Welcome to telephony! Glad to hear that you learned about ptime and a few other tricks. There are plenty of new things to learn when it comes to VoIP and FreeSWITCH. Keep up the good work.
-MC

P.S. - you'd better get a sandbox/test server if you're going to be putting 1.5.x into production.


On Tue, Apr 23, 2013 at 9:50 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:

Sorry Dmitry.  I probably did, but 10 hours ago I didn’t understand what ptime was or how to add more default codecs.  It’s not just that I didn’t understand the problem, I didn’t understand how to do the fixes that were proposed.  I do certainly appreciate you help.

 

There are so many new things for me in SIP/FS/VOIP, I am learning though.  Thanks for your patients.

 

Sean

 

 

 

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Dmitry Lysenko
Sent: Tuesday, April 23, 2013 11:49 AM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] URGENT HELP! Customers cant make any calls -after MAKE CURRENT

 

Sean, I posted message about this workaround 10 hours ago. You didn't see it?

 

Dmitry.

 

2013/4/23 Sean Devoy <sdevoy@bizfocused.com>

Ha, Sergey where were you at midnight last night?

I now understand the issue, and even what ptime is, and your fix would have worked perfectly.

 

Linksys and Cisco devices all seem to default to RTP Packet Size:  0.030 and FS defaults to 0.020 (20ms).  The difference is that the current head does not automatically adjust so I needed to specify the 30ms Codecs in defaults.  The codecs are loaded, just not identified as valid defaults.

 

Thank you to everyone who helped or at least tried.

Sean

 

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sergey Okhapkin
Sent: Tuesday, April 23, 2013 9:53 AM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] URGENT HELP! Customers cant make any calls -after MAKE CURRENT

 

As a workaround I added "PCMU@8000h@30i,G729@8000h@30i" to global_codec_prefs.



It looks like PCMU in codec list now means PCMU@20i instead of expected PCMU@anyi

On 04/23/13 09:34, Jeff Lenk wrote:

There seems to be a possible behavior change between head a stable with this.
I opened a potential bug report to discuss this with the team.
http://jira.freeswitch.org:8080/browse/FS-5331
 
 
 
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View this message in context: http://freeswitch-users.2379917.n2.nabble.com/URGENT-HELP-Customers-cant-make-any-calls-after-MAKE-CURRENT-tp7589980p7590009.html
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_________________________________________________________________________
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
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 


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




--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org