Re: [asterisk-users] TDM2400 card FXS problems
I don't know what it showed at that time, I didn't know to look for that. I'll try to get the customer's permission to re-create the symptoms this weekend and post back with the lsdahdi output. Thank you, Noah Engelberth Direct Link Computer Systems - Message: 12 Date: Thu, 28 Jan 2010 23:13:56 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] TDM2400 card FXS problems To: asterisk-users@lists.digium.com Message-ID: 20100128211356.gx3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Thu, Jan 28, 2010 at 07:25:18PM +, Noah I. Engelberth wrote: We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts What's the output of lsdahdi in that case? (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 card FXS problems
Hi Noah, IMHO replace the FXS module with new one or working module. That could be proved where is the root cause. Mostly, the trouble issue is caused by FXS modules in my experience. Good Luck, Johnson On Fri, Jan 29, 2010 at 10:10 PM, Noah I. Engelberth n...@directlinkcomputers.com wrote: I don't know what it showed at that time, I didn't know to look for that. I'll try to get the customer's permission to re-create the symptoms this weekend and post back with the lsdahdi output. Thank you, Noah Engelberth Direct Link Computer Systems - Message: 12 Date: Thu, 28 Jan 2010 23:13:56 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] TDM2400 card FXS problems To: asterisk-users@lists.digium.com Message-ID: 20100128211356.gx3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Thu, Jan 28, 2010 at 07:25:18PM +, Noah I. Engelberth wrote: We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts What's the output of lsdahdi in that case? (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 card FXS problems
Hello Noah, Just shifting your TDM2400 and system to other computer, it might be figured out your issue. As my opinion, the issue is involved with computer reset and TDM2400 epld programming. Would figuring out the issue completely, it is better to replace the TDM2400 card with Digium's. Garry On Fri, Jan 29, 2010 at 10:10 PM, Noah I. Engelberth n...@directlinkcomputers.com wrote: I don't know what it showed at that time, I didn't know to look for that. I'll try to get the customer's permission to re-create the symptoms this weekend and post back with the lsdahdi output. Thank you, Noah Engelberth Direct Link Computer Systems - Message: 12 Date: Thu, 28 Jan 2010 23:13:56 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] TDM2400 card FXS problems To: asterisk-users@lists.digium.com Message-ID: 20100128211356.gx3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Thu, Jan 28, 2010 at 07:25:18PM +, Noah I. Engelberth wrote: We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts What's the output of lsdahdi in that case? (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 card FXS problems
HI: I had this problem before with TDM2400P but with fxo modules and VPMADT032 (echo canceller),there was no audio at all.but then i unpulgged the ehco canceller module (VPMADT032) from the TDM2400P board and started the server and then i didnt face this issue any more. In your case first check the output of #dmesg if it shows repeated message of Unable to set SW Companding on channel ,then your problem is with the echocanceler module (VPMADT032) and do same what i did,dont talk to digium cause they themselves dont know about this error and dont have solution for it ,i tried before without any success. best regards; --- On Fri, 1/29/10, Noah I. Engelberth n...@directlinkcomputers.com wrote: From: Noah I. Engelberth n...@directlinkcomputers.com Subject: Re: [asterisk-users] TDM2400 card FXS problems To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Date: Friday, January 29, 2010, 2:10 PM I don't know what it showed at that time, I didn't know to look for that. I'll try to get the customer's permission to re-create the symptoms this weekend and post back with the lsdahdi output. Thank you, Noah Engelberth Direct Link Computer Systems - Message: 12 Date: Thu, 28 Jan 2010 23:13:56 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] TDM2400 card FXS problems To: asterisk-users@lists.digium.com Message-ID: 20100128211356.gx3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Thu, Jan 28, 2010 at 07:25:18PM +, Noah I. Engelberth wrote: We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts What's the output of lsdahdi in that case? (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 card FXS problems
We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. So far, this has happened on both times the server has been restarted (once planned, once unplanned) since the system was deployed and the phone lines were punched down to the block that is connected to the TDM card. Does anyone have suggestions on where I should start trying to troubleshoot the root cause of the FXS problem? Obviously having to manually restart Asterisk/DAHDI every time the server reboots isn't a practical long term solution. Thank you, Noah Engelberth Direct Link Computer Systems -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 card FXS problems
On Thu, Jan 28, 2010 at 07:25:18PM +, Noah I. Engelberth wrote: We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts What's the output of lsdahdi in that case? (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 answer detection
Hello All I have a TDM2400 card with 4 FXO, and the the following problem: This card answered all the calls, but for the caller, the call is ringing and I don't hear nothing when it has picked up. Here is piece of my log: Thanks for any help. == Starting post polarity CID detection on channel 21 -- Starting simple switch on 'Zap/21-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/21-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/21-1, SIP/ramal01SIP/ramal02SIP/ramal03|30|tT|r) in new stack -- Called ramal01 -- Called ramal02 -- Called ramal03 -- SIP/ramal03-0070e020 is ringing -- SIP/ramal01-006fd4f0 is ringing -- SIP/ramal02-00705d70 is ringing -- SIP/ramal01-006fd4f0 answered Zap/21-1 == Spawn extension (entrada, s, 2) exited non-zero on 'Zap/21-1' -- Hungup 'Zap/21-1' Asterisk 1.4.11 / Zaptel 1.4.5.1 Regards, McCoy Silva ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 and 3.3v pci
On 2/12/07, Paradise Dove [EMAIL PROTECTED] wrote: my card has just fxo modules and is put in a 3.3v slot. when running modprobe wctdm24xxp it waits for ever and dmesg shows Resetting the modules what could be the problem? when i put this card in another system with 5v slot it works fine. I would call Digium's tech support. They open in 20 minutes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 and 3.3v pci
Really? It's 9:23am EST and they aren't open yet. I would call Digium's tech support. They open in 20 minutes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 and 3.3v pci
does TDM2400 work on 3.3v pci slot? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 and 3.3v pci
On 2/11/07, Paradise Dove [EMAIL PROTECTED] wrote: does TDM2400 work on 3.3v pci slot? Yes, all of Digium's analog cards are dual voltage and can work with either 3.3V or 5V slots. You just need to make sure you have an extra molex connector if you're going to be using FXS modules on the card. William ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 and 3.3v pci
my card has just fxo modules and is put in a 3.3v slot. when running modprobe wctdm24xxp it waits for ever and dmesg shows Resetting the modules what could be the problem? when i put this card in another system with 5v slot it works fine. On 2/12/07, William Moore [EMAIL PROTECTED] wrote: On 2/11/07, Paradise Dove [EMAIL PROTECTED] wrote: does TDM2400 work on 3.3v pci slot? Yes, all of Digium's analog cards are dual voltage and can work with either 3.3V or 5V slots. You just need to make sure you have an extra molex connector if you're going to be using FXS modules on the card. William ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400: some FXS module fail
Hello, I've installed two Digium TDM2400 cards on my server. One has 24FXS and the other has 16 FXS and 4 FXO. They are both connected to power. Unfortunately some of the FXS module fail to initialize and I find following messages in the logs (the rest of the FXS modules work well). Could someone give me some advice? !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 Feb 9 18:09:45 [kernel] !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 Feb 9 18:09:45 [kernel] !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 Feb 9 18:09:45 [kernel] !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 Feb 9 18:09:45 [kernel] !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 Feb 9 18:09:45 [kernel] !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 4C00 Feb 9 18:09:45 [kernel] !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 Feb 9 18:09:45 [kernel] !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 Feb 9 18:09:45 [kernel] !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 Feb 9 18:09:45 [kernel] !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C Feb 9 18:09:45 [kernel] !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 Feb 9 18:09:46 [kernel] !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 Feb 9 18:09:46 [kernel] !!! CM_BIAS_RINGING iREG 28 = should be C00 Feb 9 18:09:46 [kernel] !!! DCDC_MIN_V iREG 29 = should be C00 Feb 9 18:09:46 [kernel] !!! DCDC_XTRA iREG 2A = should be 1000 Feb 9 18:09:46 [kernel] !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 Feb 9 18:09:46 [kernel] ! Init Indirect Registers UNSUCCESSFULLY. Feb 9 18:09:46 [kernel] Indirect Registers failed verification. Feb 9 18:09:46 [kernel] Port 5: FAILED FXS (FCC) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM2400 Hardware Echo Cancel[Spam score: 8%][Scanned]
That's interesting, as I've still not managed to completely resolve the problem. I've managed to reduce it by upgrading to Zaptel-1.4 and rerunning fxotune, but there is still a distinct crackle, which is more noticeable on calls to mobiles. I am yet to try removing the hardware echo module, but as you say this is not ideal for a production system. If you don't mind me asking, what alternative hardware are you now using? Is it a similar device that supports up to 24 analogue lines? Adam Sharples -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gagnon Sent: 25 January 2007 07:30 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] TDM2400 Hardware Echo Cancel[Spam score: 8%][Scanned] I had the exact same problem, removing the hardware echo fix the problem but this is not a solution for a production system. I'm now using another brand of hardware. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Webster, Andrew Envoyé : 23 janvier 2007 14:42 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [asterisk-users] TDM2400 Hardware Echo Cancel I have been having the same problems since installing a TDM2400 with hardware echo canceller. The best way to describe the sound is a background crackle or hiss that just can't be filtered out. Increasing the RX gain just makes the problem worse. SIP to SIP calls are flawless. An acquaintance told me the analog line level is too low, but when plugging a regular phone into the line, the signal is plenty loud enough. I am curious if anyone else had similar issues with the TDM2400 card and if they have resolved it. -- Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Sharples Sent: Tuesday, January 16, 2007 09:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM2400 Hardware Echo Cancel Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? Would I be better removing the hardware module and tuning the software echo canceller? The asterisk box is currently running 1.2.13, with zaptel 1.2. Would you advise upgrading to the newer Zaptel drivers? I don't want to upgrade Asterisk itself just yet. Any help or pointers to documentation regarding the hardware echo cancel module would be greatly appreciated, Thanks, Adam Sharples ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM2400 Hardware Echo Cancel
I had the exact same problem, removing the hardware echo fix the problem but this is not a solution for a production system. I'm now using another brand of hardware. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Webster, Andrew Envoyé : 23 janvier 2007 14:42 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [asterisk-users] TDM2400 Hardware Echo Cancel I have been having the same problems since installing a TDM2400 with hardware echo canceller. The best way to describe the sound is a background crackle or hiss that just can't be filtered out. Increasing the RX gain just makes the problem worse. SIP to SIP calls are flawless. An acquaintance told me the analog line level is too low, but when plugging a regular phone into the line, the signal is plenty loud enough. I am curious if anyone else had similar issues with the TDM2400 card and if they have resolved it. -- Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Sharples Sent: Tuesday, January 16, 2007 09:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM2400 Hardware Echo Cancel Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? Would I be better removing the hardware module and tuning the software echo canceller? The asterisk box is currently running 1.2.13, with zaptel 1.2. Would you advise upgrading to the newer Zaptel drivers? I don't want to upgrade Asterisk itself just yet. Any help or pointers to documentation regarding the hardware echo cancel module would be greatly appreciated, Thanks, Adam Sharples ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM2400 Hardware Echo Cancel
I have been having the same problems since installing a TDM2400 with hardware echo canceller. The best way to describe the sound is a background crackle or hiss that just can't be filtered out. Increasing the RX gain just makes the problem worse. SIP to SIP calls are flawless. An acquaintance told me the analog line level is too low, but when plugging a regular phone into the line, the signal is plenty loud enough. I am curious if anyone else had similar issues with the TDM2400 card and if they have resolved it. -- Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Sharples Sent: Tuesday, January 16, 2007 09:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM2400 Hardware Echo Cancel Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? Would I be better removing the hardware module and tuning the software echo canceller? The asterisk box is currently running 1.2.13, with zaptel 1.2. Would you advise upgrading to the newer Zaptel drivers? I don't want to upgrade Asterisk itself just yet. Any help or pointers to documentation regarding the hardware echo cancel module would be greatly appreciated, Thanks, Adam Sharples ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 Hardware Echo Cancel
Had the exact same issue with the hardware canceller. If I set echocancel=no then the problem goes away. Very weird that the static only happens on our side and we only hear it. - Original Message - From: Webster, Andrew [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 23, 2007 2:42 PM Subject: RE: [asterisk-users] TDM2400 Hardware Echo Cancel I have been having the same problems since installing a TDM2400 with hardware echo canceller. The best way to describe the sound is a background crackle or hiss that just can't be filtered out. Increasing the RX gain just makes the problem worse. SIP to SIP calls are flawless. An acquaintance told me the analog line level is too low, but when plugging a regular phone into the line, the signal is plenty loud enough. I am curious if anyone else had similar issues with the TDM2400 card and if they have resolved it. -- Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Sharples Sent: Tuesday, January 16, 2007 09:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM2400 Hardware Echo Cancel Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? Would I be better removing the hardware module and tuning the software echo canceller? The asterisk box is currently running 1.2.13, with zaptel 1.2. Would you advise upgrading to the newer Zaptel drivers? I don't want to upgrade Asterisk itself just yet. Any help or pointers to documentation regarding the hardware echo cancel module would be greatly appreciated, Thanks, Adam Sharples ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 Hardware Echo Cancel
Hi Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. Just a thought, but check that your gain levels are not too high? Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [asterisk-users] TDM2400 Hardware Echo Cancel
Check without the echocan module (remove it) if any 'crackle is listen again. If yes, the echocan is not faulty. If yes, check another echocan module temporary. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ed W Envoyé : jeudi 18 janvier 2007 12:39 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] TDM2400 Hardware Echo Cancel Hi Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. Just a thought, but check that your gain levels are not too high? Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 Hardware Echo Cancel (Adam Sharples)
Hi Adam, I have the same problem. Are you sure is an echo canceller problem? Following advices from this list I discovered that I had an IRQ shared. Untill now I didn't try the new setup but I really hope that this was the problem. If you manage to solve the problem in any way please give me advise. Ciao You can make sure the card is sitting on it's own IRQ - use the command cat /proc/interrupts You can also check that the card isn't losing interrupts by running the zttest proram: /sbin/zttest There's more on interrupts here: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting It's aimes at the TDM400P card, but I'd be surprised if the 2400P is that much different (but someone please correct me if it is!) lspci -vb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 Hardware Echo Cancel
Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? Would I be better removing the hardware module and tuning the software echo canceller? The asterisk box is currently running 1.2.13, with zaptel 1.2. Would you advise upgrading to the newer Zaptel drivers? I don't want to upgrade Asterisk itself just yet. Any help or pointers to documentation regarding the hardware echo cancel module would be greatly appreciated, Thanks, Adam Sharples ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 Hardware Echo Cancel
Adam Sharples wrote: I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? No. Hardware echo cancelers on Digium cards are either 'on' or 'off', there are no tuning parameters. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400
I am having 2 more issues, when starting asterisk I got the below message:
Dec 17 22:27:54 NOTICE[4554]: indications.c:505
ast_unregister_indication_country: Removed default indication country 'us'
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
signalling
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
answeronpolarityswitch
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
hanguponpolarityswitch
my setup is : softphone---softswitch(asterisk)Termination GW(asterisk
with TDM card)
when dialing from my softphone I got :
-- Executing Dial(SIP/10.8.0.6-0947d008, (Zap/g1/6159587)) in new
stack
Dec 17 23:04:16 WARNING[5052]: channel.c:2597 ast_request: No channel type
registered for '(Zap'
Dec 17 23:04:16 NOTICE[5052]: app_dial.c:1056 dial_exec_full: Unable to
create channel of type '(Zap' (cause 66 - Channel not implemented)
my extensions.conf file has:
[globals]
TRUNK=Zap/g1
[topstn]
exten=_2.,1,Dial,(${TRUNK}/${EXTEN:3})
Please help ...
Thanks,
On 12/12/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Dec 11, 2006 at 08:08:18PM -0500, Time Bandit wrote:
[channels]
context=default
signalling=fxs_ls
;channel=1-16
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
;accountcode=lss0101
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
To the best of my knowledge, all the settings you put after defining
the channles (channel= line) are useless. You have to set all the
settings BEFORE you define the channels.
Should be. However in practice after the first reload all of them will
be applied (in this specific case).
/me points again to genzaptelconf that should have made this thread
unnecessary.
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] TDM2400
After restarting the machine I am getting the below messages when dialing:
Dec 18 00:09:35 WARNING[2897]: channel.c:2571 ast_request: No translator
path exists for channel type Zap (native 68) to 256
Dec 18 00:09:35 NOTICE[2897]: app_dial.c:1056 dial_exec_full: Unable to
create channel of type 'Zap' (cause 0 - Unknown)
On 12/17/06, O. Kamal [EMAIL PROTECTED] wrote:
I am having 2 more issues, when starting asterisk I got the below message:
Dec 17 22:27:54 NOTICE[4554]: indications.c:505
ast_unregister_indication_country: Removed default indication country 'us'
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
signalling
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
answeronpolarityswitch
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
hanguponpolarityswitch
my setup is : softphone---softswitch(asterisk)Termination GW(asterisk
with TDM card)
when dialing from my softphone I got :
-- Executing Dial(SIP/10.8.0.6-0947d008, (Zap/g1/6159587)) in new
stack
Dec 17 23:04:16 WARNING[5052]: channel.c:2597 ast_request: No channel type
registered for '(Zap'
Dec 17 23:04:16 NOTICE[5052]: app_dial.c:1056 dial_exec_full: Unable to
create channel of type '(Zap' (cause 66 - Channel not implemented)
my extensions.conf file has:
[globals]
TRUNK=Zap/g1
[topstn]
exten=_2.,1,Dial,(${TRUNK}/${EXTEN:3})
Please help ...
Thanks,
On 12/12/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Dec 11, 2006 at 08:08:18PM -0500, Time Bandit wrote:
[channels]
context=default
signalling=fxs_ls
;channel=1-16
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
;accountcode=lss0101
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
To the best of my knowledge, all the settings you put after defining
the channles (channel= line) are useless. You have to set all the
settings BEFORE you define the channels.
Should be. However in practice after the first reload all of them will
be applied (in this specific case).
/me points again to genzaptelconf that should have made this thread
unnecessary.
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto: [EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] TDM2400
On Mon, Dec 11, 2006 at 08:08:18PM -0500, Time Bandit wrote: [channels] context=default signalling=fxs_ls ;channel=1-16 usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 ;accountcode=lss0101 answeronpolarityswitch=yes hanguponpolarityswitch=yes To the best of my knowledge, all the settings you put after defining the channles (channel= line) are useless. You have to set all the settings BEFORE you define the channels. Should be. However in practice after the first reload all of them will be applied (in this specific case). /me points again to genzaptelconf that should have made this thread unnecessary. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400
here is the latest update: in zaptel.conf i used fxsks=1-4 fxsks=5-8 fxsks=9-12 fxsks=13-16 zttool shows hardware OK ztcfg worked normally in zapata.conf when i define the channels channel=1-16 and restaring asterisk it gives the below errors: Dec 12 00:48:28 WARNING[3141]: chan_zap.c:921 zt_open: Unable to specify channel 1: No such device Dec 12 00:48:28 ERROR[3141]: chan_zap.c:6879 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Dec 12 00:48:28 ERROR[3141]: chan_zap.c:10307 setup_zap: Unable to register channel '1-16' Dec 12 00:48:28 WARNING[3141]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Dec 12 00:48:28 WARNING[3141]: loader.c:554 load_modules: Loading module chan_zap.so failed! When I remove the channel=1-16, it loads normally. zapata.conf is below: [channels] context=default signalling=fxs_ls ;channel=1-16 usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 ;accountcode=lss0101 answeronpolarityswitch=yes hanguponpolarityswitch=yes any clue? On 12/11/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Dec 11, 2006 at 09:40:18AM +1100, Howard Lowndes wrote: O.Kamal wrote: I have 16 channels FXO (4 FXO Modules), I did follow the below link, but maybe I understand it wrong (what is a module and slot?), I need an example. http://kb.digium.com/entry/1/90/ http://kb.digium.com/entry/1/90/ For each FXO module, you should have a coresponding line that reads: fxs followed by the type of signalling (gs, ls, or ks) and the equals sign (=) followed by the position of the module times 4 minus 3 a dash, and then the number of the slot times 4. For example, if you had a FXO module on slot 2 of the board using loopstart signalling, the line would read: fxols=5-8, or if the module was on slot 5, the line would read: fxols=17-20 OK, try either: fxsks=1-16 or: fxsks=1-4 fxsks=5-8 fxsks=9-12 fxsks=13-16 probably the latter will be correct Both are. Alternatively, genzaptelconf (xpp/utils/genzaptelconf) will generate a working (though a bit verbose) configuration. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400
I figured out the problem, it is the location of FXO boards on cards, channels are from 9-24 not 1-16. Thanks all for your help, specially Tzafrir, genzaptelconf shows it clearly. On 12/11/06, O. Kamal [EMAIL PROTECTED] wrote: here is the latest update: in zaptel.conf i used fxsks=1-4 fxsks=5-8 fxsks=9-12 fxsks=13-16 zttool shows hardware OK ztcfg worked normally in zapata.conf when i define the channels channel=1-16 and restaring asterisk it gives the below errors: Dec 12 00:48:28 WARNING[3141]: chan_zap.c:921 zt_open: Unable to specify channel 1: No such device Dec 12 00:48:28 ERROR[3141]: chan_zap.c:6879 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Dec 12 00:48:28 ERROR[3141]: chan_zap.c:10307 setup_zap: Unable to register channel '1-16' Dec 12 00:48:28 WARNING[3141]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Dec 12 00:48:28 WARNING[3141]: loader.c:554 load_modules: Loading module chan_zap.so failed! When I remove the channel=1-16, it loads normally. zapata.conf is below: [channels] context=default signalling=fxs_ls ;channel=1-16 usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 ;accountcode=lss0101 answeronpolarityswitch=yes hanguponpolarityswitch=yes any clue? On 12/11/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Dec 11, 2006 at 09:40:18AM +1100, Howard Lowndes wrote: O.Kamal wrote: I have 16 channels FXO (4 FXO Modules), I did follow the below link, but maybe I understand it wrong (what is a module and slot?), I need an example. http://kb.digium.com/entry/1/90/ http://kb.digium.com/entry/1/90/ For each FXO module, you should have a coresponding line that reads: fxs followed by the type of signalling (gs, ls, or ks) and the equals sign (=) followed by the position of the module times 4 minus 3 a dash, and then the number of the slot times 4. For example, if you had a FXO module on slot 2 of the board using loopstart signalling, the line would read: fxols=5-8, or if the module was on slot 5, the line would read: fxols=17-20 OK, try either: fxsks=1-16 or: fxsks=1-4 fxsks=5-8 fxsks=9-12 fxsks=13-16 probably the latter will be correct Both are. Alternatively, genzaptelconf (xpp/utils/genzaptelconf) will generate a working (though a bit verbose) configuration. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400
[channels] context=default signalling=fxs_ls ;channel=1-16 usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 ;accountcode=lss0101 answeronpolarityswitch=yes hanguponpolarityswitch=yes To the best of my knowledge, all the settings you put after defining the channles (channel= line) are useless. You have to set all the settings BEFORE you define the channels. hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400
Yes, signalling should be fks_ks, channel line must be the last one, I will start live testing next week On 12/12/06, Time Bandit [EMAIL PROTECTED] wrote: [channels] context=default signalling=fxs_ls ;channel=1-16 usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 ;accountcode=lss0101 answeronpolarityswitch=yes hanguponpolarityswitch=yes To the best of my knowledge, all the settings you put after defining the channles (channel= line) are useless. You have to set all the settings BEFORE you define the channels. hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 wired description and skiping frames
Erik, the TDM2400 is not in conflict with other interrupts BUT, i saw something very weird: Lspci:04:09.0 Ethernet controller: Digium, Inc.: Unknown device 2400 (rev 11)It is seen by Ubuntu as an Ethernet board but when I cat /proc/interrupts look: CPU0 0: 949222 IO-APIC-edge timer 1: 8913 IO-APIC-edge i8042 7: 1 IO-APIC-edge parport0 8: 3 IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 410852 IO-APIC-edge i804214: 167803 IO-APIC-edge ide050: 111397 IO-APIC-level eth158: 9467531 IO-APIC-level wctdm24xxp209: 29589 IO-APIC-level libata217: 0 IO-APIC-level ehci_hcd:usb1 225: 0 IO-APIC-level ohci_hcd:usb2233: 2862 IO-APIC-level HDA IntelNMI: 0LOC: 949127ERR: 0MIS: 0The strange thing is that everything seems to be running fast like musiconhold and IVRlike doing fowarding on a tape player... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 wired description and skiping frames
On Thu, Sep 21, 2006 at 02:52:03PM -0300, Robson Ribeiro wrote: Erik, the TDM2400 is not in conflict with other interrupts BUT, i saw something very weird: Lspci :04:09.0 Ethernet controller: Digium, Inc.: Unknown device 2400 (rev 11) To get the latest list of PCI IDs descriptions, try: update-pciids lspci -n should tell you the actual numeric IDs. It is seen by Ubuntu as an Ethernet board but when I cat /proc/interrupts look: CPU0 0: 949222IO-APIC-edge timer 1: 8913IO-APIC-edge i8042 7: 1IO-APIC-edge parport0 8: 3IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 410852IO-APIC-edge i8042 14: 167803IO-APIC-edge ide0 50: 111397 IO-APIC-level eth1 58:9467531 IO-APIC-level wctdm24xxp 209: 29589 IO-APIC-level libata 217: 0 IO-APIC-level ehci_hcd:usb1 225: 0 IO-APIC-level ohci_hcd:usb2 233: 2862 IO-APIC-level HDA Intel NMI: 0 LOC: 949127 ERR: 0 MIS: 0 The strange thing is that everything seems to be running fast like musiconhold and IVRlike doing fowarding on a tape player... What do you get on zttest -v ? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 wired description and skiping frames
What is reported depends totally on the version of lspci and it's libraries. It's cosmetic. Robson Ribeiro wrote: Erik, the TDM2400 is not in conflict with other interrupts BUT, i saw something very weird: Lspci :04:09.0 Ethernet controller: Digium, Inc.: Unknown device 2400 (rev 11) It is seen by Ubuntu as an Ethernet board but when I cat /proc/interrupts look: CPU0 0: 949222IO-APIC-edge timer 1: 8913IO-APIC-edge i8042 7: 1IO-APIC-edge parport0 8: 3IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 410852IO-APIC-edge i8042 14: 167803IO-APIC-edge ide0 50: 111397 IO-APIC-level eth1 58:9467531 IO-APIC-level wctdm24xxp 209: 29589 IO-APIC-level libata 217: 0 IO-APIC-level ehci_hcd:usb1 225: 0 IO-APIC-level ohci_hcd:usb2 233: 2862 IO-APIC-level HDA Intel NMI: 0 LOC: 949127 ERR: 0 MIS: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 wired description and skiping frames
Hi Tzafrir, i did update the PCI ID and now it shows::04:09.0 Ethernet controller: Digium, Inc. Wildcard TDM2400P (rev 11) What do you get on zttest -v ?here is what i get on zttest -v:Opened pseudo zap interface, measuring accuracy...8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%[2]+ Stopped ./zttest -v Hope it helps me solve the problem i am runnign out of time and ideas.thanks,Robson Ribeiro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 problem isolated with POLYCOM IP301 phones!!!
I managed to isolate the problem. I installed a Sipura ATA 1001 and it does not generates the problem so I figure the issue is with the Polycom phones. Now the issue is to find out whats wrong with the Polycoms. On 9/21/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Sep 21, 2006 at 02:52:03PM -0300, Robson Ribeiro wrote: Erik, the TDM2400 is not in conflict with other interrupts BUT, i saw something very weird: Lspci :04:09.0 Ethernet controller: Digium, Inc.: Unknown device 2400 (rev 11) To get the latest list of PCI IDs descriptions, try:update-pciidslspci -n should tell you the actual numeric IDs. It is seen by Ubuntu as an Ethernet board but when I cat /proc/interrupts look: CPU00: 949222IO-APIC-edgetimer1: 8913IO-APIC-edgei80427:1IO-APIC-edgeparport08:3IO-APIC-edgertc 9:0 IO-APIC-levelacpi 12: 410852IO-APIC-edgei8042 14: 167803IO-APIC-edgeide0 50: 111397 IO-APIC-leveleth1 58:9467531 IO-APIC-levelwctdm24xxp 209:29589 IO-APIC-levellibata 217:0 IO-APIC-levelehci_hcd:usb1 225:0 IO-APIC-levelohci_hcd:usb2 233: 2862 IO-APIC-levelHDA Intel NMI:0 LOC: 949127 ERR:0 MIS:0 The strange thing is that everything seems to be running fast like musiconhold and IVRlike doing fowarding on a tape player... What do you get on zttest -v ?--Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755 iax:[EMAIL PROTECTED]+972-50-7952406jabber:[EMAIL PROTECTED][EMAIL PROTECTED] http://www.xorcom.com___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM2400 wierdness
Kerry: I hope this helps. I had EXACTLY the same symptom when I was trying to get an X100P clone to work yesterday. Bumping the toneduration parameter in zapata.conf to 200 milliseconds cured the problem. Roger Kerry Garrison wrote: Asterisk 1.2.1 Updated the TDM2400 driver over the weekend Incoming calls seem to work perfectly Outbound calls never connect. If you listen in on the call to a 7 digit local number, you hear the first 6 digits, then a small delay, then the last digit. Then there is a long pause before the line is picked up, then a very long pause before the telco fires back you call could not be completed at this time. Calling using an analog phone on that line works fine. Do I possibly have some DTMF issues or something like that? Any suggestions would be appreciated. This is my only installation with the TDM2400 so I am kind of at a loss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM2400 wierdness
Thanks, I will try that. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Hill Sent: Thursday, December 29, 2005 1:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM2400 wierdness Kerry: I hope this helps. I had EXACTLY the same symptom when I was trying to get an X100P clone to work yesterday. Bumping the toneduration parameter in zapata.conf to 200 milliseconds cured the problem. Roger Kerry Garrison wrote: Asterisk 1.2.1 Updated the TDM2400 driver over the weekend Incoming calls seem to work perfectly Outbound calls never connect. If you listen in on the call to a 7 digit local number, you hear the first 6 digits, then a small delay, then the last digit. Then there is a long pause before the line is picked up, then a very long pause before the telco fires back you call could not be completed at this time. Calling using an analog phone on that line works fine. Do I possibly have some DTMF issues or something like that? Any suggestions would be appreciated. This is my only installation with the TDM2400 so I am kind of at a loss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM2400 wierdness
Try adding a w in your dial statement. Asterisk will dial even if the line is not ready with a dialtone, adding a w will wait a bit and then dial the number. On 12/29/05, Kerry Garrison [EMAIL PROTECTED] wrote: Thanks, I will try that. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Hill Sent: Thursday, December 29, 2005 1:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM2400 wierdness Kerry: I hope this helps. I had EXACTLY the same symptom when I was trying to get an X100P clone to work yesterday. Bumping the toneduration parameter in zapata.conf to 200 milliseconds cured the problem. Roger Kerry Garrison wrote: Asterisk 1.2.1 Updated the TDM2400 driver over the weekend Incoming calls seem to work perfectly Outbound calls never connect. If you listen in on the call to a 7 digit local number, you hear the first 6 digits, then a small delay, then the last digit. Then there is a long pause before the line is picked up, then a very long pause before the telco fires back you call could not be completed at this time. Calling using an analog phone on that line works fine. Do I possibly have some DTMF issues or something like that? Any suggestions would be appreciated. This is my only installation with the TDM2400 so I am kind of at a loss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM2400 wierdness
The toneduration setting seems to have fixed it. Thanks for the tip! -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Thursday, December 29, 2005 7:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM2400 wierdness Try adding a w in your dial statement. Asterisk will dial even if the line is not ready with a dialtone, adding a w will wait a bit and then dial the number. On 12/29/05, Kerry Garrison [EMAIL PROTECTED] wrote: Thanks, I will try that. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Hill Sent: Thursday, December 29, 2005 1:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM2400 wierdness Kerry: I hope this helps. I had EXACTLY the same symptom when I was trying to get an X100P clone to work yesterday. Bumping the toneduration parameter in zapata.conf to 200 milliseconds cured the problem. Roger Kerry Garrison wrote: Asterisk 1.2.1 Updated the TDM2400 driver over the weekend Incoming calls seem to work perfectly Outbound calls never connect. If you listen in on the call to a 7 digit local number, you hear the first 6 digits, then a small delay, then the last digit. Then there is a long pause before the line is picked up, then a very long pause before the telco fires back you call could not be completed at this time. Calling using an analog phone on that line works fine. Do I possibly have some DTMF issues or something like that? Any suggestions would be appreciated. This is my only installation with the TDM2400 so I am kind of at a loss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM2400 wierdness
Asterisk 1.2.1 Updated the TDM2400 driver over the weekend Incoming calls seem to work perfectly Outbound calls never connect. If you listen in on the call to a 7 digit local number, you hear the first 6 digits, then a small delay, then the last digit. Then there is a long pause before the line is picked up, then a very long pause before the telco fires back you call could not be completed at this time. Calling using an analog phone on that line works fine. Do I possibly have some DTMF issues or something like that? Any suggestions would be appreciated. This is my only installation with the TDM2400 so I am kind of at a loss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM2400 wierdness
Asterisk 1.2.1 Updated the TDM2400 driver over the weekend Incoming calls seem to work perfectly Outbound calls never connect. If you listen in on the call to a 7 digit local number, you hear the first 6 digits, then a small delay, then the last digit. Then there is a long pause before the line is picked up, then a very long pause before the telco fires back you call could not be completed at this time. Calling using an analog phone on that line works fine. Do I possibly have some DTMF issues or something like that? Any suggestions would be appreciated. This is my only installation with the TDM2400 so I am kind of at a loss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] TDM2400
Hello folks ! TDM2400 with E for echocan module is ok for me, replacing my old passive cards. No more echo issues now. I had many before to switch to this wonderfull card ! Perfect for my use... Here is an Asterisk SVN-branch-1.2-r7608M, in an old PII-400 MHz Linux version 2.6.12-1-686 (gcc version 4.0.2 20050917 (prerelease) (Debian 4.0.1-8) Etch. My opinion : buy it WITH the echocan option, and don't forget to buy a Centronics 50 pins mâle connector (not provided). Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Guillermo Salas M Envoyé : jeudi 22 décembre 2005 17:01 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] TDM2400 Hi all, I was checking the TDM2400 features and seems to me very interesating. I think is that I need :) I want to know your experience with this card and if you know abouts bugs, configuration and everithing thah I need to know before acquire it :) The http://www.voipsupply.com/product_info.php?products_id=1115 is necesary ? Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM2400
Hi all, I was checking the TDM2400 features and seems to me very interesating. I think is that I need :) I want to know your experience with this card and if you know abouts bugs, configuration and everithing thah I need to know before acquire it :) The http://www.voipsupply.com/product_info.php?products_id=1115 is necesary ? Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM2400
On 12/22/05, Guillermo Salas M [EMAIL PROTECTED] wrote: Hi all, I was checking the TDM2400 features and seems to me very interesating. I think is that I need :) I want to know your experience with this card and if you know abouts bugs, configuration and everithing thah I need to know before acquire it :) The http://www.voipsupply.com/product_info.php?products_id=1115 is necesary ? We have put a number of them into production for our clients already and they are working very well. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM2400
Guillermo Salas M ha scritto: Hi all, I was checking the TDM2400 features and seems to me very interesating. I think is that I need :) I want to know your experience with this card and if you know abouts bugs, configuration and everithing thah I need to know before acquire it :) The http://www.voipsupply.com/product_info.php?products_id=1115 is necesary ? Best regards, Works perfectly out of the box, almost for my customers :-) The only note is to disable echo training. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM2400
Massimo De Nadal wrote: Works perfectly out of the box, almost for my customers :-) The only note is to disable echo training. Could you please elaborate which exact model you're using and what are your opinion about the echo can/training quality? Have you tried spandsp faxing? Thanks in advance, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
