Hello Community, The problem is that a 3-rd party SIP hardware causes audio isssues in less then 1% of the cases. Need to do 100's of manual test calls to identify it and bring to the vendor. Is there a quick recepie to make the following FS test setup? Should be as simple as possible. If someone has something ready-made to offer, please contact me off-list. Here is the idea: -FS generates test SIP call and keeps it up for N minutes. -One side initates the call and sends test tone at X Hz -another side (also FS) answers the call and sends test tone at Y Hz -both sides listen to the tone coming from the other side. -If tone in either direction degrades into siginificant noise or silence, FS leaves the call up indifentely long AND sends email notification "got the sucker!" -If in N minutes tone in both directions did not degrade, FS hangs up the test call and initates a new one. Infinite testing cycle until we turn it off. Thank you, Victor
should be quite easy to do, probably a day of scripting. The only issue, how to measure the RTP quality in real time. ________________________________ From: Victor Chukalovskiy <victor.chukalovskiy@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Friday, September 27, 2013 6:57 PM Subject: [Freeswitch-users] Help needed with audio test setup Hello Community, The problem is that a 3-rd party SIP hardware causes audio isssues in less then 1% of the cases. Need to do 100's of manual test calls to identify it and bring to the vendor. Is there a quick recepie to make the following FS test setup? Should be as simple as possible. If someone has something ready-made to offer, please contact me off-list. Here is the idea: -FS generates test SIP call and keeps it up for N minutes. -One side initates the call and sends test tone at X Hz -another side (also FS) answers the call and sends test tone at Y Hz -both sides listen to the tone coming from the other side. -If tone in either direction degrades into siginificant noise or silence, FS leaves the call up indifentely long AND sends email notification "got the sucker!" -If in N minutes tone in both directions did not degrade, FS hangs up the test call and initates a new one. Infinite testing cycle until we turn it off. Thank you, Victor _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
Clarification: No need to measure RTP quality. Just need to detect when a test tone changes into white noise or dead air. I see mod_spandsp can fire events on tone detection. Is there a way to do inverse? That is, fire event on tone loss? On 13-09-27 01:22 PM, Stanislav Sinyagin wrote:
should be quite easy to do, probably a day of scripting. The only issue, how to measure the RTP quality in real time.
------------------------------------------------------------------------ *From:* Victor Chukalovskiy <victor.chukalovskiy@gmail.com> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Friday, September 27, 2013 6:57 PM *Subject:* [Freeswitch-users] Help needed with audio test setup
Hello Community,
The problem is that a 3-rd party SIP hardware causes audio isssues in less then 1% of the cases. Need to do 100's of manual test calls to identify it and bring to the vendor.
Is there a quick recepie to make the following FS test setup? Should be as simple as possible. If someone has something ready-made to offer, please contact me off-list.
Here is the idea:
-FS generates test SIP call and keeps it up for N minutes. -One side initates the call and sends test tone at X Hz -another side (also FS) answers the call and sends test tone at Y Hz -both sides listen to the tone coming from the other side. -If tone in either direction degrades into siginificant noise or silence, FS leaves the call up indifentely long AND sends email notification "got the sucker!" -If in N minutes tone in both directions did not degrade, FS hangs up the test call and initates a new one. Infinite testing cycle until we turn it off.
Thank you, Victor
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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 <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_wait_for_silence On Sep 27, 2013, at 2:34 PM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com> wrote:
Clarification:
No need to measure RTP quality. Just need to detect when a test tone changes into white noise or dead air.
I see mod_spandsp can fire events on tone detection. Is there a way to do inverse? That is, fire event on tone loss?
On 13-09-27 01:22 PM, Stanislav Sinyagin wrote:
should be quite easy to do, probably a day of scripting. The only issue, how to measure the RTP quality in real time.
From: Victor Chukalovskiy <victor.chukalovskiy@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Friday, September 27, 2013 6:57 PM Subject: [Freeswitch-users] Help needed with audio test setup
Hello Community,
The problem is that a 3-rd party SIP hardware causes audio isssues in less then 1% of the cases. Need to do 100's of manual test calls to identify it and bring to the vendor.
Is there a quick recepie to make the following FS test setup? Should be as simple as possible. If someone has something ready-made to offer, please contact me off-list.
Here is the idea:
-FS generates test SIP call and keeps it up for N minutes. -One side initates the call and sends test tone at X Hz -another side (also FS) answers the call and sends test tone at Y Hz -both sides listen to the tone coming from the other side. -If tone in either direction degrades into siginificant noise or silence, FS leaves the call up indifentely long AND sends email notification "got the sucker!" -If in N minutes tone in both directions did not degrade, FS hangs up the test call and initates a new one. Infinite testing cycle until we turn it off.
Thank you, Victor
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
Hmm, this may work. Does it generate a channel variable once complete? Also, I see hits a counted in frames. What are the units? 20 ms frame or something else? On 13-09-27 02:57 PM, Michael Jerris wrote:
https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_wait_for_silence
On Sep 27, 2013, at 2:34 PM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com <mailto:victor.chukalovskiy@gmail.com>> wrote:
Clarification:
No need to measure RTP quality. Just need to detect when a test tone changes into white noise or dead air.
I see mod_spandsp can fire events on tone detection. Is there a way to do inverse? That is, fire event on tone loss?
On 13-09-27 01:22 PM, Stanislav Sinyagin wrote:
should be quite easy to do, probably a day of scripting. The only issue, how to measure the RTP quality in real time.
------------------------------------------------------------------------ *From:* Victor Chukalovskiy <victor.chukalovskiy@gmail.com> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Friday, September 27, 2013 6:57 PM *Subject:* [Freeswitch-users] Help needed with audio test setup
Hello Community,
The problem is that a 3-rd party SIP hardware causes audio isssues in less then 1% of the cases. Need to do 100's of manual test calls to identify it and bring to the vendor.
Is there a quick recepie to make the following FS test setup? Should be as simple as possible. If someone has something ready-made to offer, please contact me off-list.
Here is the idea:
-FS generates test SIP call and keeps it up for N minutes. -One side initates the call and sends test tone at X Hz -another side (also FS) answers the call and sends test tone at Y Hz -both sides listen to the tone coming from the other side. -If tone in either direction degrades into siginificant noise or silence, FS leaves the call up indifentely long AND sends email notification "got the sucker!" -If in N minutes tone in both directions did not degrade, FS hangs up the test call and initates a new one. Infinite testing cycle until we turn it off.
Thank you, Victor
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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 <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
http://www.freeswitch.org/node/297 On Friday, September 27, 2013, Victor Chukalovskiy wrote:
Hmm, this may work. Does it generate a channel variable once complete?
Also, I see hits a counted in frames. What are the units? 20 ms frame or something else?
On 13-09-27 02:57 PM, Michael Jerris wrote:
https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_wait_for_silence
On Sep 27, 2013, at 2:34 PM, Victor Chukalovskiy < victor.chukalovskiy@gmail.com> wrote:
Clarification:
No need to measure RTP quality. Just need to detect when a test tone changes into white noise or dead air.
I see mod_spandsp can fire events on tone detection. Is there a way to do inverse? That is, fire event on tone loss?
On 13-09-27 01:22 PM, Stanislav Sinyagin wrote:
should be quite easy to do, probably a day of scripting. The only issue, how to measure the RTP quality in real time.
------------------------------ *From:* Victor Chukalovskiy <victor.chukalovskiy@gmail.com> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Friday, September 27, 2013 6:57 PM *Subject:* [Freeswitch-users] Help needed with audio test setup
Hello Community,
The problem is that a 3-rd party SIP hardware causes audio isssues in less then 1% of the cases. Need to do 100's of manual test calls to identify it and bring to the vendor.
Is there a quick recepie to make the following FS test setup? Should be as simple as possible. If someone has something ready-made to offer, please contact me off-list.
Here is the idea:
-FS generates test SIP call and keeps it up for N minutes. -One side initates the call and sends test tone at X Hz -another side (also FS) answers the call and sends test tone at Y Hz -both sides listen to the tone coming from the other side. -If tone in either direction degrades into siginificant noise or silence, FS leaves the call up indifentely long AND sends email notification "got the sucker!" -If in N minutes tone in both directions did not degrade, FS hangs up the test call and initates a new one. Infinite testing cycle until we turn it off.
Thank you, Victor
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: co
-- Sent from mobile device
This would be quite easy to throw together, and would be a fun task for you to take on :) Once you've finished it, you should consider sharing it with the community as I've seen this question come up a few times, it would be nice to give others this functionality out of the box. Let us know how you get on Cal On Fri, Sep 27, 2013 at 5:57 PM, Victor Chukalovskiy < victor.chukalovskiy@gmail.com> wrote:
Hello Community,
The problem is that a 3-rd party SIP hardware causes audio isssues in less then 1% of the cases. Need to do 100's of manual test calls to identify it and bring to the vendor.
Is there a quick recepie to make the following FS test setup? Should be as simple as possible. If someone has something ready-made to offer, please contact me off-list.
Here is the idea:
-FS generates test SIP call and keeps it up for N minutes. -One side initates the call and sends test tone at X Hz -another side (also FS) answers the call and sends test tone at Y Hz -both sides listen to the tone coming from the other side. -If tone in either direction degrades into siginificant noise or silence, FS leaves the call up indifentely long AND sends email notification "got the sucker!" -If in N minutes tone in both directions did not degrade, FS hangs up the test call and initates a new one. Infinite testing cycle until we turn it off.
Thank you, Victor
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
Hi, It's done and works for me. Was a fun thing to do indeed! Dialplan portion of it is documented here in the very bottom: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect#Samples The controlling script portion is in bash and along these lines -query if old test calls are already up with "show calls". kill any old test calls with "uuid_kill" -launch N new test calls using "originate" -wait 3 minutes, check with "show calls" if all test calls ended succesfully. If so, start the test routine again. -if not, stuck test call means attention required. Get call stats with "uuid_dump", run tcpdump on the rtp port and IPs of that call, email result. Hope this helps! -Victor On 13-09-30 04:13 AM, Cal Leeming [Simplicity Media Ltd] wrote:
This would be quite easy to throw together, and would be a fun task for you to take on :)
Once you've finished it, you should consider sharing it with the community as I've seen this question come up a few times, it would be nice to give others this functionality out of the box.
Let us know how you get on
Cal
On Fri, Sep 27, 2013 at 5:57 PM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com <mailto:victor.chukalovskiy@gmail.com>> wrote:
Hello Community,
The problem is that a 3-rd party SIP hardware causes audio isssues in less then 1% of the cases. Need to do 100's of manual test calls to identify it and bring to the vendor.
Is there a quick recepie to make the following FS test setup? Should be as simple as possible. If someone has something ready-made to offer, please contact me off-list.
Here is the idea:
-FS generates test SIP call and keeps it up for N minutes. -One side initates the call and sends test tone at X Hz -another side (also FS) answers the call and sends test tone at Y Hz -both sides listen to the tone coming from the other side. -If tone in either direction degrades into siginificant noise or silence, FS leaves the call up indifentely long AND sends email notification "got the sucker!" -If in N minutes tone in both directions did not degrade, FS hangs up the test call and initates a new one. Infinite testing cycle until we turn it off.
Thank you, Victor
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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 http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
Thanks very much for sharing, glad you were able to get it sorted and hopefully this helps others in the future! Cal On Mon, Sep 30, 2013 at 5:37 PM, Victor Chukalovskiy < victor.chukalovskiy@gmail.com> wrote:
Hi,
It's done and works for me. Was a fun thing to do indeed! Dialplan portion of it is documented here in the very bottom:
https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect#Samples
The controlling script portion is in bash and along these lines -query if old test calls are already up with "show calls". kill any old test calls with "uuid_kill" -launch N new test calls using "originate" -wait 3 minutes, check with "show calls" if all test calls ended succesfully. If so, start the test routine again. -if not, stuck test call means attention required. Get call stats with "uuid_dump", run tcpdump on the rtp port and IPs of that call, email result.
Hope this helps!
-Victor
On 13-09-30 04:13 AM, Cal Leeming [Simplicity Media Ltd] wrote:
This would be quite easy to throw together, and would be a fun task for you to take on :)
Once you've finished it, you should consider sharing it with the community as I've seen this question come up a few times, it would be nice to give others this functionality out of the box.
Let us know how you get on
Cal
On Fri, Sep 27, 2013 at 5:57 PM, Victor Chukalovskiy < victor.chukalovskiy@gmail.com> wrote:
Hello Community,
The problem is that a 3-rd party SIP hardware causes audio isssues in less then 1% of the cases. Need to do 100's of manual test calls to identify it and bring to the vendor.
Is there a quick recepie to make the following FS test setup? Should be as simple as possible. If someone has something ready-made to offer, please contact me off-list.
Here is the idea:
-FS generates test SIP call and keeps it up for N minutes. -One side initates the call and sends test tone at X Hz -another side (also FS) answers the call and sends test tone at Y Hz -both sides listen to the tone coming from the other side. -If tone in either direction degrades into siginificant noise or silence, FS leaves the call up indifentely long AND sends email notification "got the sucker!" -If in N minutes tone in both directions did not degrade, FS hangs up the test call and initates a new one. Infinite testing cycle until we turn it off.
Thank you, Victor
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
participants (5)
-
Cal Leeming [Simplicity Media Ltd]
-
Kristian Kielhofner
-
Michael Jerris
-
Stanislav Sinyagin
-
Victor Chukalovskiy