[Freeswitch-users] External voicemail issue

Thu Mar 21 16:03:00 MSK 2013

Apologies, all to easy to hit send before the message is finished! What I
meant to say was....

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

Setting *send_silence_when_idle *to true helped with the clipping but
didn't resolve it. I purchased a G729 codec license and this eliminated the
clipping but I'm still getting digital noise during silence after the vm
beep (this is without send silence on though)

I've extracted a call from the destination call recorder to illustrate the
problem, the digital noise is lost in the record but you can hear the
clipping https://docs.google.com/file/d/0B5Bx5sEYH7abc0FObURFWm5HNzg/edit

At this stage I'm assuming its the Skype connect trunk and my next move is
to try another provider.

If anyone thinks otherwise let me know, I just thought I'd give an update
in case it is of any use to people facing the same issue in the future.

On Thu, Mar 21, 2013 at 12:52 PM, Elliot Farmer
<elliotfarmer101 at gmail.com>wrote:

> 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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>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>> CudaTel Communication Server  Official
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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
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>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
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>>>>
>>>> _________________________________________________________________________
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>>>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
>>>> CudaTel Communication Server  Official
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>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> 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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