[Freeswitch-users] External voicemail issue

Thu Mar 21 15:52:00 MSK 2013

Ok so with some help from the IRC channel (thanks everyone!) I've managed
to get pcaps of the calls and they sound fine and the max jitter is <7ms.

Setting *send_silence_when_idle
*

On Wed, Mar 20, 2013 at 6:02 PM, Elliot Farmer <elliotfarmer101 at gmail.com>wrote:

> 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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>>>
>>> 
>>> 
>>>
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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
>>> FreeSWITCH-users at lists.freeswitch.org
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>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
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>>>
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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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>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>> FreeSWITCH-users mailing list
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>>>
>>
>
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