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@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...., 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!