[Freeswitch-users] External voicemail issue

[Yuriy Nasida]

Did you add this line right after playing of 2 wav files you said ? <action application="voicemail" data="default $${domain} $1"/>
Also you can check http://wiki.freeswitch.org/wiki/ZRTP#Step_2

Date: Tue, 19 Mar 2013 15:18:12 +0000
From: elliotfarmer101 at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] External voicemail issue

>From a SIP handset directly to FreeSwitch voicemail works perfectly, although I haven't tested from outside the LAN transitioning the external firewall, I'll give that a try.

I'm not sure about zrtp but I've changed the dialplan as you suggested to remove the loopback already. I'm using <action application="voicemail" data="default $${domain} $1"/>
instead of the loopback but the issue is still there.




On Tue, Mar 19, 2013 at 3:07 PM, Yuriy Nasida <nasida at live.ru> wrote:




Also looks like that you use  zrtp.
Um.. Skype + loopback + zrtp + TRANSCODING.I can just advise you to simplify your setup and look that is the cause.
Can you try with SIP-to-SIP directly to voicemail and without any additional things  ?
From: nasida at live.ru
To: freeswitch-users at lists.freeswitch.org

Date: Tue, 19 Mar 2013 18:24:51 +0400
Subject: Re: [Freeswitch-users] External voicemail issue




http://wiki.freeswitch.org/wiki/Loopback
You can look at your dial plan and look for 'loopback'.
the loopback endpoint affects to voice and if you really use this it can be the reason of your issue.In this case in my opinion a redoing of your routing logic without the loopback endpoint will be good idea.


Date: Tue, 19 Mar 2013 13:55:34 +0000
From: elliotfarmer101 at gmail.com
To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] External voicemail issue

Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm using the default config that has loopback setup as the route to voicemail as far as I can work out.


Is there a way of sending to voicemail without a loopback?



On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida <nasida at live.ru> wrote:




looked at your logs. Do you use loopback endpoint in your dialplan ? If yes, can you check without  this ?


Date: Tue, 19 Mar 2013 12:37:47 +0000
From: elliotfarmer101 at gmail.com
To: freeswitch-users at lists.freeswitch.org


Subject: Re: [Freeswitch-users] External voicemail issue

I've done some further testing and the issue appears to be isolated to the voicemail system.


To test I configured the inbound dialplan to play two wav files before transferring the call...


1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav
2) /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav



Both play fine, as soon as the call is transferred to voicemail (via extension 1000 that is not registered at the time) the quality drops and I start loosing speech.  I've tried preferring the PCMU codec within vars but I seem to have a lot of 
"TRANSCODING_NECESSARY" messages, they are not isolated to the 
voicemail steps however.

Has anyone got an idea where I can start to look? Any help would be much appreciated.



On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer <elliotfarmer101 at gmail.com> wrote:



Hi all,

I'm having an issue with the voicemail system when accessing via a Skype connect trunk. 

Internally all is working, however if I call my external Skype number from a PSTN line when the call is routed to voicemail I get a lot of intermittent speech loss during the IVR announcement and then after the beep I can hear odd sounds that I can only describe as digital interference type sounds. Also the IVR is very quick to say "the recording is below the minimum length" although if I speak quickly I can record a voicemail and it sounds fine during playback, not amazing audio quality but no major issues.





The issue seems to be the same as http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, I have tried recording a greeting but the same issues occur.





I'm running Freeswitch on CentOS 6.3 on a physical machine.

Here are some logs http://pastebin.com/jmuks47q , I've tried to remove the phone numbers and IP addresses as I don't have permission from the owners to publish them on the internet.





Thanks in advance for your help! 






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