External voicemail issue
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!
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!
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@gmail.com To: freeswitch-users@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@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! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@gmail.com To: freeswitch-users@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@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!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@gmail.com To: freeswitch-users@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@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@gmail.com To: freeswitch-users@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@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! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@live.ru To: freeswitch-users@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@gmail.com To: freeswitch-users@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@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@gmail.com To: freeswitch-users@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@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! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@live.ru To: freeswitch-users@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@gmail.com To: freeswitch-users@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@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@gmail.com To: freeswitch-users@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@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!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@gmail.com To: freeswitch-users@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@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@live.ru To: freeswitch-users@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@gmail.com To: freeswitch-users@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@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@gmail.com To: freeswitch-users@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@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! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@live.ru> wrote:
Did you add this line right after playing of 2 wav files you said ? <action application="voicemail" data="default $${domain} $1"/>
Also you can check http://wiki.freeswitch.org/wiki/ZRTP#Step_2
------------------------------ Date: Tue, 19 Mar 2013 15:18:12 +0000
From: elliotfarmer101@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] External voicemail issue
From a SIP handset directly to FreeSwitch voicemail works perfectly, although I haven't tested from outside the LAN transitioning the external firewall, I'll give that a try.
I'm not sure about zrtp but I've changed the dialplan as you suggested to remove the loopback already. I'm using <action application="voicemail" data="default $${domain} $1"/> instead of the loopback but the issue is still there.
On Tue, Mar 19, 2013 at 3:07 PM, Yuriy Nasida <nasida@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@live.ru To: freeswitch-users@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@gmail.com To: freeswitch-users@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@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@gmail.com To: freeswitch-users@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@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!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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 $${domain} $1"/> </condition> </extension>
As I said before the wav plays fine then when routed to voiemail the audio issues occur.
Another paste http://pastebin.com/ZfWCzKYb
Thanks!
On Tue, Mar 19, 2013 at 3:40 PM, Yuriy Nasida <nasida@live.ru> wrote:
Did you add this line right after playing of 2 wav files you said ? <action application="voicemail" data="default $${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@gmail.com To: freeswitch-users@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@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@live.ru To: freeswitch-users@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@gmail.com To: freeswitch-users@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@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@gmail.com To: freeswitch-users@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@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!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@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@live.ru> wrote:
Did you add this line right after playing of 2 wav files you said ? <action application="voicemail" data="default $${domain} $1"/>
Also you can check http://wiki.freeswitch.org/wiki/ZRTP#Step_2
------------------------------ Date: Tue, 19 Mar 2013 15:18:12 +0000
From: elliotfarmer101@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] External voicemail issue
From a SIP handset directly to FreeSwitch voicemail works perfectly, although I haven't tested from outside the LAN transitioning the external firewall, I'll give that a try.
I'm not sure about zrtp but I've changed the dialplan as you suggested to remove the loopback already. I'm using <action application="voicemail" data="default $${domain} $1"/> instead of the loopback but the issue is still there.
On Tue, Mar 19, 2013 at 3:07 PM, Yuriy Nasida <nasida@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@live.ru To: freeswitch-users@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@gmail.com To: freeswitch-users@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@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@gmail.com To: freeswitch-users@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@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!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@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@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 $${domain} $1"/> </condition> </extension>
As I said before the wav plays fine then when routed to voiemail the audio issues occur.
Another paste http://pastebin.com/ZfWCzKYb
Thanks!
On Tue, Mar 19, 2013 at 3:40 PM, Yuriy Nasida <nasida@live.ru> wrote:
Did you add this line right after playing of 2 wav files you said ? <action application="voicemail" data="default $${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@gmail.com To: freeswitch-users@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@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@live.ru To: freeswitch-users@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@gmail.com To: freeswitch-users@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@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@gmail.com To: freeswitch-users@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@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!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Ok I found how to remove the loopback but still having the same issue. I noticed that after I removed the loopback the hold music that was playing before the voicemail connect was also of poor quality if that means anything? My dialplan now connects the call straight to voicemail so I don't hear ringback or anything but the issue remains. Interestingly the speech breaks up at the same point of the announcement every time, I hear "The person at extension ?000 is ????available, ???? your message at the tone, press any key or stop talking to end the recording. Then some weird digital noise after the beep. Hope this is of some use On Tue, Mar 19, 2013 at 1:55 PM, Elliot Farmer <elliotfarmer101@gmail.com>wrote:
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@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@gmail.com To: freeswitch-users@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@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!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
-
Elliot Farmer
-
Yuriy Nasida