You can try to reduce your registration time.
I for one made my client apps send PUBLISH message every minute in addition
to reduced registration time.
Regards, Robert.
2009/12/28 Jerry Richards jerry.richa...@teotech.com
Is there a setting to control how fast FS distributes presence changes
Harondel J. Sibble scribbled:
I am following the wiki page here
http://wiki.freeswitch.org/wiki/OpenZAP#Zaptel_Installation
...
the compilation fails as below,
Zaptel is old. DAHDI is now the way to go. If you have the headers installed
for the kernel you are currently using, try this.
1.
On 12/29/2009 01:14 PM, Russell Mosemann wrote:
Harondel J. Sibble scribbled:
I am following the wiki page here
http://wiki.freeswitch.org/wiki/OpenZAP#Zaptel_Installation
...
the compilation fails as below,
Zaptel is old. DAHDI is now the way to go. If you have the headers installed
for
Hold is working fine I just tested it... I would need to see the whole dialog
to see what is wrong... I tested with Polycom, Snom and Aastra.
Are you doing proxy media or anything like that?
/b
On Dec 29, 2009, at 1:14 AM, Lei Tang wrote:
Hi, I think hold function in trunk 16055 is broken,
Also can you join #freeswitch-dev, include full siptrace+debug log and put it
on pastebin.
What phone are you using?
/b
On Dec 29, 2009, at 1:14 AM, Lei Tang wrote:
Hi, I think hold function in trunk 16055 is broken, I have also tried some
old trunks, it's ok in freeswitch 1.0.4.
The
Hi Brian, thanks for your help, I am using FS in proxy media mode. the sip
agent I'm using is x-lite and wxCommunicator.
I will test if trunk 16055 work when I set proxy media mode to false
tomorrow.
2009/12/29 Brian West br...@freeswitch.org
Hold is working fine I just tested it... I would
The phone I'm using is x-lite and wxCommunicator, both are sip phone
software.
I have not used pastebin, Is it a bug trace tool like bugzilla? Can you
tell me how to register a pastbin account?
2009/12/29 Brian West br...@freeswitch.org
Also can you join #freeswitch-dev, include full
the 200ok is not from FS.. its from the end point... so its not us thats not
putting the SDP into the 200ok but the device you're talking to because in
proxy media they are passed as is.
/b
On Dec 29, 2009, at 8:53 AM, Lei Tang wrote:
Hi Brian, thanks for your help, I am using FS in proxy
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following
code in sofia.c send the 200ok response
sofia.c
function sofia_handle_sip_i_state
.
switch(ss_state)
case nua_callstate_received:
.
Btw, in the same scenario, FS 1.0.4 works fine.
2009/12/29 Lei Tang lei.tl...@gmail.com
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following
code in sofia.c send the 200ok response
sofia.c
function sofia_handle_sip_i_state
.
switch(ss_state)
There is no need for you to show us traces. The fact that you are
using proxy media is enough to know that the issue is with your
device. If you look at the full sip trace you will see the same.
Mike
On Dec 29, 2009, at 10:10 AM, Lei Tang lei.tl...@gmail.com wrote:
The phone I'm using
Its null because the device on the other side didn't send one. We pass it as
is... fix the broken device or don't use proxy media.
/b
On Dec 29, 2009, at 9:37 AM, Lei Tang wrote:
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following
code in sofia.c send the 200ok
This means there was no sdp sent. Did you confirm this with siptrace?
On Dec 29, 2009, at 10:37 AM, Lei Tang lei.tl...@gmail.com wrote:
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong,
following code in sofia.c send the 200ok response
sofia.c
function
We now disable sofia SOA mode during proxy calls.
This means that sofia will not try to get involved in the media negotiation
at all which is the optimal behavior.
Previous versions would butt in and try to fix the error but now it just
stays out of the way.
You can see in your trace that the
FreeSWITCH is running nicely on OS X. I have used it since July 2006 on my
intel Macs with great success.
I am also developing a GUI application using Cocoa. I started that a year ago,
but haven't looked at it for a while, but this Christmas I have started working
on it again.
Ivan
Den 27.
Ivan,
I have been trying to gather up everyone to start a FreeSWITCH based
softphone project for Mac, Linux and Windows... you think we could collaborate
with you to accomplish this? I think if we do this right we can have a really
nice phone with lots of options.
Thanks,
/b
On Dec
This could be easily done with the Qt framework and would work nicely
on osx, linux and windows. Contact me off list or on IRC (Math) if you
want some help, I'd be happy to participate in such a project.
Cheers,
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1
Yes, I would like to participate on this. I have lot of experience with Cocoa
on Mac, so I could help with that platform. But I am mostly a GUI programmer,
Objective-C my language.
But if this is OK with you, I would love to help out here.
Ivan
Den 29. des. 2009 kl. 18.11 skrev Brian West:
I have done a little research into this for my employer.
You may want to look at:
http://www.qutecom.org/ - I think its QT based.
http://code.google.com/p/telephone/ - Its pure Cocoa. I use this for all my
testing, it lets me initiate up to 8 calls at a time.
On Tue, Dec 29, 2009 at 11:40 AM,
Add me to the app list I use MAC mostly ...
Also can you list the new (better, gentler) list of commands to install
FreeSwitch on a MAC OSX ... ?
Thanks in advance,
-E
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Here is my personal refactored QuteCom ...
*File:* AgileSIP-MAC.zip
You have 24 hours to retrieve this file at
http://datrshare.datr.ws/download/e567e741e54116f488dc7a24a478e498/edpimentl%40gmail.com/AgileSIP-MAC.zip
If you like it... i will create one with a FreeSwitch watermark logo for
I would love to have a FreeSWITCH based softphone for all three platforms... I
just feel a project like that would be kick ass.
Must work on 32bit and 64bit of Windows, Mac and Linux ... and not suck like
most softphones do.
/b
On Dec 29, 2009, at 2:08 PM, EdPimentl wrote:
Add me to the
When I uncomment the following tag, internally held calls no longer hear
MOH.
param name=inbound-bypass-media value=true/
Is there a way to have the above uncommented and still provide MOH to held
calls?
Best Regards,
Jerry
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FreeSWITCH-users
Sounds cool to me also.
There is one softphone for OSX that doesn't suck. It's called
JackenIAX.
A universal softphone sounds cool, but not if it's dumbed down with
regards to it's OSX integration.
Thanks to all,
Marty
On Dec 29, 2009, at 12:34 PM, Brian West wrote:
I would love to
param name=media-option value=resume-media-on-hold/
But it doesn't go back to bypass after Maybe you can post a bounty for
that functionality.
/b
On Dec 29, 2009, at 2:42 PM, Jerry Richards wrote:
When I uncomment the following tag, internally held calls no longer hear
MOH.
param
Does it only do IAX? If so we'll need someone to re-write an IAX2 stack since
the libiax2 from Digium is no longer updated to keep pace with Asterisk and is
now incompatible. Which is the main reason we are thinking about dropping IAX
support unless someone writes a license compatible lib or
I have two inbound event socket sessions.
I send a custom event like this on one socket:
sendevent CUSTOM
Event-Subclass: wpbx::Bcsm
Bcsm-Operation: Bcsm-Event
Bcsm-Uuid: fd96fe6f-ad88-4882-8691-4f849074554d
Bcsm-Peer-Uuid:
I am using iSoftPhone, works great with FreeSWITCH.
Ivan
Den 29. des. 2009 kl. 22.14 skrev Brian West:
Does it only do IAX? If so we'll need someone to re-write an IAX2 stack
since the libiax2 from Digium is no longer updated to keep pace with Asterisk
and is now incompatible. Which is
Guessing the biggest issue is I want to create a softphone project using
FreeSWITCH as the core of the project... is this something people would be
interested in joining?
/b
On Dec 29, 2009, at 3:35 PM, Ivan C Myrvold wrote:
I am using iSoftPhone, works great with FreeSWITCH.
Ivan
Den
Yes, I am for this. Happy to join such a project.
Ivan
Den 29. des. 2009 kl. 22.45 skrev Brian West:
Guessing the biggest issue is I want to create a softphone project using
FreeSWITCH as the core of the project... is this something people would be
interested in joining?
/b
On Dec
Anyone have access to these phones? Two of them if possible and provisioning
information?
Thanks,
Brian
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I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version 1.0.5pre9 and the
bug is still present. Would libpri possibly help? I'm currently using the
native wanpipe PRI stack and default openzap configs in Freeswitch.
Best Regards,
Jerry
_
From: Anthony Minessale
try these drivers:
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.8.7.smg_pri.4.tgz
Mike
On Dec 29, 2009, at 6:17 PM, Jerry Richards wrote:
I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version 1.0.5pre9 and the
bug is still present. Would libpri possibly help? I'm currently
Sounds like a plan to me... who wants to take the lead on the project... we'll
host it.. setup SVN, provide jira access, fisheye and wiki space...
/b
On Dec 29, 2009, at 6:44 PM, João Mesquita wrote:
Why don't we evolve FSGui to be a softphone? I could use a couple of
experienced
I would be happy to provide project managent, user testing and ui
design to this effort.
Lon
Sent from my iPhone
On Dec 29, 2009, at 5:50 PM, Brian West br...@freeswitch.org wrote:
You should join IRC and join in JM and really start the official
softphone project.
/b
On Dec 29, 2009,
Would someone please take a look at this simple PHP event socket script and
tell me what I am doing wrong - or tell me that this could be a bug
elsewhere? Any help would be appreciated.
When I run the script without the call to execute(), everything seems fine.
When I include the call to
Hi
I just want to know what should be the approx configuration of the server
for 50 concurrent call sessions having 3000-4000 users.
Regards
--
View this message in context:
http://n2.nabble.com/Server-Configuration-for-50-concurrent-sessions-tp4231122p4231122.html
Sent from the
Amplify Query... not enough data to make a logical compilation of requested
data.
/b
On Dec 30, 2009, at 12:17 AM, Sharad wrote:
Hi
I just want to know what should be the approx configuration of the server
for 50 concurrent call sessions having 3000-4000 users.
Regards
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