[Freeswitch-users] External voicemail issue

Wed Mar 20 14:36:14 MSK 2013

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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> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel
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> http://www.freeswitch.org http://wiki.freeswitch.org
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>
> 
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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