Re: [asterisk-users] asterisk 11.7.0: Delayed audio
I wanted to chime in on this one, I posted a similar problem a while back under the heading Delay before audio starts on 2/26/2013 My solution to fix this problem was to adjust my dialplan by inserting an Answer(); So I don't think it necessarily has something to do with the strictrtp setting. -Gerard On Fri, 2014-01-10 at 15:01 -0600, Matthew Jordan wrote: On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@curtissystemssoftware.com wrote: On connection to an incoming call via PSTN where asterisk [11.7.0] is Dialing an internal extension on answering the call there is about 6-7 seconds before audio is heard on either side. When looking at the CLI traces when I answer the incoming call that asterisk extensions were dialing, I see immediately upon answering 0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:portnumber then not until about 6 seconds later I see this 0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:diffportnumber and immediately hear audio what appears to be an issue is that the RTP link(audio) setup is delayed. Anyone have suggestions on how to fix this issue? If the RTP source address/port is changing, then Asterisk is receiving RTP packets from two different sources and is waiting for one of them to stabilize before it picks the actual source of the media stream. That's by design, as the locking in of the RTP source prevents a media injection attack. You can tweak how Asterisk does this using two settings in rtp.conf: ; Enable strict RTP protection. This will drop RTP packets that ; do not come from the source of the RTP stream. This option is ; enabled by default. ; strictrtp=yes ; Number of packets containing consecutive sequence values needed ; to change the RTP source socket address. This option only comes ; into play while using strictrtp=yes. Consider changing this value ; if rtp packets are dropped from one or both ends after a call is ; connected. This option is set to 4 by default. ; probation=8 Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- Gerard Saraber gsara...@rarcoa.com Rarcoa, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls not going through
I had a similar problem, basically asterisk would be sending dialled
digits before the telco was ready to pick them up, in my case some
numbers would work but most wouldn't and i'd get messages from the telco
like you described, the solution I'm using: put a delay before dialing
the number:
from my extensions.conf:
PAUSE=w
[trunklocal]
exten = _NXX,1,Dial(${TRUNK}/${PAUSE}${EXTEN},${TIMEOUT},W)
exten = _NXX,2,Congestion
similar for the long distance and international calls
basically dial a w before sending the actual number.
HTH
On Thu, 2006-02-23 at 07:33 -0700, [EMAIL PROTECTED]
wrote:
When a call is placed out the Zap interface there is a long pause followed
by an error message from the telco that the call can not be placed as
dialed.
We have a tdm2413e with 11 1FB (POTS) lines. The number being dialed is a
working local number, all dialed numbers get the same error.
What should we be looking for ?
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
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Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems
On Thu, 2006-02-23 at 14:36 -0600, Aaron Daniel wrote: We use mainly 7940's in our environment, I currently have about 60 of them on my system, and the 7.5 firmware really screwed up our phone network. Not in the same way yours has, but there seem to be a lot of glitches, even jumping from 7.4 to 7.5. Have you tried the 7.4 firmware to see if that does you any good? For what its worth, 7.4 seems to work great in my setup, I stayed away from 7.5, luckily I read about the glitching before upgrading. -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] Cisco 79xx firmware
Hi, as far as I know, you can't actually buy the firmware, you have to get the service contract, I talked to the guy at CDW who talked to his Cisco guy, and they told me to buy a $92 service contract. hope that helps.. On Wed, 2006-02-22 at 10:34 +0100, Tomislav Parčina wrote: I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me technical support agreement which cost around 75$ for every phone! -- Tomislav Parcina tparcina#lama.hr -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] Multiple TDM400P's in a single machine
3 TDM cards here, I had artifacts if any of the cards were sharing interrupts, the trick was to add the cards 1 at the time to get them each on their own irq. The system isn't in production yet, so I don't know how well it'll hold up under load, so far so good in testing though. 9xFXO 1xFXS 2xUnused CPU0 0: 592857335IO-APIC-edge timer 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 15559638 IO-APIC-level eth0 185:3677271 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217: 592717484 IO-APIC-level wctdm 225: 592712578 IO-APIC-level wctdm 233: 592725907 IO-APIC-level wctdm NMI: 40812 LOC: 592769967 ERR: 0 MIS: 0 On Tue, 2006-02-21 at 09:23 +, Sean Cook wrote: Same setup with two TDM400 (8FXO) running for over a year. On Tue, 2006-02-21 at 01:37 +0100, Thomas Artner wrote: Am Tuesday 21 February 2006 00:24 schrieb Marc Archer: Hi All, Can someone give me a definite answer as to wether or not you can reliably run multiple TDM400P's in the same machine? I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key system, but I have seen several threads suggesting that this is not a supported configuration i have two tdm400p's (2xFXO, 6xFXS) in one desktop machine used as asterisk server for a small office (so the pc hardware is nothing special). This configuration is running since two weeks without any problems! Thanks, Marc. ___ --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 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] MOH from RCA jack?
My * pc has an integrated soundcard, should be ok for this type of application, I'd get an RCA-to-line jack cable (radioshack should have those ;) I know * can play hold music from a streaming server, and I know some streaming servers can stream from a line in, so the combination of the two should do the trick I think. On Fri, 2006-02-17 at 07:42 -0600, Rich Adamson wrote: Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? ___ --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 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] MOH from RCA jack?
Actually, with my suggestion, you would be using the soundcard with whatever streaming mp3 server you choose to use, some kinda shoutcast server I guess, so there shouldn't be any asterisk-soundcard interaction.. * just takes its moh from the streaming server. On Fri, 2006-02-17 at 09:18 -0600, Rich Adamson wrote: Any idea how difficult it might be to get an integrated sound card to work properly with asterisk? (That seems to be the limiting factor or more time consuming part of doing this. Adapting the cables and audio levels is easy.) My * pc has an integrated soundcard, should be ok for this type of application, I'd get an RCA-to-line jack cable (radioshack should have those ;) I know * can play hold music from a streaming server, and I know some streaming servers can stream from a line in, so the combination of the two should do the trick I think. On Fri, 2006-02-17 at 07:42 -0600, Rich Adamson wrote: Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? ___ --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 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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 ---End of Original Message- ___ --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 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] A unique 'click to call' project - Could use some advice
I'm using the telnet manager interface with the 'originate' command, just a little perl script that connects and has asterisk dial the selected number. It rings the extension first, if they pick up, it'll dial the remote number. It's one of the showcase features of the new phonesystem for us :) and it was surprisingly easy to implement. -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] odd 'digital' sound artifacts
Interesting, I seem to have gotten more 'artifact' complaints after switching to the svn trunk MG2 echo canceler, I just tested with the echo canceler turned off completely, the artifacts are still there. I haven't heard the loud buzzing sound since I stopped using the aggressive canceler. I've also tested with echocancel=256 per your suggestion to see if I can reproduce it, the little blips are still there, just heard one after being on hold for 3 min 30 secs. but no more loud buzzing :) I just heard a different artifact, the blip turned into a buzz and the sound (hold music) coming in repeated itself about 3 times. sorta like this: ...your call will be answered as *digital sounding beep*quickl*quickl*quickly as possible Next up I'm going to try a different mainboard with only one TDM card in it. On Mon, 2006-02-13 at 10:40 +1100, Mike Pollitt wrote: Hi Gerard -- I found that I get the really loud buzzing sound in the handset earpiece when I set echocancel=256 instead of echocancel=yes (the default = 128 taps). It seemed to occur irrespective of the actual echo canceller chosen. Mike. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gerard Saraber Sent: Saturday, 11 February 2006 2:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] odd 'digital' sound artifacts So nobody heard these before? or did I do something stupid that anyone should know and nobody wanted to yell at me for it ;) On Wed, 2006-02-08 at 12:54 -0600, Gerard Saraber wrote: Hi, I've got some weird sound artifacts happening during calls, they're very hard to describe, so I have a 122kb recording: http://openprojects.rarcoa.com/~miztic/artifact.wav normally the artifacts are just short blips, not quite as long as the one above, but they sound the same. When using the aggressive echo suppressor, it seems like those artifacts cause a really loud buzzing sound to come out of the cisco phone, pretty much made using the aggressive canceler impossible to use, it's too bad because it worked the best out of all of them, mark3 works ok but still gives echos on at least 20% of the calls. I thought they might be caused by IRQ sharing, so I pulled one of the TDM400P cards out and made sure the remaining two were on their own IRQ, the artifacts were still there. I've also tried running a kernel with all the low-latency stuff turned on, and the same kernel with it all turned off (2.6.16-rc2) doesn't appear to make any difference either. I'm not sure what else to try, any input would be appreciated. Thanks, Gerard Saraber [EMAIL PROTECTED] hardware: AMD64 1.8Ghz 512M ram MSI nforce3 socket 754 mainboard 3 Digium TDM400P cards, 10 FXO + 2 FXS modules /proc/interrupts CPU0 0:2784232IO-APIC-edge timer 1: 8IO-APIC-edge i8042 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 71552 IO-APIC-level eth0 185: 9412 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217:5577811 IO-APIC-level wctdm, wctdm 225:2769262 IO-APIC-level wctdm lspci (for completeness): 02:07.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at ac00 [size=256] Memory at fdeff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 225 I/O ports at a800 [size=256] Memory at fdefe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at a400 [size=256] Memory at fdefd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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
RE: [Asterisk-Users] odd 'digital' sound artifacts [1 card = no artifacts]
Ok, after 10+ minutes of mindnumbing hold music ;) no artifacts with only one TDM card installed, going to test with two cards on their own IRQ in a minute again.. Feb 14 09:59:31 [kernel] Zaptel Version: SVN-trunk-r941M Echo Canceller: MG2 Feb 14 09:59:31 [kernel] ACPI: PCI Interrupt Link [APC2] enabled at IRQ 17 Feb 14 09:59:31 [kernel] GSI 21 sharing vector 0xD9 and IRQ 21 Feb 14 09:59:31 [kernel] ACPI: PCI Interrupt :02:07.0[A] - Link [APC2] - GSI 17 (level, low) - IRQ 217 CPU0 0:1026246IO-APIC-edge timer 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 50426 IO-APIC-level eth0 185: 6822 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217:1012971 IO-APIC-level wctdm NMI:143 LOC:1026004 ERR: 0 MIS: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gerard Saraber Sent: Saturday, 11 February 2006 2:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] odd 'digital' sound artifacts So nobody heard these before? or did I do something stupid that anyone should know and nobody wanted to yell at me for it ;) On Wed, 2006-02-08 at 12:54 -0600, Gerard Saraber wrote: Hi, I've got some weird sound artifacts happening during calls, they're very hard to describe, so I have a 122kb recording: http://openprojects.rarcoa.com/~miztic/artifact.wav normally the artifacts are just short blips, not quite as long as the one above, but they sound the same. When using the aggressive echo suppressor, it seems like those artifacts cause a really loud buzzing sound to come out of the cisco phone, pretty much made using the aggressive canceler impossible to use, it's too bad because it worked the best out of all of them, mark3 works ok but still gives echos on at least 20% of the calls. I thought they might be caused by IRQ sharing, so I pulled one of the TDM400P cards out and made sure the remaining two were on their own IRQ, the artifacts were still there. I've also tried running a kernel with all the low-latency stuff turned on, and the same kernel with it all turned off (2.6.16-rc2) doesn't appear to make any difference either. I'm not sure what else to try, any input would be appreciated. Thanks, Gerard Saraber [EMAIL PROTECTED] hardware: AMD64 1.8Ghz 512M ram MSI nforce3 socket 754 mainboard 3 Digium TDM400P cards, 10 FXO + 2 FXS modules -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] odd 'digital' sound artifacts [solved]
Apparently I didn't do the first test correctly where I said I made sure I had only two cards, each on their own IRQ and still got artifacts. When I repeated the test today, with both cards on their own IRQ, I got no artifacts at all, after shuffeling the cards around for a bit I was able to get each of the 3 on their own IRQ, after 16 minutes of hold music, still no artifacts, so its safe to say my problem was a shared IRQ.. The only other two things that are different are that I switched to the SVN trunk version of Zaptel and recompiled my kernel a few times. My apologies to anyone who's time I wasted, thank you for looking.. Linux phonesys 2.6.16-rc2-mm1 #1 PREEMPT Wed Feb 8 11:09:28 CST 2006 x86_64 AMD Sempron(tm) Processor 3000+ AuthenticAMD GNU/Linux Zaptel Version: SVN-trunk-r941M Echo Canceller: MG2 CPU0 0:1183747IO-APIC-edge timer 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 56255 IO-APIC-level eth0 185: 5382 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217:1170531 IO-APIC-level wctdm 225:1169188 IO-APIC-level wctdm 233:1167705 IO-APIC-level wctdm NMI:195 LOC:1183483 ERR: 0 MIS: 0 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] more cpu intensive echo cancellers ?
On Fri, 2006-02-10 at 16:05 -0600, Matthew Fredrickson wrote: On Feb 10, 2006, at 1:21 PM, Gerard Saraber wrote: Found it, going to go test it right now :) thanks! So far reports have been positive on the echo, but its a slow day ;) We're using cisco 7960 phones, they're pricy, but they work great and sound good, if it wasn't for the echo issue, I would have been able to roll the whole setup out already. Actually that's not quite true, I still have to make the 7914 addon module work with the 7960 phone, but that's not a show stopper. Either way, so far big thumbs up for the MG2 echo can, and if any developers read this, feel free to add a compile flag to make it more cpu intensive ;) and do more canceling. Does latest MG2 behave better than KB1 on your analog lines? I heard in the past that in some cases (primarily with analog lines) that KB1 worked better. Also, have you tried the echotraining=800 (in zapata.conf) tweak as well? --- Matthew Fredrickson In my case, MG2 blows KB1 away, the trunk version is a huge improvement, in the past, echotraining=some number was always worse compared to echotraining=yes so I didn't change it. I'll definately try that if I get any echo complaints, so far, so good though. -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] odd 'digital' sound artifacts
So nobody heard these before? or did I do something stupid that anyone should know and nobody wanted to yell at me for it ;) On Wed, 2006-02-08 at 12:54 -0600, Gerard Saraber wrote: Hi, I've got some weird sound artifacts happening during calls, they're very hard to describe, so I have a 122kb recording: http://openprojects.rarcoa.com/~miztic/artifact.wav normally the artifacts are just short blips, not quite as long as the one above, but they sound the same. When using the aggressive echo suppressor, it seems like those artifacts cause a really loud buzzing sound to come out of the cisco phone, pretty much made using the aggressive canceler impossible to use, it's too bad because it worked the best out of all of them, mark3 works ok but still gives echos on at least 20% of the calls. I thought they might be caused by IRQ sharing, so I pulled one of the TDM400P cards out and made sure the remaining two were on their own IRQ, the artifacts were still there. I've also tried running a kernel with all the low-latency stuff turned on, and the same kernel with it all turned off (2.6.16-rc2) doesn't appear to make any difference either. I'm not sure what else to try, any input would be appreciated. Thanks, Gerard Saraber [EMAIL PROTECTED] hardware: AMD64 1.8Ghz 512M ram MSI nforce3 socket 754 mainboard 3 Digium TDM400P cards, 10 FXO + 2 FXS modules /proc/interrupts CPU0 0:2784232IO-APIC-edge timer 1: 8IO-APIC-edge i8042 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 71552 IO-APIC-level eth0 185: 9412 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217:5577811 IO-APIC-level wctdm, wctdm 225:2769262 IO-APIC-level wctdm lspci (for completeness): 02:07.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at ac00 [size=256] Memory at fdeff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 225 I/O ports at a800 [size=256] Memory at fdefe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at a400 [size=256] Memory at fdefd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] more cpu intensive echo cancellers ?
On Fri, 2006-02-10 at 11:01 -0600, Clint Sharp wrote: Gerard Saraber wrote: Thanks! testing it now, on my test calls it appears to start out with less echo then the Mark3 canceler, but it trains slower, seems like it took a long time for the echo to completely disappear, the real test will be seeing what the people at my company have to say. Feb 9 14:47:51 [kernel] Zapata Telephony Interface Registered on major 196 Feb 9 14:47:51 [kernel] Zaptel Version: SVN-trunk-r934M Echo Canceller: MG2 I've had really good luck with the echocan preload patch that was posted on the asterisk dev list a while back as well, and I've been recommending it to people as well. This has really helped minimize the echo problems to a minimal level, although I don't know about recommending this system to our customers. I still think a lot of my audio quality problems are being caused by my phones (not echo, but clicks and pops and various overmodulation problems). We're getting there, but I'm still nervous with trying to sell an * system to someone who is used to the quality of a traditional PBX or key system. Clint Found it, going to go test it right now :) thanks! So far reports have been positive on the echo, but its a slow day ;) We're using cisco 7960 phones, they're pricy, but they work great and sound good, if it wasn't for the echo issue, I would have been able to roll the whole setup out already. Actually that's not quite true, I still have to make the 7914 addon module work with the 7960 phone, but that's not a show stopper. Either way, so far big thumbs up for the MG2 echo can, and if any developers read this, feel free to add a compile flag to make it more cpu intensive ;) and do more canceling. -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] odd 'digital' sound artifacts
On Fri, 2006-02-10 at 11:20 -0800, Anthony Rodgers wrote: Your output looks like you have 3 cards, two of which are sharing interrupts - or am I missing something? That is correct, the thing is, I pulled one of the cards out (as stated in my first email), and made sure each was on their own IRQ, and I *still* got the same artifacts, so I'm not sure that IRQ sharing is the problem. /proc/interrupts CPU0 0:2784232IO-APIC-edge timer 1: 8IO-APIC-edge i8042 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 71552 IO-APIC-level eth0 185: 9412 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217:5577811 IO-APIC-level wctdm, wctdm 225:2769262 IO-APIC-level wctdm -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] more cpu intensive echo cancellers ?
I appreciate the input, but after doing a little research on that card, it looks like I'll still need the channel bank, I think with some carefull ebaying, I should be able to do the hardware canceling for about $1000 less then what i saw the 104d card for. not to mention it seems total overkill, I've got a wimpy 10 pstn phone lines, a quad T1 card seems a little excessive to me. If there was a single T1 with the G.168 echo canceler card for say $800, I'd be all over that (still researching). I've got all this extra cpu power, and nothing to use it on ;) in the mean time, I'll put the 104d card on the list of possibilities, Thanks, Gerard Saraber [EMAIL PROTECTED] On Wed, 2006-02-08 at 17:26 -0800, Canuck15 wrote: Gerard, Just get yourself a Sangoma card with hardware echo can and be done with it. It is worth every penny just for the headaches it will save you. It's a better solution for most situations compared to a channel bank. Cheaper, simpler and works just as good IMHO. -Original Message- From: Gerard Saraber [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 08, 2006 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] more cpu intensive echo cancellers ? Thanks for the quick reply :) It happens when we call a lot of different people, and obviously doesn't happen with our old analog phone system, so even if its caused by someone else, *we* still have to fix it. we're kind of weighing our options, I'm hoping to take care of this with some fancy software, but if not we'll be going the hardware canceller route. Thanks, Gerard Saraber [EMAIL PROTECTED] On Wed, 2006-02-08 at 11:45 -0800, Michael Collins wrote: Gerard, I'll bet your side is working great for echo cancellation. It sounds like the equipment at the other end of the call might need some help. You know the old rule if you and I are talking on the phone: If I hear echo, you've got a problem; if you hear echo, I've got a problem. If only all echo problems were so easy to diagnose! In any case, is it possible that some of the echo you're hearing is being caused by poor echo handling on the other end of the line? Just a thought. -MC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gerard Saraber Sent: Wednesday, February 08, 2006 11:03 AM To: Asterisk Users Mailing List Subject: [Asterisk-Users] more cpu intensive echo cancellers ? Hi, I've had some decent luck with the mark3 echo canceller from the zaptel driver, echos on about 20% of the calls, people I've called say I sound great now, but our side hears echos. I was wondering if there was any way to tweak the current software cancelers into using more CPU (and hopefully doing a better job, close to a hardware canceler), I only have 10 lines, and a single call takes 0.5% cpu, I would have no problem if it went up to 5-10% if they would work better. Or should I just give up now and buy the channel bank, tellabs hardware echo canceler and a T1 pci card? (hope TDM400P cards have decent resale value ;) -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] ztdummy on gentoo 2005.1
Invalid module format usually means that the module you're trying to load was not compiled with the same parameters as the kernel you're trying to load it into, make sure your /usr/src/linux symlink points to the kernel you are actually running, /var/log/messages etc. will usually have more specific information on why the module format is 'wrong' . I would suggest after checking the /usr/src/linux symlink, to recompile the kernel, the ztdummy module and booting into the newly compiled kernel. its possible that all it takes is to recompile the module though. Regards, Gerard Saraber [EMAIL PROTECTED] On Thu, 2006-02-09 at 09:30 -0600, Miguel wrote: I changed the line with your sugestion but same result (after reboot): voip # lsmod Module Size Used by voip # modprobe ztdummy FATAL: Error inserting ztdummy (/lib/modules/2.6.12-gentoo-r10/misc/ztdummy.o): Invalid module format FATAL: Error running install command for ztdummy voip mmiranda # --- Miguel ___ --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 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] more cpu intensive echo cancellers ?
On Thu, 2006-02-09 at 10:05 -0600, Matthew Fredrickson wrote: Yeah there is, upgrade to trunk and use the new echo canceller there (MG2). It's supposed to rock, at least from what I've heard. All the MEC cancellers are _OLD_. At least switch to 1.2 and the KB1 echo canceler before giving up. --- Matthew Fredrickson Aha! good to know, I am running asterisk 1.2.4 and zaptel-1.2.3, should I switch to CVS ? I've tried the MG2 canceler with the above versions, each time I tried it, I had a constant echo, where with the mark3 it went away after a second or two at the beginning of the call. (which I can live with, but some of the calls are completely unusable due to continuous or returning echos) I'll go play with the mg2 and kb1 again and see what happens -- Thanks, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] more cpu intensive echo cancellers ?
On Thu, 2006-02-09 at 14:16 -0600, Matthew Fredrickson wrote: Try MG2 with trunk and KB1 with 1.2. KB1 is supposed to be fairly reliable in 1.2, and MG2 in trunk has a good possibility of outperforming KB1 from 1.2. Matthew Fredrickson Thanks! testing it now, on my test calls it appears to start out with less echo then the Mark3 canceler, but it trains slower, seems like it took a long time for the echo to completely disappear, the real test will be seeing what the people at my company have to say. Feb 9 14:47:51 [kernel] Zapata Telephony Interface Registered on major 196 Feb 9 14:47:51 [kernel] Zaptel Version: SVN-trunk-r934M Echo Canceller: MG2 -- Thanks again, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] odd 'digital' sound artifacts
Hi, I've got some weird sound artifacts happening during calls, they're very hard to describe, so I have a 122kb recording: http://openprojects.rarcoa.com/~miztic/artifact.wav normally the artifacts are just short blips, not quite as long as the one above, but they sound the same. When using the aggressive echo suppressor, it seems like those artifacts cause a really loud buzzing sound to come out of the cisco phone, pretty much made using the aggressive canceler impossible to use, it's too bad because it worked the best out of all of them, mark3 works ok but still gives echos on at least 20% of the calls. I thought they might be caused by IRQ sharing, so I pulled one of the TDM400P cards out and made sure the remaining two were on their own IRQ, the artifacts were still there. I've also tried running a kernel with all the low-latency stuff turned on, and the same kernel with it all turned off (2.6.16-rc2) doesn't appear to make any difference either. I'm not sure what else to try, any input would be appreciated. Thanks, Gerard Saraber [EMAIL PROTECTED] hardware: AMD64 1.8Ghz 512M ram MSI nforce3 socket 754 mainboard 3 Digium TDM400P cards, 10 FXO + 2 FXS modules /proc/interrupts CPU0 0:2784232IO-APIC-edge timer 1: 8IO-APIC-edge i8042 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 71552 IO-APIC-level eth0 185: 9412 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217:5577811 IO-APIC-level wctdm, wctdm 225:2769262 IO-APIC-level wctdm lspci (for completeness): 02:07.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at ac00 [size=256] Memory at fdeff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 225 I/O ports at a800 [size=256] Memory at fdefe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at a400 [size=256] Memory at fdefd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] more cpu intensive echo cancellers ?
Hi, I've had some decent luck with the mark3 echo canceller from the zaptel driver, echos on about 20% of the calls, people I've called say I sound great now, but our side hears echos. I was wondering if there was any way to tweak the current software cancelers into using more CPU (and hopefully doing a better job, close to a hardware canceler), I only have 10 lines, and a single call takes 0.5% cpu, I would have no problem if it went up to 5-10% if they would work better. Or should I just give up now and buy the channel bank, tellabs hardware echo canceler and a T1 pci card? (hope TDM400P cards have decent resale value ;) -- Thanks, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] more cpu intensive echo cancellers ?
Thanks for the quick reply :) It happens when we call a lot of different people, and obviously doesn't happen with our old analog phone system, so even if its caused by someone else, *we* still have to fix it. we're kind of weighing our options, I'm hoping to take care of this with some fancy software, but if not we'll be going the hardware canceller route. Thanks, Gerard Saraber [EMAIL PROTECTED] On Wed, 2006-02-08 at 11:45 -0800, Michael Collins wrote: Gerard, I'll bet your side is working great for echo cancellation. It sounds like the equipment at the other end of the call might need some help. You know the old rule if you and I are talking on the phone: If I hear echo, you've got a problem; if you hear echo, I've got a problem. If only all echo problems were so easy to diagnose! In any case, is it possible that some of the echo you're hearing is being caused by poor echo handling on the other end of the line? Just a thought. -MC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gerard Saraber Sent: Wednesday, February 08, 2006 11:03 AM To: Asterisk Users Mailing List Subject: [Asterisk-Users] more cpu intensive echo cancellers ? Hi, I've had some decent luck with the mark3 echo canceller from the zaptel driver, echos on about 20% of the calls, people I've called say I sound great now, but our side hears echos. I was wondering if there was any way to tweak the current software cancelers into using more CPU (and hopefully doing a better job, close to a hardware canceler), I only have 10 lines, and a single call takes 0.5% cpu, I would have no problem if it went up to 5-10% if they would work better. Or should I just give up now and buy the channel bank, tellabs hardware echo canceler and a T1 pci card? (hope TDM400P cards have decent resale value ;) -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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
