[Freeswitch-users] External voicemail issue

[Elliot Farmer]

No matter what I try I can't seem to get rid of these audio problems. Are
there any known issues with Skype connect that anyone knows about?

I'm thinking of signing up with another SIP provider to test but I don't
know of any other providers that offer pay as you go like Skype. Also as
calls are fine I'm led to believe its not the Skype trunk at fault so don't
want to waste money if I don't have to.

Is it worth installing on CentOS 5? The wiki lists some issues with 6
although I thought I would be safe on 6.3

Sorry to be asking so many questions, could just do with some pointers


On Wed, Mar 20, 2013 at 11:36 AM, Elliot Farmer
<elliotfarmer101 at gmail.com>wrote:

> Ok more testing and changes, still no closer.
>
> I tested accessing the voicemail system externally through the firewall
> with a SIP phone and the IVR is perfectly clear, no problems.
>
> I then disabled ZRTP by running this command "global_setvar
> zrtp_secure_media=false" and from what I make of the log it worked however
> the issue remains.
>
> My dialplan entry looks like this...
>
>     <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(10[01][0-9])$">
>         <action application="export" data="dialed_extension=$1"/>
>         <action application="playback"
> data="/home/user1/Downloads/8kulaw.wav"/>
>
>         <action application="voicemail" data="default $${domain} $1"/>
>       </condition>
>     </extension>
>
> As I said before the wav plays fine then when routed to voiemail the audio
> issues occur.
>
> Another paste http://pastebin.com/ZfWCzKYb
>
> Thanks!
>
>
>
>
>
>
> On Tue, Mar 19, 2013 at 3:40 PM, Yuriy Nasida <nasida at live.ru> wrote:
>
>> Did you add this line right after playing of 2 wav files you said ?
>>  <action application="voicemail" data="default $${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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>>
>> 
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>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
>> CudaTel Communication Server  Official FreeSWITCH
>> Sites http://www.freeswitch.org http://wiki.freeswitch.org
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>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
>> CudaTel Communication Server  Official FreeSWITCH
>> Sites http://www.freeswitch.org http://wiki.freeswitch.org
>> http://www.cluecon.com FreeSWITCH-users mailing list
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>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
>> CudaTel Communication Server  Official FreeSWITCH
>> Sites http://www.freeswitch.org http://wiki.freeswitch.org
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>> FreeSWITCH-users at lists.freeswitch.org
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>> http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
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>
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