Bad sound quality and Abnormally large timer gap 100953 detected!
I just set up Freeswitch Precompiled for Windows x64 ( http://files-sync.freeswitch.org/windows/installer/) on a VPS Windows Server 2008. Sound is really bad and I'm getting the following: 2011-07-27 12:33:15.012695 [WARNING] switch_time.c:244 Abnormally large timer gap 100953 detected! Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems. also 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 6 seconds 1 heartbeat (core) 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 2 check_ip (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 1 heartbeat (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 2 check_ip (core) What can I do? Also, yesterday, when I first set it up, it was all good. Thanks, John
You can't count on the time keeping of a vps. Use windows directly on real hardware, not a virtual machine. -giovanni On 7/27/11, Joao Leme <joaocarlosleme@gmail.com> wrote:
I just set up Freeswitch Precompiled for Windows x64 ( http://files-sync.freeswitch.org/windows/installer/) on a VPS Windows Server 2008. Sound is really bad and I'm getting the following:
2011-07-27 12:33:15.012695 [WARNING] switch_time.c:244 Abnormally large timer gap 100953 detected! Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems.
also
2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 6 seconds 1 heartbeat (core) 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 2 check_ip (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 1 heartbeat (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 2 check_ip (core)
What can I do?
Also, yesterday, when I first set it up, it was all good.
Thanks, John
-- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
Thanks! So there is no other way around? On Wed, Jul 27, 2011 at 1:25 PM, Giovanni Maruzzelli <gmaruzz@gmail.com>wrote:
You can't count on the time keeping of a vps.
Use windows directly on real hardware, not a virtual machine.
-giovanni
On 7/27/11, Joao Leme <joaocarlosleme@gmail.com> wrote:
I just set up Freeswitch Precompiled for Windows x64 ( http://files-sync.freeswitch.org/windows/installer/) on a VPS Windows Server 2008. Sound is really bad and I'm getting the following:
2011-07-27 12:33:15.012695 [WARNING] switch_time.c:244 Abnormally large timer gap 100953 detected! Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems.
also
2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 6 seconds 1 heartbeat (core) 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 2 check_ip (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 1 heartbeat (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 2 check_ip (core)
What can I do?
Also, yesterday, when I first set it up, it was all good.
Thanks, John
-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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No, it's never "safe" (not for production at least) to run VoIP on virtual machines. The reason for better results yesterday was probably because the hardware had less total load by that time, so the problem is probably both the virtual machine, plus the hardware has too much load on it. /Peter ________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Joao Leme [joaocarlosleme@gmail.com] Skickat: den 27 juli 2011 22:51 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected! Thanks! So there is no other way around? On Wed, Jul 27, 2011 at 1:25 PM, Giovanni Maruzzelli <gmaruzz@gmail.com<mailto:gmaruzz@gmail.com>> wrote: You can't count on the time keeping of a vps. Use windows directly on real hardware, not a virtual machine. -giovanni On 7/27/11, Joao Leme <joaocarlosleme@gmail.com<mailto:joaocarlosleme@gmail.com>> wrote:
I just set up Freeswitch Precompiled for Windows x64 ( http://files-sync.freeswitch.org/windows/installer/) on a VPS Windows Server 2008. Sound is really bad and I'm getting the following:
2011-07-27 12:33:15.012695 [WARNING] switch_time.c:244 Abnormally large timer gap 100953 detected! Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems.
also
2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 6 seconds 1 heartbeat (core) 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 2 check_ip (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 1 heartbeat (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 2 check_ip (core)
What can I do?
Also, yesterday, when I first set it up, it was all good.
Thanks, John
-- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618<tel:%2B39-347-2665618> _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 !DSPAM:4e307b2032763660377611!
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it. ________________________________ From: Peter Olsson <peter.olsson@visionutveckling.se> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, July 27, 2011 2:06 PM Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected! No, it's never "safe" (not for production at least) to run VoIP on virtual machines. The reason for better results yesterday was probably because the hardware had less total load by that time, so the problem is probably both the virtual machine, plus the hardware has too much load on it. /Peter ________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Joao Leme [joaocarlosleme@gmail.com] Skickat: den 27 juli 2011 22:51 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected! Thanks! So there is no other way around? On Wed, Jul 27, 2011 at 1:25 PM, Giovanni Maruzzelli <gmaruzz@gmail.com<mailto:gmaruzz@gmail.com>> wrote: You can't count on the time keeping of a vps. Use windows directly on real hardware, not a virtual machine. -giovanni On 7/27/11, Joao Leme <joaocarlosleme@gmail.com<mailto:joaocarlosleme@gmail.com>> wrote:
I just set up Freeswitch Precompiled for Windows x64 ( http://files-sync.freeswitch.org/windows/installer/) on a VPS Windows Server 2008. Sound is really bad and I'm getting the following:
2011-07-27 12:33:15.012695 [WARNING] switch_time.c:244 Abnormally large timer gap 100953 detected! Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems.
also
2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 6 seconds 1 heartbeat (core) 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 2 check_ip (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 1 heartbeat (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 2 check_ip (core)
What can I do?
Also, yesterday, when I first set it up, it was all good.
Thanks, John
-- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618<tel:%2B39-347-2665618> _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 !DSPAM:4e307b2032763660377611! _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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
On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote:
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P -MC
I'm running FS under Xen provided by Linode. I only run max 10 concurrent channels in production, but an MOH test sounded good through 200 concurrent channels. Linode says the CPU is the most underused resource on their systems... -Avi On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc@freeswitch.org>wrote:
On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote:
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P
-MC
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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i forgot to mention that the 50 to 60 calls also are doing majority of g729 transcoding as well which does put th cpu to 85 to 90% full utilization. So don't knock virtual machines. ________________________________ From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, July 27, 2011 3:20 PM Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected! I'm running FS under Xen provided by Linode. I only run max 10 concurrent channels in production, but an MOH test sounded good through 200 concurrent channels. Linode says the CPU is the most underused resource on their systems... -Avi On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc@freeswitch.org> wrote:
On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote:
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P
-MC _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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Hey guys 'n gal, the original post is asking for Windows!!! -giovanni On 7/28/11, Sam <lakersman2006@yahoo.com> wrote:
i forgot to mention that the 50 to 60 calls also are doing majority of g729 transcoding as well which does put th cpu to 85 to 90% full utilization. So don't knock virtual machines.
________________________________ From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, July 27, 2011 3:20 PM Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
I'm running FS under Xen provided by Linode. I only run max 10 concurrent channels in production, but an MOH test sounded good through 200 concurrent channels. Linode says the CPU is the most underused resource on their systems...
-Avi
On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc@freeswitch.org> wrote:
On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote:
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P
-MC _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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
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-- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
I feel I need to explain my last statement a little further :) First of all, you can run FS on virtual machines (even though I never recommend it), but you need to be in control of the virtual environment, or totally trust the ones controlling it. There are two basic problems when running on virtual hardware - timing and network I/O. For timing to work you need to make sure that the one controlling the virtual environment NEVER give out more CPU cores then there are available on the hardware. If they give out more, and your machine need to share resources with another machine, you will be in trouble. As soon as the other virtual machine starts to load the CPU, your machine will get bad timing. Also - usually a one core VM will get better timing then a two+ core VM. Also, to be safe on the network side, you need your own NIC, that is not shared with other machines. If you feel you can control this, you might be successful with the setup, if you can't control this, you're either lucky (and things work anyway), or you'll get bad audio. Also, just passing RTP for normal call isn't the most critical issue (you don't even have to use a timer there), the most critical timing issues will occur in conferences or when using a local IVR on the machine. I've done lots of testing for this in my lab, since many of my customers wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual hosts, and I've found the above to be quite critical to succeed with your virtual implementation. /Peter ____________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Giovanni Maruzzelli [gmaruzz@gmail.com] Skickat: den 28 juli 2011 09:37 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected! Hey guys 'n gal, the original post is asking for Windows!!! -giovanni On 7/28/11, Sam <lakersman2006@yahoo.com> wrote:
i forgot to mention that the 50 to 60 calls also are doing majority of g729 transcoding as well which does put th cpu to 85 to 90% full utilization. So don't knock virtual machines.
________________________________ From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, July 27, 2011 3:20 PM Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
I'm running FS under Xen provided by Linode. I only run max 10 concurrent channels in production, but an MOH test sounded good through 200 concurrent channels. Linode says the CPU is the most underused resource on their systems...
-Avi
On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc@freeswitch.org> wrote:
On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote:
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P
-MC _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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-- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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 !DSPAM:4e31126d32765770742302!
Peter, that's a pretty clear explanation, and answers many frequently asked questions. Can you please add it to the wiki (eg: "FS on virtual machines", and add links to it from "amazon" and "aws" related pages) ? I'm sure lot of people will find it interesting, and hopefully some other people (and/or you) will add more info and experiences to it. -giovanni On 7/28/11, Peter Olsson <peter.olsson@visionutveckling.se> wrote:
I feel I need to explain my last statement a little further :)
First of all, you can run FS on virtual machines (even though I never recommend it), but you need to be in control of the virtual environment, or totally trust the ones controlling it.
There are two basic problems when running on virtual hardware - timing and network I/O. For timing to work you need to make sure that the one controlling the virtual environment NEVER give out more CPU cores then there are available on the hardware. If they give out more, and your machine need to share resources with another machine, you will be in trouble. As soon as the other virtual machine starts to load the CPU, your machine will get bad timing. Also - usually a one core VM will get better timing then a two+ core VM.
Also, to be safe on the network side, you need your own NIC, that is not shared with other machines.
If you feel you can control this, you might be successful with the setup, if you can't control this, you're either lucky (and things work anyway), or you'll get bad audio.
Also, just passing RTP for normal call isn't the most critical issue (you don't even have to use a timer there), the most critical timing issues will occur in conferences or when using a local IVR on the machine.
I've done lots of testing for this in my lab, since many of my customers wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual hosts, and I've found the above to be quite critical to succeed with your virtual implementation.
/Peter
____________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Giovanni Maruzzelli [gmaruzz@gmail.com] Skickat: den 28 juli 2011 09:37 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
Hey guys 'n gal,
the original post is asking for Windows!!!
-giovanni
On 7/28/11, Sam <lakersman2006@yahoo.com> wrote:
i forgot to mention that the 50 to 60 calls also are doing majority of g729 transcoding as well which does put th cpu to 85 to 90% full utilization. So don't knock virtual machines.
________________________________ From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, July 27, 2011 3:20 PM Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
I'm running FS under Xen provided by Linode. I only run max 10 concurrent channels in production, but an MOH test sounded good through 200 concurrent channels. Linode says the CPU is the most underused resource on their systems...
-Avi
On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc@freeswitch.org> wrote:
On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote:
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P
-MC _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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
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-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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
!DSPAM:4e31126d32765770742302!
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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-- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
Peter, thanks for those details.
What sort of trade-offs, if any, are useful to improve audio for conferences and IVRs on VMs? Will increasing the interval setting for a conference help? What about using larger frames for codecs? I know that latency, beyond a point, is a serious quality issue of its own. Still, do you know what compromises provide the best improvement? Bryan
On 7/28/11, Peter Olsson <peter.olsson@visionutveckling.se> wrote:
I feel I need to explain my last statement a little further :)
First of all, you can run FS on virtual machines (even though I never recommend it), but you need to be in control of the virtual environment, or totally trust the ones controlling it.
There are two basic problems when running on virtual hardware - timing and network I/O. For timing to work you need to make sure that the one controlling the virtual environment NEVER give out more CPU cores then there are available on the hardware. If they give out more, and your machine need to share resources with another machine, you will be in trouble. As soon as the other virtual machine starts to load the CPU, your machine will get bad timing. Also - usually a one core VM will get better timing then a two+ core VM.
Also, to be safe on the network side, you need your own NIC, that is not shared with other machines.
If you feel you can control this, you might be successful with the setup, if you can't control this, you're either lucky (and things work anyway), or you'll get bad audio.
Also, just passing RTP for normal call isn't the most critical issue (you don't even have to use a timer there), the most critical timing issues will occur in conferences or when using a local IVR on the machine.
I've done lots of testing for this in my lab, since many of my customers wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual hosts, and I've found the above to be quite critical to succeed with your virtual implementation.
/Peter
____________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Giovanni Maruzzelli [gmaruzz@gmail.com] Skickat: den 28 juli 2011 09:37 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
Hey guys 'n gal,
the original post is asking for Windows!!!
-giovanni
On 7/28/11, Sam <lakersman2006@yahoo.com> wrote:
i forgot to mention that the 50 to 60 calls also are doing majority of g729 transcoding as well which does put th cpu to 85 to 90% full utilization. So don't knock virtual machines.
________________________________ From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, July 27, 2011 3:20 PM Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
I'm running FS under Xen provided by Linode. I only run max 10 concurrent channels in production, but an MOH test sounded good through 200 concurrent channels. Linode says the CPU is the most underused resource on their systems...
-Avi
On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc@freeswitch.org> wrote:
On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote:
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P
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It all depends really. On some solutions (depending on hypervisor, hardware and guest OS) the main problem can be that 20ms is a too short period to sleep, in that case a larger frame (ie 40ms) might help out. But mostly the problem is that the clock is not accurate enough, and fluxtuates from time to time (sometimes 20ms is 10, next round 30 etc). After my latest testing I've found out that latest ESXi (4.1) works quite good, as long as you make sure not to share out more CPU's to the guests then you have actual cores on the hardware, and makes sure to use a separate NIC for that guest OS. I will summarize the results from my tests on the wiki soon, but I'm away for now, and won't get access to all the information I need until I get back - after the ClueCon event(!). If using any kind of cloud environment (like the VPS the original questions was about) that's not under your own control, I highly recommend to solve it another way. Those solutions are usually to much loaded to get an accurate time. However, I do know that using for instance EC2 (xen) has worked for some, but that would also require that host to be on Linux (the original question was about Windows). /Peter ________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Bryan Smart [bryansmart@bryansmart.com] Skickat: den 28 juli 2011 13:52 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
Peter, thanks for those details.
What sort of trade-offs, if any, are useful to improve audio for conferences and IVRs on VMs? Will increasing the interval setting for a conference help? What about using larger frames for codecs? I know that latency, beyond a point, is a serious quality issue of its own. Still, do you know what compromises provide the best improvement? Bryan
On 7/28/11, Peter Olsson <peter.olsson@visionutveckling.se> wrote:
I feel I need to explain my last statement a little further :)
First of all, you can run FS on virtual machines (even though I never recommend it), but you need to be in control of the virtual environment, or totally trust the ones controlling it.
There are two basic problems when running on virtual hardware - timing and network I/O. For timing to work you need to make sure that the one controlling the virtual environment NEVER give out more CPU cores then there are available on the hardware. If they give out more, and your machine need to share resources with another machine, you will be in trouble. As soon as the other virtual machine starts to load the CPU, your machine will get bad timing. Also - usually a one core VM will get better timing then a two+ core VM.
Also, to be safe on the network side, you need your own NIC, that is not shared with other machines.
If you feel you can control this, you might be successful with the setup, if you can't control this, you're either lucky (and things work anyway), or you'll get bad audio.
Also, just passing RTP for normal call isn't the most critical issue (you don't even have to use a timer there), the most critical timing issues will occur in conferences or when using a local IVR on the machine.
I've done lots of testing for this in my lab, since many of my customers wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual hosts, and I've found the above to be quite critical to succeed with your virtual implementation.
/Peter
____________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Giovanni Maruzzelli [gmaruzz@gmail.com] Skickat: den 28 juli 2011 09:37 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
Hey guys 'n gal,
the original post is asking for Windows!!!
-giovanni
On 7/28/11, Sam <lakersman2006@yahoo.com> wrote:
i forgot to mention that the 50 to 60 calls also are doing majority of g729 transcoding as well which does put th cpu to 85 to 90% full utilization. So don't knock virtual machines.
________________________________ From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, July 27, 2011 3:20 PM Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
I'm running FS under Xen provided by Linode. I only run max 10 concurrent channels in production, but an MOH test sounded good through 200 concurrent channels. Linode says the CPU is the most underused resource on their systems...
-Avi
On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc@freeswitch.org> wrote:
On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote:
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P
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I have configured a new gateway for outbound calls from FS as <list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom_carrier/$1"/> </condition> </extension> </include> Created sip_profiles\external\custom_carrier.xml <include> <gateway name="custom_carrier"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> <param name="register" value="false"/> </gateway> </include> When I make an outbound call from registered extension (1001) xlite I get the following error. (I also tried using SJ phone) 2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel sofia/external/14152484142 entering state [terminated][503] 2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal sofia/external/14152484142 [KILL] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/14152484142 [BREAK] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314 (sofia/external/14152484142) Running State Change CS_HANGUP 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498 (sofia/external/14152484142) State HANGUP 2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435 sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other leg I have attached the complete log too. Iam a new FC users. I have referred to the following thread http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm... and also read the section "Connecting to the world with gateways" in FreeSwitch book several times. Any help would be appreciated. Note: Iam able to make outbond calls from softphone through the trunk without FS. Srini
I think you should try to change "custom_carrier" under dialplan and sip_profile to "xxx.xxx.xxx.xxx": <list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/> </condition> </extension> </include> Created sip_profiles\external\custom_carrier.xml <include> <gateway name="xxx.xxx.xxx.xxx"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> <param name="register" value="false"/> </gateway> </include> Regards, Dmitry. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Srinivasan Krishnamurthy Sent: 28 Jul 2011 20:53 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Trunk without authentication configuration I have configured a new gateway for outbound calls from FS as <list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom_carrier/$1"/> </condition> </extension> </include> Created sip_profiles\external\custom_carrier.xml <include> <gateway name="custom_carrier"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> <param name="register" value="false"/> </gateway> </include> When I make an outbound call from registered extension (1001) xlite I get the following error. (I also tried using SJ phone) 2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel sofia/external/14152484142 entering state [terminated][503] 2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal sofia/external/14152484142 [KILL] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/14152484142 [BREAK] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314 (sofia/external/14152484142) Running State Change CS_HANGUP 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498 (sofia/external/14152484142) State HANGUP 2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435 sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other leg I have attached the complete log too. Iam a new FC users. I have referred to the following thread http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm l and also read the section "Connecting to the world with gateways" in FreeSwitch book several times. Any help would be appreciated. Note: Iam able to make outbond calls from softphone through the trunk without FS. Srini
Thanks for your suggestion. I tried but got Invalid gateway err!!! ----------------------------------------------------------------------------- EXECUTE sofia/internal/1001@10.10.2.3:5070 bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142) 2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway 2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW] 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY 2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep 2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed. Cause: INVALID_NUMBER_FORMAT ------------------------------------------------------------------------------- Regards Srini -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Dmitry Saratsky Sent: Thursday, July 28, 2011 2:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration I think you should try to change "custom_carrier" under dialplan and sip_profile to "xxx.xxx.xxx.xxx": <list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/> </condition> </extension> </include> Created sip_profiles\external\custom_carrier.xml <include> <gateway name="xxx.xxx.xxx.xxx"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> <param name="register" value="false"/> </gateway> </include> Regards, Dmitry. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Srinivasan Krishnamurthy Sent: 28 Jul 2011 20:53 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Trunk without authentication configuration I have configured a new gateway for outbound calls from FS as <list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom_carrier/$1"/> </condition> </extension> </include> Created sip_profiles\external\custom_carrier.xml <include> <gateway name="custom_carrier"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> <param name="register" value="false"/> </gateway> </include> When I make an outbound call from registered extension (1001) xlite I get the following error. (I also tried using SJ phone) 2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel sofia/external/14152484142 entering state [terminated][503] 2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal sofia/external/14152484142 [KILL] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/14152484142 [BREAK] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314 (sofia/external/14152484142) Running State Change CS_HANGUP 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498 (sofia/external/14152484142) State HANGUP 2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435 sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other leg I have attached the complete log too. Iam a new FC users. I have referred to the following thread http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm l and also read the section "Connecting to the world with gateways" in FreeSwitch book several times. Any help would be appreciated. Note: Iam able to make outbond calls from softphone through the trunk without FS. Srini _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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 restarted FS after that change? Or you only have reloaded xmls??? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Srinivasan Krishnamurthy Sent: 29 Jul 2011 00:55 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration Thanks for your suggestion. I tried but got Invalid gateway err!!! ---------------------------------------------------------------------------- - EXECUTE sofia/internal/1001@10.10.2.3:5070 bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142) 2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway 2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW] 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY 2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep 2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed. Cause: INVALID_NUMBER_FORMAT ---------------------------------------------------------------------------- --- Regards Srini -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Dmitry Saratsky Sent: Thursday, July 28, 2011 2:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration I think you should try to change "custom_carrier" under dialplan and sip_profile to "xxx.xxx.xxx.xxx": <list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/> </condition> </extension> </include> Created sip_profiles\external\custom_carrier.xml <include> <gateway name="xxx.xxx.xxx.xxx"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> <param name="register" value="false"/> </gateway> </include> Regards, Dmitry. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Srinivasan Krishnamurthy Sent: 28 Jul 2011 20:53 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Trunk without authentication configuration I have configured a new gateway for outbound calls from FS as <list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom_carrier/$1"/> </condition> </extension> </include> Created sip_profiles\external\custom_carrier.xml <include> <gateway name="custom_carrier"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> <param name="register" value="false"/> </gateway> </include> When I make an outbound call from registered extension (1001) xlite I get the following error. (I also tried using SJ phone) 2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel sofia/external/14152484142 entering state [terminated][503] 2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal sofia/external/14152484142 [KILL] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/14152484142 [BREAK] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314 (sofia/external/14152484142) Running State Change CS_HANGUP 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498 (sofia/external/14152484142) State HANGUP 2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435 sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other leg I have attached the complete log too. Iam a new FC users. I have referred to the following thread http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm l and also read the section "Connecting to the world with gateways" in FreeSwitch book several times. Any help would be appreciated. Note: Iam able to make outbond calls from softphone through the trunk without FS. Srini _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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
Might want to doublecheck the format your carrier accepts for outgoing calls. Also, attatch a full siptrace with your debug log, that helps a lot. freeswitch> sofia profile external siptrace on On Thu, Jul 28, 2011 at 5:58 PM, Dmitry Saratsky <freeswitch@simpot.com>wrote:
Did you restarted FS after that change? Or you only have reloaded xmls???
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Srinivasan Krishnamurthy Sent: 29 Jul 2011 00:55 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration
Thanks for your suggestion. I tried but got Invalid gateway err!!!
---------------------------------------------------------------------------- - EXECUTE sofia/internal/1001@10.10.2.3:5070 bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142) 2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway 2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW] 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY 2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep 2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed. Cause: INVALID_NUMBER_FORMAT
---------------------------------------------------------------------------- --- Regards Srini
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Dmitry Saratsky Sent: Thursday, July 28, 2011 2:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration
I think you should try to change "custom_carrier" under dialplan and sip_profile to "xxx.xxx.xxx.xxx":
<list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml
Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/> </condition> </extension> </include>
Created sip_profiles\external\custom_carrier.xml <include> <gateway name="xxx.xxx.xxx.xxx"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/>
<param name="register" value="false"/> </gateway> </include>
Regards, Dmitry.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Srinivasan Krishnamurthy Sent: 28 Jul 2011 20:53 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Trunk without authentication configuration
I have configured a new gateway for outbound calls from FS as
<list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml
Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom_carrier/$1"/> </condition> </extension> </include>
Created sip_profiles\external\custom_carrier.xml <include> <gateway name="custom_carrier"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/>
<param name="register" value="false"/> </gateway> </include>
When I make an outbound call from registered extension (1001) xlite I get the following error. (I also tried using SJ phone)
2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel sofia/external/14152484142 entering state [terminated][503] 2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal sofia/external/14152484142 [KILL] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/14152484142 [BREAK] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314 (sofia/external/14152484142) Running State Change CS_HANGUP 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498 (sofia/external/14152484142) State HANGUP 2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435 sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other leg
I have attached the complete log too.
Iam a new FC users. I have referred to the following thread
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm l
and also read the section "Connecting to the world with gateways" in FreeSwitch book several times.
Any help would be appreciated.
Note: Iam able to make outbond calls from softphone through the trunk without FS.
Srini
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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Now my sip_profiles\external\custom_carrier.xml looks like <include> <gateway name="custom_carrier"> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> </gateway> </include> I have restarted FS. Enabled sofia log in debug “sofia tracelevel 6’ and sip trace is on too. Refer to complete log @ http://pastebin.freeswitch.org/16918 2011-07-29 10:00:31.372738 [ERR] mod_sofia.c:3492 Invalid Gateway 2011-07-29 10:00:31.372738 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW] 2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY 2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY 2011-07-29 10:00:31.372738 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY 2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep 2011-07-29 10:00:31.372738 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] Regards Srini From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brad Mina Sent: Thursday, July 28, 2011 7:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration Might want to doublecheck the format your carrier accepts for outgoing calls. Also, attatch a full siptrace with your debug log, that helps a lot. freeswitch> sofia profile external siptrace on On Thu, Jul 28, 2011 at 5:58 PM, Dmitry Saratsky <freeswitch@simpot.com<mailto:freeswitch@simpot.com>> wrote: Did you restarted FS after that change? Or you only have reloaded xmls??? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Srinivasan Krishnamurthy Sent: 29 Jul 2011 00:55 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration Thanks for your suggestion. I tried but got Invalid gateway err!!! ---------------------------------------------------------------------------- - EXECUTE sofia/internal/1001@10.10.2.3:5070<http://1001@10.10.2.3:5070> bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142<tel:14152484142>) 2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway 2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW] 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY 2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep 2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed. Cause: INVALID_NUMBER_FORMAT ---------------------------------------------------------------------------- --- Regards Srini -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Dmitry Saratsky Sent: Thursday, July 28, 2011 2:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration I think you should try to change "custom_carrier" under dialplan and sip_profile to "xxx.xxx.xxx.xxx": <list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/> </condition> </extension> </include> Created sip_profiles\external\custom_carrier.xml <include> <gateway name="xxx.xxx.xxx.xxx"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> <param name="register" value="false"/> </gateway> </include> Regards, Dmitry. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Srinivasan Krishnamurthy Sent: 28 Jul 2011 20:53 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Trunk without authentication configuration I have configured a new gateway for outbound calls from FS as <list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom_carrier/$1"/> </condition> </extension> </include> Created sip_profiles\external\custom_carrier.xml <include> <gateway name="custom_carrier"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> <param name="register" value="false"/> </gateway> </include> When I make an outbound call from registered extension (1001) xlite I get the following error. (I also tried using SJ phone) 2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel sofia/external/14152484142<tel:14152484142> entering state [terminated][503] 2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup sofia/external/14152484142<tel:14152484142> [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal sofia/external/14152484142<tel:14152484142> [KILL] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/14152484142<tel:14152484142> [BREAK] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314 (sofia/external/14152484142<tel:14152484142>) Running State Change CS_HANGUP 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498 (sofia/external/14152484142<tel:14152484142>) State HANGUP 2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435 sofia/external/14152484142<tel:14152484142> Overriding SIP cause 503 with 503 from the other leg I have attached the complete log too. Iam a new FC users. I have referred to the following thread http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm l and also read the section "Connecting to the world with gateways" in FreeSwitch book several times. Any help would be appreciated. Note: Iam able to make outbond calls from softphone through the trunk without FS. Srini _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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
Srini, I think you misunderstood what my suggestion. What you've got in your gateway definition are parameters that belong in the SIP profile. Again, look in conf/sip_profiles/external/example.xml. Don't add any parameters that are not already in that file. A gateway definition should be no more complicated than this: <include> <gateway name="custom_carrier"> <param name="username" value="my_username"/> <param name="realm" value="target_realm"/> <param name="password" value="my_password"/> <param name="register" value="false"/> </gateway> </include> Also, when doing a sip trace, simply do this: sofia global siptrace on If you do that sofialoglevel stuff you'll end up with way more information than you need. -MC On Fri, Jul 29, 2011 at 10:19 AM, Srinivasan Krishnamurthy < skrishnamurthy@attinteractive.com> wrote:
Now my sip_profiles\external\custom_carrier.xml looks like****
** **
<include>****
<gateway name="custom_carrier">****
<param name="auth-calls" value="false"/>** **
<param name="apply-outbound-acl" value="custom_carrier"/>****
<param name="apply-inbound-acl" value="custom_carrier"/> ****
</gateway>****
</include>****
** **
I have restarted FS. Enabled sofia log in debug “sofia tracelevel 6’ and sip trace is on too.****
Refer to complete log @ http://pastebin.freeswitch.org/16918****
** **
2011-07-29 10:00:31.372738 [ERR] mod_sofia.c:3492 Invalid Gateway****
2011-07-29 10:00:31.372738 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW]****
2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY****
2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY****
2011-07-29 10:00:31.372738 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY****
2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep****
2011-07-29 10:00:31.372738 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]****
** **
Regards****
Srini****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brad Mina *Sent:* Thursday, July 28, 2011 7:33 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Trunk without authentication configuration****
** **
Might want to doublecheck the format your carrier accepts for outgoing calls.****
** **
Also, attatch a full siptrace with your debug log, that helps a lot.
freeswitch> sofia profile external siptrace on****
On Thu, Jul 28, 2011 at 5:58 PM, Dmitry Saratsky <freeswitch@simpot.com> wrote:****
Did you restarted FS after that change? Or you only have reloaded xmls???* ***
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Srinivasan Krishnamurthy****
Sent: 29 Jul 2011 00:55****
To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration
Thanks for your suggestion. I tried but got Invalid gateway err!!!
---------------------------------------------------------------------------- - EXECUTE sofia/internal/1001@10.10.2.3:5070 bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142) 2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway 2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW] 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY 2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep 2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed. Cause: INVALID_NUMBER_FORMAT
---------------------------------------------------------------------------- --- Regards Srini
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Dmitry Saratsky Sent: Thursday, July 28, 2011 2:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration
I think you should try to change "custom_carrier" under dialplan and sip_profile to "xxx.xxx.xxx.xxx":
<list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml
Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/> </condition> </extension> </include>
Created sip_profiles\external\custom_carrier.xml <include> <gateway name="xxx.xxx.xxx.xxx"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/>
<param name="register" value="false"/> </gateway> </include>
Regards, Dmitry.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Srinivasan Krishnamurthy Sent: 28 Jul 2011 20:53 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Trunk without authentication configuration
I have configured a new gateway for outbound calls from FS as
<list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml
Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom_carrier/$1"/> </condition> </extension> </include>
Created sip_profiles\external\custom_carrier.xml <include> <gateway name="custom_carrier"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/>
<param name="register" value="false"/> </gateway> </include>
When I make an outbound call from registered extension (1001) xlite I get the following error. (I also tried using SJ phone)
2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel sofia/external/14152484142 entering state [terminated][503] 2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal sofia/external/14152484142 [KILL] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/14152484142 [BREAK] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314 (sofia/external/14152484142) Running State Change CS_HANGUP 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498 (sofia/external/14152484142) State HANGUP 2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435 sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other leg
I have attached the complete log too.
Iam a new FC users. I have referred to the following thread
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm l
and also read the section "Connecting to the world with gateways" in FreeSwitch book several times.
Any help would be appreciated.
Note: Iam able to make outbond calls from softphone through the trunk without FS.
Srini
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Thanks Michael now it works. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, July 29, 2011 10:34 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration Srini, I think you misunderstood what my suggestion. What you've got in your gateway definition are parameters that belong in the SIP profile. Again, look in conf/sip_profiles/external/example.xml. Don't add any parameters that are not already in that file. A gateway definition should be no more complicated than this: <include> <gateway name="custom_carrier"> <param name="username" value="my_username"/> <param name="realm" value="target_realm"/> <param name="password" value="my_password"/> <param name="register" value="false"/> </gateway> </include> Also, when doing a sip trace, simply do this: sofia global siptrace on If you do that sofialoglevel stuff you'll end up with way more information than you need. -MC On Fri, Jul 29, 2011 at 10:19 AM, Srinivasan Krishnamurthy <skrishnamurthy@attinteractive.com<mailto:skrishnamurthy@attinteractive.com>> wrote: Now my sip_profiles\external\custom_carrier.xml looks like <include> <gateway name="custom_carrier"> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> </gateway> </include> I have restarted FS. Enabled sofia log in debug "sofia tracelevel 6' and sip trace is on too. Refer to complete log @ http://pastebin.freeswitch.org/16918 2011-07-29 10:00:31.372738 [ERR] mod_sofia.c:3492 Invalid Gateway 2011-07-29 10:00:31.372738 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW] 2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY 2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY 2011-07-29 10:00:31.372738 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY 2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep 2011-07-29 10:00:31.372738 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] Regards Srini From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Brad Mina Sent: Thursday, July 28, 2011 7:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration Might want to doublecheck the format your carrier accepts for outgoing calls. Also, attatch a full siptrace with your debug log, that helps a lot. freeswitch> sofia profile external siptrace on On Thu, Jul 28, 2011 at 5:58 PM, Dmitry Saratsky <freeswitch@simpot.com<mailto:freeswitch@simpot.com>> wrote: Did you restarted FS after that change? Or you only have reloaded xmls??? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Srinivasan Krishnamurthy Sent: 29 Jul 2011 00:55 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration Thanks for your suggestion. I tried but got Invalid gateway err!!! ---------------------------------------------------------------------------- - EXECUTE sofia/internal/1001@10.10.2.3:5070<http://1001@10.10.2.3:5070> bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142<tel:14152484142>) 2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway 2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW] 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY 2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep 2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed. Cause: INVALID_NUMBER_FORMAT ---------------------------------------------------------------------------- --- Regards Srini -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Dmitry Saratsky Sent: Thursday, July 28, 2011 2:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trunk without authentication configuration I think you should try to change "custom_carrier" under dialplan and sip_profile to "xxx.xxx.xxx.xxx": <list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/> </condition> </extension> </include> Created sip_profiles\external\custom_carrier.xml <include> <gateway name="xxx.xxx.xxx.xxx"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> <param name="register" value="false"/> </gateway> </include> Regards, Dmitry. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Srinivasan Krishnamurthy Sent: 28 Jul 2011 20:53 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Trunk without authentication configuration I have configured a new gateway for outbound calls from FS as <list name="custom_carrier" default="deny"> <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> </list> In acl.conf.xml Created dialplan\default\01_custom_carrier.xml file as <include> <!-- Dial any 10 digit number 1(2223334444) here --> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom_carrier/$1"/> </condition> </extension> </include> Created sip_profiles\external\custom_carrier.xml <include> <gateway name="custom_carrier"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="auth-calls" value="false"/> <param name="apply-outbound-acl" value="custom_carrier"/> <param name="apply-inbound-acl" value="custom_carrier"/> <param name="register" value="false"/> </gateway> </include> When I make an outbound call from registered extension (1001) xlite I get the following error. (I also tried using SJ phone) 2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel sofia/external/14152484142<tel:14152484142> entering state [terminated][503] 2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup sofia/external/14152484142<tel:14152484142> [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal sofia/external/14152484142<tel:14152484142> [KILL] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/14152484142<tel:14152484142> [BREAK] 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314 (sofia/external/14152484142<tel:14152484142>) Running State Change CS_HANGUP 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498 (sofia/external/14152484142<tel:14152484142>) State HANGUP 2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435 sofia/external/14152484142<tel:14152484142> Overriding SIP cause 503 with 503 from the other leg I have attached the complete log too. Iam a new FC users. I have referred to the following thread http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm l and also read the section "Connecting to the world with gateways" in FreeSwitch book several times. Any help would be appreciated. Note: Iam able to make outbond calls from softphone through the trunk without FS. Srini _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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
Thanks you all! I decided to keep hosting locally instead of upgrading from VPS to a Dedicated Server and got DSL service with static ip to be able to log in remotely. Thanks again, John On Thu, Jul 28, 2011 at 8:16 AM, Peter Olsson < peter.olsson@visionutveckling.se> wrote:
It all depends really. On some solutions (depending on hypervisor, hardware and guest OS) the main problem can be that 20ms is a too short period to sleep, in that case a larger frame (ie 40ms) might help out. But mostly the problem is that the clock is not accurate enough, and fluxtuates from time to time (sometimes 20ms is 10, next round 30 etc).
After my latest testing I've found out that latest ESXi (4.1) works quite good, as long as you make sure not to share out more CPU's to the guests then you have actual cores on the hardware, and makes sure to use a separate NIC for that guest OS. I will summarize the results from my tests on the wiki soon, but I'm away for now, and won't get access to all the information I need until I get back - after the ClueCon event(!).
If using any kind of cloud environment (like the VPS the original questions was about) that's not under your own control, I highly recommend to solve it another way. Those solutions are usually to much loaded to get an accurate time. However, I do know that using for instance EC2 (xen) has worked for some, but that would also require that host to be on Linux (the original question was about Windows).
/Peter ________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [ freeswitch-users-bounces@lists.freeswitch.org] för Bryan Smart [ bryansmart@bryansmart.com] Skickat: den 28 juli 2011 13:52 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
Peter, thanks for those details.
What sort of trade-offs, if any, are useful to improve audio for conferences and IVRs on VMs? Will increasing the interval setting for a conference help? What about using larger frames for codecs? I know that latency, beyond a point, is a serious quality issue of its own. Still, do you know what compromises provide the best improvement?
Bryan
On 7/28/11, Peter Olsson <peter.olsson@visionutveckling.se> wrote:
I feel I need to explain my last statement a little further :)
First of all, you can run FS on virtual machines (even though I never recommend it), but you need to be in control of the virtual environment, or totally trust the ones controlling it.
There are two basic problems when running on virtual hardware - timing and network I/O. For timing to work you need to make sure that the one controlling the virtual environment NEVER give out more CPU cores then there are available on the hardware. If they give out more, and your machine need to share resources with another machine, you will be in trouble. As soon as the other virtual machine starts to load the CPU, your machine will get bad timing. Also - usually a one core VM will get better timing then a two+ core VM.
Also, to be safe on the network side, you need your own NIC, that is not shared with other machines.
If you feel you can control this, you might be successful with the setup, if you can't control this, you're either lucky (and things work anyway), or you'll get bad audio.
Also, just passing RTP for normal call isn't the most critical issue (you don't even have to use a timer there), the most critical timing issues will occur in conferences or when using a local IVR on the machine.
I've done lots of testing for this in my lab, since many of my customers wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual hosts, and I've found the above to be quite critical to succeed with your virtual implementation.
/Peter
____________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Giovanni Maruzzelli [gmaruzz@gmail.com] Skickat: den 28 juli 2011 09:37 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
Hey guys 'n gal,
the original post is asking for Windows!!!
-giovanni
On 7/28/11, Sam <lakersman2006@yahoo.com> wrote:
i forgot to mention that the 50 to 60 calls also are doing majority of g729 transcoding as well which does put th cpu to 85 to 90% full utilization. So don't knock virtual machines.
________________________________ From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, July 27, 2011 3:20 PM Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
I'm running FS under Xen provided by Linode. I only run max 10 concurrent channels in production, but an MOH test sounded good through 200 concurrent channels. Linode says the CPU is the most underused resource on their systems...
-Avi
On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc@freeswitch.org> wrote:
On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote:
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you
use?
And do you have to continually make sacrifices to the digital gods to keep them appeased? :P
-MC _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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!DSPAM:4e314e1632764912765535!
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Thanks, I will make sure to add this to the wiki, I also have some more detailed information after all my testing, so I will make sure to get it all upp on the wiki. /Peter ________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Giovanni Maruzzelli [gmaruzz@celliax.org] Skickat: den 28 juli 2011 11:33 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected! Peter, that's a pretty clear explanation, and answers many frequently asked questions. Can you please add it to the wiki (eg: "FS on virtual machines", and add links to it from "amazon" and "aws" related pages) ? I'm sure lot of people will find it interesting, and hopefully some other people (and/or you) will add more info and experiences to it. -giovanni On 7/28/11, Peter Olsson <peter.olsson@visionutveckling.se> wrote:
I feel I need to explain my last statement a little further :)
First of all, you can run FS on virtual machines (even though I never recommend it), but you need to be in control of the virtual environment, or totally trust the ones controlling it.
There are two basic problems when running on virtual hardware - timing and network I/O. For timing to work you need to make sure that the one controlling the virtual environment NEVER give out more CPU cores then there are available on the hardware. If they give out more, and your machine need to share resources with another machine, you will be in trouble. As soon as the other virtual machine starts to load the CPU, your machine will get bad timing. Also - usually a one core VM will get better timing then a two+ core VM.
Also, to be safe on the network side, you need your own NIC, that is not shared with other machines.
If you feel you can control this, you might be successful with the setup, if you can't control this, you're either lucky (and things work anyway), or you'll get bad audio.
Also, just passing RTP for normal call isn't the most critical issue (you don't even have to use a timer there), the most critical timing issues will occur in conferences or when using a local IVR on the machine.
I've done lots of testing for this in my lab, since many of my customers wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual hosts, and I've found the above to be quite critical to succeed with your virtual implementation.
/Peter
____________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Giovanni Maruzzelli [gmaruzz@gmail.com] Skickat: den 28 juli 2011 09:37 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
Hey guys 'n gal,
the original post is asking for Windows!!!
-giovanni
On 7/28/11, Sam <lakersman2006@yahoo.com> wrote:
i forgot to mention that the 50 to 60 calls also are doing majority of g729 transcoding as well which does put th cpu to 85 to 90% full utilization. So don't knock virtual machines.
________________________________ From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, July 27, 2011 3:20 PM Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
I'm running FS under Xen provided by Linode. I only run max 10 concurrent channels in production, but an MOH test sounded good through 200 concurrent channels. Linode says the CPU is the most underused resource on their systems...
-Avi
On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc@freeswitch.org> wrote:
On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote:
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P
-MC _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
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-- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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 !DSPAM:4e312d4a32762001915859!
I am running asterisk1.6.2.15 and the only real issues we encounter are occasional inband DTMF problems, but that could be related to Sonus issues. ________________________________ From: Michael Collins <msc@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, July 27, 2011 2:57 PM Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected! On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote: I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P -MC _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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 (11)
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Avi Marcus
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Brad Mina
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Bryan Smart
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Dmitry Saratsky
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Giovanni Maruzzelli
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Giovanni Maruzzelli
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Joao Leme
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Michael Collins
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Peter Olsson
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Sam
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Srinivasan Krishnamurthy