[asterisk-users] Choppy Audio in One Direction
Hello everyone, We had one of our PBXs crash due to a hardware failure, and rebuilt it with PBX in a Flash. We are using the current versions of libpri, zaptel and *. It's the same server with replacement hard drives - a Dell 2850 with a TE410 T1 card, single PRI. It was running v1.2 for years with no problems at all. I'll call this PBX B. Our main PBX in our corporate office - PBX A - has also been upgraded to the latest libpri, zaptel and * (not running PIAF, but fc7 and a manual installation), although the same problem occurred with the previous version. We also tried different hardware on PBX B with the same symptoms. What is happening is that occasionally, audio from PBX B to PBX A will become choppy, while the audio from PBX A to PBX B is unaffected. It doesn't seem to matter how many active calls there are, and we've tried SIP and IAX trunks with the same result. The issue will go away for awhile, and then return. It just happened again, and ping times with full-size packets are a consistent 85-90 ms. If you call through the PSTN, then the symptoms disappear. It could be a networking problem, but this didn't start happening until the server crash, and nothing has changed in the networking infrastructure. Cat /proc/interrupts show that there are no shared IRQs. We just set the switch port to force 100Mb, Full-Duplex. The OS on PBX B is CentOS 5.2, kernel 2.6.18-92.1.6.el5. Hyperthreading is turned on, just as it has always been on the server. Can anyone give my any ideas as to what might be going on? Thanks for any help, Kevin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
Eric ManxPower Wieling wrote: Make the card stop sharing it's IRQ with your IDE controller. Try moving the card to another slot. Asterisk has to send an audio packet every 20ms for VoIP calls. I believe Zaptel expects no more than a few ms of latency. If something is causing a delay, like the IDE controller locking interrupts and doing disk activity then you're not going to get your interrupts serviced fast enough and you will have audio issues. Good suggestion. Can I also recommend to use the same types of soundfiles as your current playing channel? (eg ulaw sounds for phones using the ulaw codec) -bk Doug Crompton wrote: I am not sure who all see's this list but I do have a few questions that probably only the developers or somone really in the know of Asterisk could answer. - What is the requirement for timing vs. audio playback in Asterisk. Specifically voicemail and IVR's (Not meetme) - Has this requirment changed in newer versions? This obviously is when using Asterisk with no internal cards. I used Asterisk for several years with a P3 Linux system, NO timing, and it worked flawlessly. Now with this new Pentium Dual core system I do not have the perfect audio I experienced with the less powerful system. I fully know there are MANY variable here. It could be a combination of many things, including the OS (Linux Kernel) etc. BUT I offer this input, Music on Hold works fine. This uses mpg123. So why can this palyback fine and the other wav/gsm audio be choppy? I would gladly switch to a newer Asterisk (using 1.2.29) if someone said this was solved in that version. My system can obviously play (mpg123 - background) audio fine. Why then does Asterisk internal audio not also play well? Doug * Doug Crompton* * Richboro, PA 18954 * * 215-431-6307 * * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
OK in my research here is what I found. I seem to get the idea from what I read that ztdummy is not needed for 1.2 (and above) versions for anything but meetme. I never used ztdummy in my old system and it worked just fine. I really see no difference here with it running or not. Please confirm - is this timing required for just simple wav/gsm playback - like voicmail, etc. ? I played around with kernel options echo 0 /sys/devices/system/cpu/cpu1/online echo performance /sys/devices/system/cpu/cpu0/cpufreq/scaling_governor Which essentially makes it a 1 CPU system at max performance. NO CHANGE. Then put it back... echo ondemand /sys/devices/system/cpu/cpu0/cpufreq/scaling_governor echo 1 /sys/devices/system/cpu/cpu1/online Tried this echo 1 /sys/module/processor/parameters/max_cstate Default was 8. NO CHANGE. Let me reconfirm what I am hearing. The audio choppiness is subtle but definitely there. It seems to happen at the exact SAME place everytime I play it, which is suspicious! Could this be sometihng completely different then what we are suspecting? It seems to get out of sync. sometimes it seems it is playing future audio on top of current by only Ms's or maybe it is putting a hole there. Hard to tell. OH I also tried compiling with the O2 optimization rather then the O8 in the default Asterisk Makefile. Again NO CHANGE. Any more ideas would be helpful. Doug On Tue, 1 Jul 2008, Benjamin Jacob wrote: modprobe zaptel; modprobe ztdummy That will start zaptel and ztdummy after the 'zaptel stop'. Then restart asterisk. --- On Wed, 7/2/08, Doug Crompton [EMAIL PROTECTED] wrote: From: Doug Crompton [EMAIL PROTECTED] Subject: Re: [asterisk-users] Choppy audio To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, July 2, 2008, 1:58 AM OK just to be clear on what you recommend... Stop everything, unload zaptel and zrdummy modules... then just restart asterisk? Does it start zaptel? This is NOT a slow box. P6 dual core 4 gig cache, 3800 bogomips. Doug On Tue, 1 Jul 2008, bkruse wrote: I would recommend stopping asterisk (/etc/init.d/asterisk stop) /etc/init.d/zaptel stop (unload all modules) modprobe zaptel; modprobe ztdummy (in the case that you don't have another card for a timing device) /etc/init.d/asterisk start ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
On Tue, Jul 01, 2008 at 02:47:34PM -0500, spectro wrote: On Tue, Jul 1, 2008 at 2:15 AM, Doug Crompton [EMAIL PROTECTED] wrote: Using 1.2.29 on Linux version 2.6.18.8-0.3 (SUSE 10.2) Intel 686 dual core. I am getting choppy audio in voicemail and general message playback. see if disabling APM in your kernel solves the issue, add apm=off to kernel boot options. Anything still uses APM (as oposed to ACPI) noawadays? $ /sbin/acpi_available echo yes yes $ /sbin/apm_available echo yes (nothing) -- 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 http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
Tzafrir, I have neither of those commands available here. Did a search to see if they were somewhere else but nothing. Using SUSE 10.2. In /proc I only have apci directory. Doug On Wed, 2 Jul 2008, Tzafrir Cohen wrote: On Tue, Jul 01, 2008 at 02:47:34PM -0500, spectro wrote: On Tue, Jul 1, 2008 at 2:15 AM, Doug Crompton [EMAIL PROTECTED] wrote: Using 1.2.29 on Linux version 2.6.18.8-0.3 (SUSE 10.2) Intel 686 dual core. I am getting choppy audio in voicemail and general message playback. see if disabling APM in your kernel solves the issue, add apm=off to kernel boot options. Anything still uses APM (as oposed to ACPI) noawadays? $ /sbin/acpi_available echo yes yes $ /sbin/apm_available echo yes (nothing) -- 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 http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
I am not sure who all see's this list but I do have a few questions that probably only the developers or somone really in the know of Asterisk could answer. - What is the requirement for timing vs. audio playback in Asterisk. Specifically voicemail and IVR's (Not meetme) - Has this requirment changed in newer versions? This obviously is when using Asterisk with no internal cards. I used Asterisk for several years with a P3 Linux system, NO timing, and it worked flawlessly. Now with this new Pentium Dual core system I do not have the perfect audio I experienced with the less powerful system. I fully know there are MANY variable here. It could be a combination of many things, including the OS (Linux Kernel) etc. BUT I offer this input, Music on Hold works fine. This uses mpg123. So why can this palyback fine and the other wav/gsm audio be choppy? I would gladly switch to a newer Asterisk (using 1.2.29) if someone said this was solved in that version. My system can obviously play (mpg123 - background) audio fine. Why then does Asterisk internal audio not also play well? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
Make the card stop sharing it's IRQ with your IDE controller. Try moving the card to another slot. Asterisk has to send an audio packet every 20ms for VoIP calls. I believe Zaptel expects no more than a few ms of latency. If something is causing a delay, like the IDE controller locking interrupts and doing disk activity then you're not going to get your interrupts serviced fast enough and you will have audio issues. Doug Crompton wrote: I am not sure who all see's this list but I do have a few questions that probably only the developers or somone really in the know of Asterisk could answer. - What is the requirement for timing vs. audio playback in Asterisk. Specifically voicemail and IVR's (Not meetme) - Has this requirment changed in newer versions? This obviously is when using Asterisk with no internal cards. I used Asterisk for several years with a P3 Linux system, NO timing, and it worked flawlessly. Now with this new Pentium Dual core system I do not have the perfect audio I experienced with the less powerful system. I fully know there are MANY variable here. It could be a combination of many things, including the OS (Linux Kernel) etc. BUT I offer this input, Music on Hold works fine. This uses mpg123. So why can this palyback fine and the other wav/gsm audio be choppy? I would gladly switch to a newer Asterisk (using 1.2.29) if someone said this was solved in that version. My system can obviously play (mpg123 - background) audio fine. Why then does Asterisk internal audio not also play well? Doug * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choppy audio
Using 1.2.29 on Linux version 2.6.18.8-0.3 (SUSE 10.2) Intel 686 dual core. I just switched over to this system from an older SUSE 2.4.10 kernel system. I am getting choppy audio in voicemail and general message playback. I installed Zaptel and ztdummy module and the following is zaptel status: slate:/etc/init.d # cat /proc/zaptel/1 Span 1: ZTDUMMY/1 ZTDUMMY/1 (source: RTC) 1 Is this indicating proper installation? Is there anything else I should try/do?? The choppyness is not extreme, just not perfect. I had no problem in my old system with 2.4. I had not even installed zaptel or ztdummy there. Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
As an addendum to my original message... In my research it appears this often happens when using more than one processor. I am using a dual core Pentium. I guess my dilema here is which way to go. Clearly the audio is not working the way I would like it to and the way I came to expect from my old system. When playing messages it seems to get out of sync. Sometimes skipping ms's of audio. This seems to happen at about a 2-4 second rate. I believe that I have things setup to use the RTC as a timing device (see below) but that did not seem to change the problem. It may have made it better but not much. What are my choices? HW card?, Upgrade Asterisk?, Doug On Tue, 1 Jul 2008, Doug Crompton wrote: Using 1.2.29 on Linux version 2.6.18.8-0.3 (SUSE 10.2) Intel 686 dual core. I just switched over to this system from an older SUSE 2.4.10 kernel system. I am getting choppy audio in voicemail and general message playback. I installed Zaptel and ztdummy module and the following is zaptel status: slate:/etc/init.d # cat /proc/zaptel/1 Span 1: ZTDUMMY/1 ZTDUMMY/1 (source: RTC) 1 Is this indicating proper installation? Is there anything else I should try/do?? The choppyness is not extreme, just not perfect. I had no problem in my old system with 2.4. I had not even installed zaptel or ztdummy there. Doug * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
Hi Doug - In my research it appears this often happens when using more than one processor. I am using a dual core Pentium. I guess my dilema here is which way to go. Clearly the audio is not working the way I would like it to and the way I came to expect from my old system. When playing messages it seems to get out of sync. Sometimes skipping ms's of audio. This seems to happen at about a 2-4 second rate. I believe that I have things setup to use the RTC as a timing device (see below) but that did not seem to change the problem. It may have made it better but not much. What are my choices? HW card?, Upgrade Asterisk?, The symptoms don't sound exactly the same, but is it possible that this is the GSM/GCC playback bug? http://bugs.digium.com/view.php?id=11243 - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
I saw that bug. Most of my files are WAV though. Would it apply to them also? Doug On Tue, 1 Jul 2008, Noah Miller wrote: Hi Doug - In my research it appears this often happens when using more than one processor. I am using a dual core Pentium. I guess my dilema here is which way to go. Clearly the audio is not working the way I would like it to and the way I came to expect from my old system. When playing messages it seems to get out of sync. Sometimes skipping ms's of audio. This seems to happen at about a 2-4 second rate. I believe that I have things setup to use the RTC as a timing device (see below) but that did not seem to change the problem. It may have made it better but not much. What are my choices? HW card?, Upgrade Asterisk?, The symptoms don't sound exactly the same, but is it possible that this is the GSM/GCC playback bug? http://bugs.digium.com/view.php?id=11243 - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
On Tue, Jul 1, 2008 at 2:15 AM, Doug Crompton [EMAIL PROTECTED] wrote: Using 1.2.29 on Linux version 2.6.18.8-0.3 (SUSE 10.2) Intel 686 dual core. I am getting choppy audio in voicemail and general message playback. see if disabling APM in your kernel solves the issue, add apm=off to kernel boot options. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
Doug Crompton wrote: I saw that bug. Most of my files are WAV though. Would it apply to them also? Doug On Tue, 1 Jul 2008, Noah Miller wrote: Hi Doug - In my research it appears this often happens when using more than one processor. I am using a dual core Pentium. I guess my dilema here is which way to go. Clearly the audio is not working the way I would like it to and the way I came to expect from my old system. When playing messages it seems to get out of sync. Sometimes skipping ms's of audio. This seems to happen at about a 2-4 second rate. I believe that I have things setup to use the RTC as a timing device (see below) but that did not seem to change the problem. It may have made it better but not much. What are my choices? HW card?, Upgrade Asterisk?, The symptoms don't sound exactly the same, but is it possible that this is the GSM/GCC playback bug? http://bugs.digium.com/view.php?id=11243 - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I would recommend stopping asterisk (/etc/init.d/asterisk stop) /etc/init.d/zaptel stop (unload all modules) modprobe zaptel; modprobe ztdummy (in the case that you don't have another card for a timing device) /etc/init.d/asterisk start If it is a relatively slow box, try getting the exact sound files you will be playing back, if you have the space (make menuselect). -bk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
OK just to be clear on what you recommend... Stop everything, unload zaptel and zrdummy modules... then just restart asterisk? Does it start zaptel? This is NOT a slow box. P6 dual core 4 gig cache, 3800 bogomips. Doug On Tue, 1 Jul 2008, bkruse wrote: I would recommend stopping asterisk (/etc/init.d/asterisk stop) /etc/init.d/zaptel stop (unload all modules) modprobe zaptel; modprobe ztdummy (in the case that you don't have another card for a timing device) /etc/init.d/asterisk start If it is a relatively slow box, try getting the exact sound files you will be playing back, if you have the space (make menuselect). -bk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
modprobe zaptel; modprobe ztdummy That will start zaptel and ztdummy after the 'zaptel stop'. Then restart asterisk. --- On Wed, 7/2/08, Doug Crompton [EMAIL PROTECTED] wrote: From: Doug Crompton [EMAIL PROTECTED] Subject: Re: [asterisk-users] Choppy audio To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, July 2, 2008, 1:58 AM OK just to be clear on what you recommend... Stop everything, unload zaptel and zrdummy modules... then just restart asterisk? Does it start zaptel? This is NOT a slow box. P6 dual core 4 gig cache, 3800 bogomips. Doug On Tue, 1 Jul 2008, bkruse wrote: I would recommend stopping asterisk (/etc/init.d/asterisk stop) /etc/init.d/zaptel stop (unload all modules) modprobe zaptel; modprobe ztdummy (in the case that you don't have another card for a timing device) /etc/init.d/asterisk start ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] choppy audio sip - capi
Further to my previous email, I have definitely established that the audio gets choppy only when the path includes sip and capi. PAP2 to Asterisk to MyNetFone to PSTN is fine. PAP2 to Asterisk MOH is fine. PBX (via capi) to Asterisk MOH is fine PBX (via capi) to Asterisk to PAP2 is choppy PBX (via capi) to Asterisk to MyNetFone to PSTN is choppy (PAP2 is a LinkSys FXS ATA) SIP phone, PBX, and MyNetFone are all configured to use alaw (G.711a), so transcoding should be almost irrelevant. Network bandwidth is not a problem because pure SIP calls are crystal clear - people I have called cannot tell the difference. I also have a X100 card which is providing timing. Nothing is sharing interrupts with anything. Any suggestions? Thanks James ___ --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] Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)
Do you have silence suppression enabled on your clients? Asterisk can't work with silence suppression. Take a look at http://bugs.digium.com/view.php?id=5374 , the patch works fine to me, now I'm able to set SS and save bandwidth. On Mon, 2005-11-07 at 19:41 -0800, Chris Tracy wrote: I recently resurrected an old athlon system and put CentOS 4.2 on it to play with asterisk. First I tried asterisk-1.0.9, now I'm using 1.2.0-b2. Both have the same audio issues that have me stumped. I looked through all the lists and forums and the closest I could get were some messages from 2003: http://lists.digium.com/pipermail/asterisk-users/2003-August/017171.html I've got asterisk set up with my xten-lite softphone on extension 200 over SIP. I've configured extension 611 as an echo test and 612 will play 30 seconds of MusicOnHold. I can connect to both just fine, however, they sound rather bad when they work (quite muddy) and periodically they just drop out for as much as 5 seconds before coming back. Enabling all the debugging and verbosity options, I've found a few messages that occur during each drop. During the MOH run, every time there's a drop, the console scrolls: res_musiconhold.c:535 monmp3thread: Only wrote -1 of 640 bytes to pipe over and over until the sound comes back, at which point, the console message: rtp.c:1247 ast_rtp_raw_write: Difference is 33824, ms is 4248 is displayed. (Not always the same numbers in that one, obviously) In the echo test, again, after a drop, the audio returns and a message similar to: rtp.c:1247 ast_rtp_raw_write: Difference is 12496, ms is 1582 is displayed. The asterisk server is on a single Athlon MP 1600+ (1.4GHz) with 512MB of RAM. It's got a K7D-Master mobo, and is connected to the system running the softphone through a 100Mbit LAN. I've not enabled any of the MMX optimizations as there were warnings that they didn't play nice with AMD chips. If there's any further info I can provide, I'd be happy to. Thanks, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)
I recently resurrected an old athlon system and put CentOS 4.2 on it to play with asterisk. First I tried asterisk-1.0.9, now I'm using 1.2.0-b2. Both have the same audio issues that have me stumped. I looked through all the lists and forums and the closest I could get were some messages from 2003: http://lists.digium.com/pipermail/asterisk-users/2003-August/017171.html I've got asterisk set up with my xten-lite softphone on extension 200 over SIP. I've configured extension 611 as an echo test and 612 will play 30 seconds of MusicOnHold. I can connect to both just fine, however, they sound rather bad when they work (quite muddy) and periodically they just drop out for as much as 5 seconds before coming back. In the xlite configuration, look for an option something like 'transmit silence' and set that to yes. (Might be called 'silence suppression', I don't remember.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)
Rich Adamson wrote: I recently resurrected an old athlon system and put CentOS 4.2 on it to play with asterisk. First I tried asterisk-1.0.9, now I'm using 1.2.0-b2. Both have the same audio issues that have me stumped. I looked through all the lists and forums and the closest I could get were some messages from 2003: http://lists.digium.com/pipermail/asterisk-users/2003-August/017171.html I've got asterisk set up with my xten-lite softphone on extension 200 over SIP. I've configured extension 611 as an echo test and 612 will play 30 seconds of MusicOnHold. I can connect to both just fine, however, they sound rather bad when they work (quite muddy) and periodically they just drop out for as much as 5 seconds before coming back. In the xlite configuration, look for an option something like 'transmit silence' and set that to yes. (Might be called 'silence suppression', I don't remember.) I think it's called VAD (Voice Activity Detection) in xten products. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)
Well, that fixed the choppiness. (Transmit Silence needed to be enabled) Now to figure out why it sounds so bad. Thanks for the helpful replies. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)
I recently resurrected an old athlon system and put CentOS 4.2 on it to play with asterisk. First I tried asterisk-1.0.9, now I'm using 1.2.0-b2. Both have the same audio issues that have me stumped. I looked through all the lists and forums and the closest I could get were some messages from 2003: http://lists.digium.com/pipermail/asterisk-users/2003-August/017171.html I've got asterisk set up with my xten-lite softphone on extension 200 over SIP. I've configured extension 611 as an echo test and 612 will play 30 seconds of MusicOnHold. I can connect to both just fine, however, they sound rather bad when they work (quite muddy) and periodically they just drop out for as much as 5 seconds before coming back. Enabling all the debugging and verbosity options, I've found a few messages that occur during each drop. During the MOH run, every time there's a drop, the console scrolls: res_musiconhold.c:535 monmp3thread: Only wrote -1 of 640 bytes to pipe over and over until the sound comes back, at which point, the console message: rtp.c:1247 ast_rtp_raw_write: Difference is 33824, ms is 4248 is displayed. (Not always the same numbers in that one, obviously) In the echo test, again, after a drop, the audio returns and a message similar to: rtp.c:1247 ast_rtp_raw_write: Difference is 12496, ms is 1582 is displayed. The asterisk server is on a single Athlon MP 1600+ (1.4GHz) with 512MB of RAM. It's got a K7D-Master mobo, and is connected to the system running the softphone through a 100Mbit LAN. I've not enabled any of the MMX optimizations as there were warnings that they didn't play nice with AMD chips. If there's any further info I can provide, I'd be happy to. Thanks, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold(1.2.0-b2)
We had problems with music on hold and finally decided to move to option 2 on the faking it document. We have not had any trouble since. Good luck. http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it Regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Tracy Sent: Tuesday, November 08, 2005 2:42 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold(1.2.0-b2) I recently resurrected an old athlon system and put CentOS 4.2 on it to play with asterisk. First I tried asterisk-1.0.9, now I'm using 1.2.0-b2. Both have the same audio issues that have me stumped. I looked through all the lists and forums and the closest I could get were some messages from 2003: http://lists.digium.com/pipermail/asterisk-users/2003-August/017171.html I've got asterisk set up with my xten-lite softphone on extension 200 over SIP. I've configured extension 611 as an echo test and 612 will play 30 seconds of MusicOnHold. I can connect to both just fine, however, they sound rather bad when they work (quite muddy) and periodically they just drop out for as much as 5 seconds before coming back. Enabling all the debugging and verbosity options, I've found a few messages that occur during each drop. During the MOH run, every time there's a drop, the console scrolls: res_musiconhold.c:535 monmp3thread: Only wrote -1 of 640 bytes to pipe over and over until the sound comes back, at which point, the console message: rtp.c:1247 ast_rtp_raw_write: Difference is 33824, ms is 4248 is displayed. (Not always the same numbers in that one, obviously) In the echo test, again, after a drop, the audio returns and a message similar to: rtp.c:1247 ast_rtp_raw_write: Difference is 12496, ms is 1582 is displayed. The asterisk server is on a single Athlon MP 1600+ (1.4GHz) with 512MB of RAM. It's got a K7D-Master mobo, and is connected to the system running the softphone through a 100Mbit LAN. I've not enabled any of the MMX optimizations as there were warnings that they didn't play nice with AMD chips. If there's any further info I can provide, I'd be happy to. Thanks, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Choppy audio
I have an a problem with audio between phones (if both are SIP, or one is SIP andthen AIX to the PSTN phone). When music on hold plays it is choppy, and the console always shows messages like: monmp3thread XXX bytes of audio while expecting What is the cause / what can I do to eliminate this? Thanks, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] choppy audio.
Hi: any thoughts on this? I am using broadvoice, and am creating an information line for public resources. However, the audio is very choppy. I even notice when I call myself from an external source, my voicemail is also choppy. However, enternally it sounds wonderful. Any help? Chris - Original Message - From: Howard Lowndes [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, January 20, 2005 10:02 PM Subject: [Asterisk-Users] Zap randomly hanging up I have a zap line on a X101P which will occasionally just hang up the call for no apparent reason. Is there any good way of trying to diagnose what might be causing this? Monitoring the asterisk output in verbose mode does not provide any indications. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
