Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[Freeswitch-users] Where to put Jitterbuffer?


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users
View previous topic :: View next topic  
Author Message
rajil.s at gmail.com
Guest





PostPosted: Mon Feb 15, 2016 5:46 pm    Post subject: [Freeswitch-users] Where to put Jitterbuffer? Reply with quote

Hello,

I have two FS instances connected to each other over VPN. The VPN
sometimes has poor connectivity and I was wondering whether
jitterbuffer can help to improve the call quality. I am using the OPUS
codec between the FS instances since it is supposed to behave better
in lower bandwidth. The ATA on both ends are Linksys SPA3102.

My connections is as follows:

ATA_A---->FS_A---->vpn---->FS_B---->ATA_B


When user A is calling to B where should i specify jitter buffer
amongst the following:

1. The bridge between Freeswitch instance FS_A and FS_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/>

2. The bridge between Freeswitch instance FS_B and ATA_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/>

Thanks,
Rajil

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.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
Back to top
anthony.minessale at g...
Guest





PostPosted: Mon Feb 15, 2016 6:11 pm    Post subject: [Freeswitch-users] Where to put Jitterbuffer? Reply with quote

The variable must be set on the leg before the channel engages media.you can test if its on with 


uuid_jitterbuffer <uuid> debug:10




if you want it on both legs set the following global vars in vars.xml



rtp_jitter_buffer_during_bridge=true
jitterbuffer_msec=60











On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:
Quote:
Hello,

I have two FS instances connected to each other over VPN. The VPN
sometimes has poor connectivity and I was wondering whether
jitterbuffer can help to improve the call quality. I am using the OPUS
codec between the FS instances since it is supposed to behave better
in lower bandwidth. The ATA on both ends are Linksys SPA3102.

My connections is as follows:

ATA_A---->FS_A---->vpn---->FS_B---->ATA_B


When user A is calling to B where should i specify jitter buffer
amongst the following:

1. The bridge between Freeswitch instance FS_A and FS_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/>

2. The bridge between Freeswitch instance FS_B and ATA_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/>

Thanks,
Rajil

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
http://freeswitch.org/  ☞ http://cluecon.com/  ☞ http://twitter.com/FreeSWITCH
irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+


ClueCon Weekly Development Call 

☎ sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])  ☎ +19193869900 




https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
Back to top
rajil.s at gmail.com
Guest





PostPosted: Mon Feb 15, 2016 6:56 pm    Post subject: [Freeswitch-users] Where to put Jitterbuffer? Reply with quote

The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says,
Quote:
If both sides of a bridge are RTP and both sides have a jb, its fairly
useless. In fact if anything, it can worsen call quality.So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says?


On 15 February 2016 at 17:10, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:
Quote:
The variable must be set on the leg before the channel engages media.you can test if its on with 


uuid_jitterbuffer <uuid> debug:10




if you want it on both legs set the following global vars in vars.xml



rtp_jitter_buffer_during_bridge=true
jitterbuffer_msec=60











On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:


Quote:
Hello,

I have two FS instances connected to each other over VPN. The VPN
sometimes has poor connectivity and I was wondering whether
jitterbuffer can help to improve the call quality. I am using the OPUS
codec between the FS instances since it is supposed to behave better
in lower bandwidth. The ATA on both ends are Linksys SPA3102.

My connections is as follows:

ATA_A---->FS_A---->vpn---->FS_B---->ATA_B


When user A is calling to B where should i specify jitter buffer
amongst the following:

1. The bridge between Freeswitch instance FS_A and FS_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/>

2. The bridge between Freeswitch instance FS_B and ATA_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/>

Thanks,
Rajil



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
http://freeswitch.org/  ☞ http://cluecon.com/  ☞ http://twitter.com/FreeSWITCH
irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+


ClueCon Weekly Development Call 

☎ sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])  ☎ [url=tel:%2B19193869900]+19193869900[/url] 




https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA









_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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
Back to top
anthony.minessale at g...
Guest





PostPosted: Mon Feb 15, 2016 7:24 pm    Post subject: [Freeswitch-users] Where to put Jitterbuffer? Reply with quote

If one or the other end of the call does not have its own JB then putting it on both sides can fix the stream for the other end.That is why its not defaulted to being possible but you asked how to do it.




On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:
Quote:


The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says,
Quote:
If both sides of a bridge are RTP and both sides have a jb, its fairly
useless. In fact if anything, it can worsen call quality.So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says?


On 15 February 2016 at 17:10, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:
Quote:
The variable must be set on the leg before the channel engages media.you can test if its on with 


uuid_jitterbuffer <uuid> debug:10




if you want it on both legs set the following global vars in vars.xml



rtp_jitter_buffer_during_bridge=true
jitterbuffer_msec=60











On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:


Quote:
Hello,

I have two FS instances connected to each other over VPN. The VPN
sometimes has poor connectivity and I was wondering whether
jitterbuffer can help to improve the call quality. I am using the OPUS
codec between the FS instances since it is supposed to behave better
in lower bandwidth. The ATA on both ends are Linksys SPA3102.

My connections is as follows:

ATA_A---->FS_A---->vpn---->FS_B---->ATA_B


When user A is calling to B where should i specify jitter buffer
amongst the following:

1. The bridge between Freeswitch instance FS_A and FS_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/>

2. The bridge between Freeswitch instance FS_B and ATA_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/>

Thanks,
Rajil



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
http://freeswitch.org/  ☞ http://cluecon.com/  ☞ http://twitter.com/FreeSWITCH
irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+


ClueCon Weekly Development Call 

☎ sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])  ☎ [url=tel:%2B19193869900]+19193869900[/url] 




https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA









_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
http://freeswitch.org/  ☞ http://cluecon.com/  ☞ http://twitter.com/FreeSWITCH
irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+


ClueCon Weekly Development Call 

☎ sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])  ☎ +19193869900 




https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
Back to top
rajil.s at gmail.com
Guest





PostPosted: Mon Feb 15, 2016 7:35 pm    Post subject: [Freeswitch-users] Where to put Jitterbuffer? Reply with quote

Ok, what about my specific use case where both endpoints ATA (Linksys spa3102) do have internal jitterbuffer. Should i still set the global variables as you suggested to improve the call quality over vpn? 


On 15 February 2016 at 18:23, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:
Quote:
If one or the other end of the call does not have its own JB then putting it on both sides can fix the stream for the other end.That is why its not defaulted to being possible but you asked how to do it.




On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:
Quote:


The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says,
Quote:
If both sides of a bridge are RTP and both sides have a jb, its fairly
useless. In fact if anything, it can worsen call quality.So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says?


On 15 February 2016 at 17:10, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:
Quote:
The variable must be set on the leg before the channel engages media.you can test if its on with 


uuid_jitterbuffer <uuid> debug:10




if you want it on both legs set the following global vars in vars.xml



rtp_jitter_buffer_during_bridge=true
jitterbuffer_msec=60











On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:


Quote:
Hello,

I have two FS instances connected to each other over VPN. The VPN
sometimes has poor connectivity and I was wondering whether
jitterbuffer can help to improve the call quality. I am using the OPUS
codec between the FS instances since it is supposed to behave better
in lower bandwidth. The ATA on both ends are Linksys SPA3102.

My connections is as follows:

ATA_A---->FS_A---->vpn---->FS_B---->ATA_B


When user A is calling to B where should i specify jitter buffer
amongst the following:

1. The bridge between Freeswitch instance FS_A and FS_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/>

2. The bridge between Freeswitch instance FS_B and ATA_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/>

Thanks,
Rajil



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
http://freeswitch.org/  ☞ http://cluecon.com/  ☞ http://twitter.com/FreeSWITCH
irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+


ClueCon Weekly Development Call 

☎ sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])  ☎ [url=tel:%2B19193869900]+19193869900[/url] 




https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA









_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
http://freeswitch.org/  ☞ http://cluecon.com/  ☞ http://twitter.com/FreeSWITCH
irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+


ClueCon Weekly Development Call 

☎ sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])  ☎ [url=tel:%2B19193869900]+19193869900[/url] 




https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA











_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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
Back to top
anthony.minessale at g...
Guest





PostPosted: Mon Feb 15, 2016 7:47 pm    Post subject: [Freeswitch-users] Where to put Jitterbuffer? Reply with quote

When you keep the codec the same across the whole call path then the jitterbuffer is not as important because FS will preserve the timestamps.
If you are transcoding or recording the call, then the JB can help because you are creating new timestamps so the far end can no longer perceive the jitter.


Ideally you should just try it.
Also you should make sure your vpn is using udp or gprs not tcp 


You should probably try it with everything PCMU or PCMA as well with and without the JB.
Trial and error is your friend.  But without the vars I mentioned it will never use the JB during a bridge.












On Mon, Feb 15, 2016 at 6:33 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:
Quote:
Ok, what about my specific use case where both endpoints ATA (Linksys spa3102) do have internal jitterbuffer. Should i still set the global variables as you suggested to improve the call quality over vpn? 


On 15 February 2016 at 18:23, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:
Quote:
If one or the other end of the call does not have its own JB then putting it on both sides can fix the stream for the other end.That is why its not defaulted to being possible but you asked how to do it.




On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:
Quote:


The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says,
Quote:
If both sides of a bridge are RTP and both sides have a jb, its fairly
useless. In fact if anything, it can worsen call quality.So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says?


On 15 February 2016 at 17:10, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:
Quote:
The variable must be set on the leg before the channel engages media.you can test if its on with 


uuid_jitterbuffer <uuid> debug:10




if you want it on both legs set the following global vars in vars.xml



rtp_jitter_buffer_during_bridge=true
jitterbuffer_msec=60











On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:


Quote:
Hello,

I have two FS instances connected to each other over VPN. The VPN
sometimes has poor connectivity and I was wondering whether
jitterbuffer can help to improve the call quality. I am using the OPUS
codec between the FS instances since it is supposed to behave better
in lower bandwidth. The ATA on both ends are Linksys SPA3102.

My connections is as follows:

ATA_A---->FS_A---->vpn---->FS_B---->ATA_B


When user A is calling to B where should i specify jitter buffer
amongst the following:

1. The bridge between Freeswitch instance FS_A and FS_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/>

2. The bridge between Freeswitch instance FS_B and ATA_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/>

Thanks,
Rajil



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
http://freeswitch.org/  ☞ http://cluecon.com/  ☞ http://twitter.com/FreeSWITCH
irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+


ClueCon Weekly Development Call 

☎ sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])  ☎ [url=tel:%2B19193869900]+19193869900[/url] 




https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA









_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
http://freeswitch.org/  ☞ http://cluecon.com/  ☞ http://twitter.com/FreeSWITCH
irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+


ClueCon Weekly Development Call 

☎ sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])  ☎ [url=tel:%2B19193869900]+19193869900[/url] 




https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA











_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
http://freeswitch.org/  ☞ http://cluecon.com/  ☞ http://twitter.com/FreeSWITCH
irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+


ClueCon Weekly Development Call 

☎ sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])  ☎ +19193869900 




https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services