeSwitch 1.0.4, is it safe to run the newer exe

with

Quote:

the old one already installed ?
What will it actually do ?

regards
Dave
_______________________________________________
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

Quote:

http://www.freeswitch.org

--
View this message in context:

http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html

Quote:

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

Quote:

http://www.freeswitch.org

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--é™„è½¬å‘çš„é‚®ä»¶--
From: tculjaga@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Nov 2009 16:44:08 +0100
Subject: Re: [Freeswitch-users] SIP Overlap support?

Brian is right,

pls, lets stop with exceptions and get stick to RFCs... otherwise it will
be a big mess ...

T.

On Wed, Nov 4, 2009 at 3:03 PM, Brian West <brian@freeswitch.org> wrote:

I'm going to say No!

/b

On Nov 4, 2009, at 2:23 AM, Dennis wrote:

Quote:

is there a way to send something like 484 (or something else), which

Quote:

does not make it a final answer and keep the call/socket alive?

Quote:

so we can ask the cirpack for further digits and decide what to do, if

Quote:

the cirpack does not send any digits.

_______________________________________________

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

--é™„è½¬å‘çš„é‚®ä»¶--
From: anthony.minessale@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Nov 2009 09:46:11 -0600
Subject: Re: [Freeswitch-users] SIP/2.0 503 Maximum Calls In Progress

Which revision of FreeSWITCH are you using?

On Tue, Nov 3, 2009 at 11:41 PM, RA Cohen <roy@net-vantage.com> wrote:

Here's what's in switch.conf.xml:





Yet this message: SIP/2.0 503 Maximum Calls In Progress

This is a small medical practice, 5-6 extensions, 3000 outbound minutes

per month and at least the same inbound. We did fsctl shutdown restart

and it flushed the sessions. What is going on?

Thank you for your help!

--

Roy A Cohen

Network Advantage LLC

www.net-vantage.com

413.223.9007 option 1

--------------------------------------------------

"Bringing Cost-Saving, State-of-the-Art Technology

Solutions to Small and Mid-Size Organizations"

_______________________________________________

FreeSWITCH-users mailing list

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View this message in context: http://n2.nabble.com/Skypiax-load-error-tp3946656p3946759.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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qinglan_zeng at hotmai...
Guest

Posted: Thu Nov 05, 2009 10:38 am Post subject: [Freeswitch-users] Skypiax load error

dujinfang at gmail.com
Guest

Posted: Thu Nov 05, 2009 12:41 pm Post subject: [Freeswitch-users] Skypiax load error

2009/11/5 ´óÄàÈË <qinglan_zeng@hotmail.com (qinglan_zeng@hotmail.com)>

Quote:

Hi All,

I once meet the Skypiax load error issue and some guys infomed me that there is no configuration file for Skypiax.

When I follow these intructions ->
http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances

I still have some difficulties unstanding this:
." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml to Debug\conf\autoload_configs."
I did not find such directories in my freeswitch folder. Did not understand what "src" means, I checked the freeswitch folder and did not find such a folder named"src". There is a folder named"mod" under freeswitch while look into "mod" folder there are only some DLL files and can not find endpoints and etc.

src means source code. you can check out from svn trunk or download the source code from files.freeswitch.org. or follow fisheye: http://fisheye.freeswitch.org/browse/FreeSWITCH

Quote:

2.You'll probably build the "Debug" version
I just build this from the precompiled binaries and then launched FS. I'm not sure what I launched is in debug mode or not.

If anyone can offer some help that really be appriciated.

I never try to do that on windows, but we are running skype trunks on Linux. Also there's a Chinese google group about FreeSWITCH:

http://groups.google.com/group/freeswitch-cn

And also please don't include long unrelated quoted text in your mail. It wastes time to read and hence some read emails on cellphone.

gmaruzz at celliax.org
Guest

Posted: Thu Nov 05, 2009 12:53 pm Post subject: [Freeswitch-users] Skypiax load error

2009/11/5 ´óÄàÈË <qinglan_zeng@hotmail.com>:

Quote:

I once meet the Skypiax load error issue and some guys infomed me that there
is no configuration file for Skypiax.

Daniel,

maybe this will sound not nice to your hears, but really, you better
find a friend that stay there with you and help you and teach you.

You will never be able to do things with FS with your present level of
skill. FS (and more so mod_skypiax) is not a consumer grade package.
It requires a level of skill that you do not have, at the moment.

Maybe is sad, but in my opinion is true. I tell you this just to avoid
you a lot of frustrations.

-gm

Quote:

When I follow these intructions ->
http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_start_Skype_client_instances

I still have some difficulties unstanding this:
." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml to
Debug\conf\autoload_configs."
I did not find such directories in my freeswitch folder. Did not understand
what "src" means, I checked the freeswitch folder and did not find such a
folder named"src". There is a folder named"mod" under freeswitch while look
into "mod" folder there are only some DLL files and can not find endpoints
and etc.

2.You'll probably build the "Debug" version
I just build this from the precompiled binaries and then launched FS. I'm
not sure what I launched is in debug mode or not.

If anyone can offer some help that really be appriciated.

Thanks
Daniel Zeng
From: freeswitch-users-request@lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 41, Issue 40
To: freeswitch-users@lists.freeswitch.org
Date: Thu, 5 Nov 2009 06:57:44 -0800

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

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
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You can reach the person managing the list at
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."

--¸½×ª·¢µÄÓÊ¼þ--
From: anthony.minessale@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date : Thu, 5 Nov 2009 08:13:46 -0600
Subject: Re: [Freeswitch-users] no REINVITE on Blind Transfer with
bypass_media

I did not ask you to send me a ladder diagram.
I asked you to send me a console trace from FreeSWITCH using latest trunk
(1.0.4 does not help me)

1) start FreeSWITCH
2) run the cli command: console loglevel debug

3) run the cli command: sofia profile internal siptrace on
4) reproduce your issue and put the trace on freeswitch pastebin
http://pastebin.freeswitch.org (login and pass are stated in the auth
dialog)

Also please answer brian's question. What phones and/or sip devices are
involved in this call.

On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana <hjqlopez@hotmail.com>
wrote:

Thanks for your time,

-The scenario is still the same:

Always bypass media.

Environment 100% NAT free :-)

Call established from A to B through FS. Then...

Blind transfer from B to C (Refer-to: C)

RTP should go directly between A and C.

-With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the
REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the
lack of reINVITE to A, after C answers).

Please check SIP diagram here:

http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html

-What it's wrong with r15332 is there is not such call to C. For sure I know
SIP is a protocol, may be my description was not clear but this SIP diagram
speaks by itself ;-)

http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html

-You could check the sofia debug for r15332 here:

http://pastebin.com/m6f2b3836

Best regards,

Humberto

Quote:

I don't know what you are talking about anymore.

Quote:

The scenario I had tested is when a call is bridged in bypass_media=true

Quote:

bridge

Quote:

and you blind transfer that call back to the dialplan

Quote:

as soon as it hits the routing state it will resume media.

Quote:

it has been confirmed to not work and confirmed to have been fixed several

Quote:

time and if you are still having a problem you must have something
blocking

Quote:

some of your packets or something .

Quote:

You have to understand that sip is a protocol and your description is

Quote:

completely non-standard.

Quote:

Perhaps you should get a console trace and attach it to a jira. The trace

Quote:

probably makes more sense to me.

Quote:

sofia profile internal siptrace on

Quote:

console loglevel debug

Quote:

reprodu ce and attach the whole capture.

Quote:

On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote:

Quote:

Hi,

Quote:

I tried r15332 and set in the sofia profile:

Quote:

a) bypass_media_after_bridge=true only

Quote:

b) bypass_media_after_bridge=true, param name="media-option"

Quote:

value="resume-media-on-hold"/>

Quote:

In both cases FS is hanging up the initial call (A to FS) after accepting

Quote:

the REFER to C:

Quote:

A <- reINVITE with FS' SDP <- FS

Quote:

A -> 200 -> FS

Quote:

A <- ACK <- FS

Quote:

A <- BYE <- FS

Quote:

The call to C is not even tried.

Quote:

I found this line is the logs that could give some idea:

Quote:

2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup

Quote:

sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING]
[RECOVERY_ON_TIMER_EXPIRE]

Quote:

after sending the ACK for the reINVITE

Quote:

Regards,

Quote:

Humberto

Quote:

please try r15326

Quote:

I think i have it working.

Quote:

I recommend for optimal results you set bypass_media_after_bridge=true

Quote:

either as a global or in your DP in place of bypass_media=true

Quote:

On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana

Quote:

hotmail.com>wrote:

Quote:

Hi Mike,

Quote:

I re-tried with trunk rev 15319 but I got almost the same behavior:

Quote:

There

Quote:

is now a reINVITE (with FS' SDP) going to A when the REFER is accepted.

Quote:

But

Quote:

still there is no reINVITE for A (with C's SDP) after the call from FS

Quote:

to C

Quote:

is established.

Quote:

Anyway, we decided for now to do a different implementation but if you

Quote:

want

Quote:

to explore more in this issue count me in Wink

Quote:

Thank you very much!

Quote:

Humberto

Quote:

_____________________________________ ____________________________

Quote:

Windows Live: Friends get your Flickr, Yelp, and Digg updates when they

Quote:

e-mail you.

Quote:

http://go.microsoft.com/?linkid=9691817

Quote:

_______________________________________________

Quote:

FreeSWITCH-users mailing list

Quote:

FreeSWITCH-users at lists.freeswitch.org

Quote:

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

Quote:

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

Quote:

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Quote:

Anthony Minessale II

Quote:

_________________________________________________________________

Quote:

Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows
7 now

Quote:

http://go.microsoft.com/?linkid=9691818

_________________________________________________________________

Windows Live: Make it easier for your friends to see what you're up to on
Facebook.

http://go.microsoft.com/?linkid=9691816

_______________________________________________

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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888

googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400

--¸½×ª·¢µÄÓÊ¼þ--
From: rob4manhere@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Thu, 5 Nov 2009 08:52:05 -0600
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

Hi UK,

From what I've done and read, the caller-controls (in
conference.conf.xml) can be modified to almost anything you can think
of, BUT,
they are mapped 1-to-1 to a conference- ie you can't map a
caller control just for those with the moderator flag. So unless you
want everyone able to mute/kick everyone then you can't do it.

The wiki seems to indicate this as well:

"Be aware that the caller-controls are applied across the entire
conference. You cannot enter one member of the conference using caller-
controls ABC and then enter a second member using caller-controls XYZ."

http://wiki.freeswitch.org/wiki/Mod_conference

I think this might be a limitation of mod_conference. Perhaps one of
the pros can chime in if I'm off-base or there's some nifty way to
accomplish this.

Cheers,
Rob

On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote:

Quote:

Any ideas on the below...has anyone implemented the below:

Once I have the Moderator and Participants logg

ed on, how do I

Quote:

invoke the moderator previlidges, LIk esay muting everyone/someone
or kicking someone out of the Conf and the like?

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org
] On Behalf Of Ujjval Karihaloo
Sent: Monday, November 02, 2009 12:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

Rob:

Once I have the Moderator and Participants logged on, how do I
invoke the moderator previlidges, LIk esay muting everyone/someone
or kicking someone out of the Conf and the like?

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org
] On Behalf Of Rob Forman> Sent: Friday, October 30, 2009 9:34 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

Hm, strange. I haven't seen that before. Can you pastebin your logs
at debug level?

On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:

Quote:

It's strange... a tcpdump tells me that there is no DTMF from my
provider when using IVR, but when I call into a TN that goes
directly into the Conference App, I see DTMF from the provider.

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 7:23 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference w

ith Moderator

Quote:

I've never had any problem with that. Is your logging at debug level
so you can see the RECV DTFM in the log/fs_cli? Are you calling from
a SIP phone on the pbx, or via a PSTN provider? Maybe your provider
isn't passing them through.

Make sure your logging is turned up then try something simpler, like
calling the echo application, and see if DTFM comes through.

Rob

On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:

Quote:

Rob:

For some reason, I don't see the DTMF appear on the fs_CLI when
using the below configuration....so it basically timesout.

UK

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:

freeswitch-users-bounces@lists.freeswitch.org

Quote:

] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 26, 2009 9:21 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

Thx a lot Rob, reading the wiki your way or using IVR seems
correct..
===============
The wiki also says that the wait-mod might be "used in conjunction
with an IVR where the moderators are authenticated with an extra
pass-
code", which is what I did. I guess that's why I didn't understand
the point of the +pin.
======================

I will try it out.

Again thx a lot for your help. Will keep everyone posted.

_______________________

_________________

Quote:

From: freeswitch-users-bounces@lists.freeswitch.org
[freeswitch-users-bounces@lists.freeswitch.org
] On Behalf Of Rob Forman [rob4manhere@gmail.com]
Sent: Friday, October 23, 2009 12:22 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

I just re-tested with the pin in my dial plan:

<action application="conference" data="conference 123456@default
+flags{}+1234" />

And it doesn't challenge me for the pin. I just drop right in. I
figured this is how it was intended, since the wiki says the pin is
set initially and only challenged in later attempts [by future
callers]:

"The first time a conference name (confname) is used, i

t will be

Quote:

created on demand, and the pin will be set to what ever is specified
at that time: the pin in the data string if specified, or if not,
the
"pin" setting in the conference profile, and if that is also
unspecified, then there is no pin protection. Any later attempt to
join the conference must specify the same pin number, if one existed
when it was created. "

The wiki also says that the wait-mod might be "used in conjunction
with an IVR where the moderators are authenticated with an extra
pass-
code", which is what I did. I guess that's why I didn't understand
the point of the +pin.

I'm sure there's a scenario where its used and useful, the wiki just
doesn't explain it.

Rob

Quote:

On Oct 23, 2009, at 12:43 PM, Brian West wrote:

Quote:

Well first off you're not defining a pine here...

confname@profilename+flags{mute|deaf|waste|moderator}+[conference
pin
number]

That might be why its not asking for a pin.

/b

On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:

Quote:

_____________________________________

Quote:

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
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switch.org/mailman/options/freeswitch-users

Quote:

http://www.freeswitch.org

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Quote:

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--¸½×ª·¢µÄÓÊ¼þ--
From: rob4manhere@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: T hu, 5 Nov 2009 08:57:08 -0600
Subject: [Freeswitch-users] Wideband / HD phones

Hey all,

Looking at buying some high def phones. Any recommendations
(preferably based on experience) for hardware based on product
quality, standards compliance, features integration with Freeswitch,
etc?

Thank you!
Rob Forman

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