Hello, I am running the following configuration: 2.8ghz P4 with 1GB of RAM Audiocodes MP-108 connected to 5 POTS lines Polycom IP-500 phones Asterisk@Home 1.3 (this is Asterisk 1.0.9) End users are complaining of an echo and static on the inside end (the internal side), but the outside end of the conversation doeesn't notice anything. Does anyone have any suggestions on troubleshooting / fixing this problem? Thanks! Jeff
Jeff Busch wrote:>Hello, > >I am running the following configuration: > >2.8ghz P4 with 1GB of RAM >Audiocodes MP-108 connected to 5 POTS lines >Polycom IP-500 phones >Asterisk@Home 1.3 (this is Asterisk 1.0.9) > >End users are complaining of an echo and static on the inside end (the >internal side), but the outside end of the conversation doeesn't notice >anything. > > >I'm assuming inside means on the IP network and outside means on the PSTN which is accessible via a gateway is this correct? If so do you have QoS enabled on the inside?>Does anyone have any suggestions on troubleshooting / fixing this >problem? > >Thanks! > >Jeff > > > >_______________________________________________ >--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 > >
You are correct. Inside means the IP Network and Outside means the PSTN accessible via the Audiocodes MP-108 gateway. No QoS. Thanks - Jeff -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steve Blair Sent: Tuesday, November 29, 2005 4:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Static on inside end of conversation Jeff Busch wrote:>Hello, > >I am running the following configuration: > >2.8ghz P4 with 1GB of RAM >Audiocodes MP-108 connected to 5 POTS lines Polycom IP-500 phones >Asterisk@Home 1.3 (this is Asterisk 1.0.9) > >End users are complaining of an echo and static on the inside end (the >internal side), but the outside end of the conversation doeesn't notice>anything. > > >I'm assuming inside means on the IP network and outside means on the PSTN which is accessible via a gateway is this correct? If so do you have QoS enabled on the inside?>Does anyone have any suggestions on troubleshooting / fixing this >problem? > >Thanks! > >Jeff > > > >_______________________________________________ >--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
This definitely could be the issue. I am running 15 total devices (7 IP500 phones and 7 PC's along with a networked fax/scanner, and the Asterisk Server) through a single 16 port switch. We run one MS Access app on almost all the desktops that is a client/server app that creates a lot of traffic. I will go to the site tomorrow and make some changes to the topology of the LAN. Thanks! Jeff -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steve Blair Sent: Tuesday, November 29, 2005 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Static on inside end of conversation Jeff Busch wrote:>You are correct. Inside means the IP Network and Outside means the >PSTN accessible via the Audiocodes MP-108 gateway. > >No QoS. > > >We've seen echo on congested LANs within our Enterprise. I'm not sure if this fits what your seeing or not. We've placed phones in their own vlan and added 802.1p QoS to expedite forwarding upto the first hop router for the outbound call leg and the echo is gone. We did the same for the inbound call leg also.>Thanks - Jeff > >-----Original Message----- >From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steve >Blair >Sent: Tuesday, November 29, 2005 4:20 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] Static on inside end of conversation > > > >Jeff Busch wrote: > > > >>Hello, >> >>I am running the following configuration: >> >>2.8ghz P4 with 1GB of RAM >>Audiocodes MP-108 connected to 5 POTS lines Polycom IP-500 phones >>Asterisk@Home 1.3 (this is Asterisk 1.0.9) >> >>End users are complaining of an echo and static on the inside end (the>>internal side), but the outside end of the conversation doeesn't >>notice >> >> > > > >>anything. >> >> >> >> >> >I'm assuming inside means on the IP network and outside means on the >PSTN which is accessible via a gateway is this correct? If so do you >have QoS enabled on the inside? > > > >>Does anyone have any suggestions on troubleshooting / fixing this >>problem? >> >>Thanks! >> >>Jeff >> >> >> >>_______________________________________________ >>--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
----- Original Message ----- From: "Jeff Busch" <Jeff.Busch@lewisbuilds.com>> Hello, > > I am running the following configuration: > > 2.8ghz P4 with 1GB of RAM > Audiocodes MP-108 connected to 5 POTS lines > Polycom IP-500 phones > Asterisk@Home 1.3 (this is Asterisk 1.0.9) > > End users are complaining of an echo and static on the inside end (the > internal side), but the outside end of the conversation doeesn't notice > anything. > > Does anyone have any suggestions on troubleshooting / fixing this > problem? >Hi Jeff, I recommend upgrading to Asterisk@home 2.0 which was just released. It uses Asterisk 1.2 and a 2.6.9 based kernel which handles i/o and interrupts much better. While the link below discusses issues with digium cards, in general the interrupts and IDE vs SATA drive discussions are of use no matter what. http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html I have 2 Digium T400P cards connected to 8 POTS lines and 10 sipura spa-841 phones. I went through two PCs we had (2.9 GHZ celeron, 2.1HGZ Athlon XP) both with IDE drives. I finally declared war on echo and the Rice Krispies syndrome (Snap, Crackle, Pop) on the internal end of the conversations. I went and bought an ASUS P5LD2 motherboard with 1GB memory, 3.2GHz P4 with hyperthreading and a 2MB cache, and a SATA drive. I installed Asterisk@Home 2.0. After some minor magic to get the correct merlin gigabit ethernet driver for CentOS 4.2, everything came up perfect. This was my thanksgiving project and so far sound quality has been perfect. No echo and no rice krispies. I simulated network load on the system by copying multi-GBs of files through the net from another server with scp while I called out and back into the system on multiple lines. Even with the scp reporting 9.5MB/sec, the phone sound quality was fantastic. I then upped the ante by copying multi-gb files on the hard drive which when viewing stats with top (hyperthreading shows as 2 CPUs and you run the SMP kernel) both CPUs showed no idle time. IO- Wait state never greater than 10%. Phone calls were still perfect with no echo or noise. Out of the 10 vmails I left as part of the test, only one had 3 very faint pops in a 30 second message. They could have come from the POTS line for all I know. I ran extended phone conversations by calling the Asterisk system from our old phone system, picking up the extension on the called SIP phone and then playing Law & Order dvd episodes (lots of talking) and placing the handset near to the speaker and taking the old phone system handset and listening and talking back into it for 30 minutes at a time. Necessity is indeed a mother.... ;-) The calls were perfect. I'm amazed at how clear the dvd audio came through. I then reversed the process and played the dvd audio through the Asterisk system handsets while listening and talking back through the old phone handsets. After 30 minutes the quality was still excellent. Hope you find some of this ramble useful. Mike
Damian Funnell
2005-Nov-29 19:27 UTC
[Asterisk-Users] Static on inside end of conversation
We had a similar problem a while back and found that it was being caused by Hyperthreading. If you are using analogue cards then unfortunately you need to disable H/T if you haven't already done so. You also need to confirm that your fxo/fxs card isn't sharing IRQ's with anything. Don't trust 'cat /proc/interrupts', use 'lspci -v' instead. Also got this advice from Digium, although it has never applied to us:>>> If you are running an IDE hard drive please verify that you are using >>> DMA mode with a UDMA setting of no lower than 2 or higher than 3. UDMA >>> mode 2 is ATA33. UDMA mode 3 is ATA44. This can be done using hdparm. >>> We suggest using "hdparm -d 1 -X udma2 -c 3 /dev/[IDE Device]". You >>> can check the status using "hdparm /dev/[IDE Device]" and "hdparm -i >>> /dev/[IDE Device]". If you make modifications to your IDE hard drive >>> settings they will only be kept until you reboot. > > Not sure if this is what is causing your issues or not, but hopefully it will > help if it is. > > Cheers, > Damian Funnell. > > FFF Managed Technology Ltd. > 60 Cook St > P.O. 6368 Wellesley St > Auckland > t +64 9 356 2911 > f +64 9 358 9070 > m +64 21 415 297 > w www.fff.co.nz > > > > Quoting Health Masters <techsupport@progressivehomehealth.com>: > >> We have the same problem lately we thought maybe our upgrade and >> testing of the .13 firmware.. but we are running 20 phones on a p4 >> 2.6 w/512 4 PSTN lines on TDM400P and have 3 fat client pc's and 7 >> pcexpanions ( http://ncomputing.com/ ) running to a fat client. We >> did not have the issue on firmware .09 or .12. What firmware are you >> using? >> >> Jeff Busch wrote: >> >>> This definitely could be the issue. I am running 15 total devices (7 >>> IP500 phones and 7 PC's along with a networked fax/scanner, and the >>> Asterisk Server) through a single 16 port switch. We run one MS >>> Access app on almost all the desktops that is a >>> client/server app that creates a lot of traffic. I will go to the site >>> tomorrow and make some changes to the topology of the LAN. >>> >>> Thanks! >>> >>> Jeff >>> >>> -----Original Message----- >>> From: asterisk-users-bounces@lists.digium.com >>> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steve >>> Blair >>> Sent: Tuesday, November 29, 2005 4:33 PM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: Re: [Asterisk-Users] Static on inside end of conversation >>> >>> >>> >>> Jeff Busch wrote: >>> >>> >>>> You are correct. Inside means the IP Network and Outside means >>>> the PSTN accessible via the Audiocodes MP-108 gateway. >>>> >>>> No QoS. >>>> >>>> >>>> >>>> >>> We've seen echo on congested LANs within our Enterprise. I'm not sure if >>> this fits what your seeing or not. We've placed phones in their own vlan >>> and added 802.1p QoS to expedite forwarding upto the first hop router >>> for the outbound call leg and the echo is gone. We did the same for the >>> inbound call leg also. >>> >>> >>>> Thanks - Jeff >>>> >>>> -----Original Message----- >>>> From: asterisk-users-bounces@lists.digium.com >>>> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steve Blair >>>> Sent: Tuesday, November 29, 2005 4:20 PM >>>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>>> Subject: Re: [Asterisk-Users] Static on inside end of conversation >>>> >>>> >>>> >>>> Jeff Busch wrote: >>>> >>>> >>>> >>>> >>>>> Hello, >>>>> >>>>> I am running the following configuration: >>>>> >>>>> 2.8ghz P4 with 1GB of RAM >>>>> Audiocodes MP-108 connected to 5 POTS lines Polycom IP-500 phones >>>>> Asterisk@Home 1.3 (this is Asterisk 1.0.9) >>>>> >>>>> End users are complaining of an echo and static on the inside end (the >>>>> >>> >>> >>>>> internal side), but the outside end of the conversation doeesn't notice >>>>> >>>> >>>> >>>> >>>>> anything. >>>>> >>>>> >>>>> >>>>> >>>> I'm assuming inside means on the IP network and outside means on >>>> the PSTN which is accessible via a gateway is this correct? If so >>>> do you have QoS enabled on the inside? >>>> >>>> >>>> >>>> >>>>> Does anyone have any suggestions on troubleshooting / fixing this >>>>> problem? >>>>> >>>>> Thanks! >>>>> >>>>> Jeff >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> --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 >>> >>> _______________________________________________ >>> --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 >>> >> > > > >
Mojo with Horan & Company, LLC
2005-Nov-30 10:46 UTC
[Asterisk-Users] Static on inside end of conversation
I have a very similar server, pstn setup, phones, and user base, and I switched over to G729 codec 'cause the polycoms support it. While the call quality has dropped ever so slightly (I have received no complaints from my users however), snaps, crackles, clicks and pops are gone. I did not have as extreme a case of static it seems, though. This will take more processor power, and will probably make any lost interrupts more evident. Moj Jeff Busch wrote:> Hello, > > I am running the following configuration: > > 2.8ghz P4 with 1GB of RAM > Audiocodes MP-108 connected to 5 POTS lines > Polycom IP-500 phones > Asterisk@Home 1.3 (this is Asterisk 1.0.9) > > End users are complaining of an echo and static on the inside end (the > internal side), but the outside end of the conversation doeesn't notice > anything. > > Does anyone have any suggestions on troubleshooting / fixing this > problem? > > Thanks! > > Jeff > > > > _______________________________________________ > --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 >-- Mojo <mojo@horanappraisals.com> Office Manger, Horan & Company, LLC (907) 747-6666 x112
Update on this... And it is still not solved. This is actually fairly interesting. I have two installations at a construction company. They are both running similar class machines (I was wrong in my initial post) they are: System "A" 2.4 ghz Celeron 1 gb RAM IDE Drives Asterisk@Home 1.13 (Asterisk 1.0.9) An Audiocodes MP-108 to interface with PSTN 9 Polycom IP-500 Phone system is on 16 port linksys switch. All PC's are on separate 16 port switch. Both switches are plugged into our VPN firewall (Netscreen 5xp) System "B" 2.4 ghz Celeron 1 gb RAM IDE Drives Asterisk@Home 2.1 (asterisk 1.2) 5 Sipura SPA-3000 to interface with PSTN 12 Polycom IP-500 Phone and data on NEW Netgear 48 port Smart Switch. Voice is on Vlan 02 with QoS set to HIGH. Data is on Vlan 01 with QoS set to LOW. Switch is connected to Netscreen 5xp VPN firewall Polycoms are running the newest possible firmware for this phone: Bootrom 2.6.1 and SIP 1.5.2 Everything is working fine on both systems, but BOTH systems are experiencing echo on the IP side of the conversation. Some phones are worse than others. System A is experiencing echo both on internal Station to station calls, and on outbound calls. System B is experiencing echo on internal station to station calls, and no one has complained about echo on outbound calls yet. The Polycoms are running G.711u as their 1st configured codec (G711a as #2 and G.729AB as #3) and Asterisk is configured to disallow=all & allow=ulaw. I have searched and searched concerning echo and Polycom phones and haven't found anything that seems to be relevant. Any help would be appreciated. One more note, I am NOT running TDM4xxp cards on either of these machines, so Zaptel information concerning echo wouldn't be relevant, correct? Thanks! Jeff Busch -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mike McMullen Sent: Tuesday, November 29, 2005 6:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Static on inside end of conversation ----- Original Message ----- From: "Jeff Busch" <Jeff.Busch@lewisbuilds.com>> Hello, > > I am running the following configuration: > > 2.8ghz P4 with 1GB of RAM > Audiocodes MP-108 connected to 5 POTS lines Polycom IP-500 phones > Asterisk@Home 1.3 (this is Asterisk 1.0.9) > > End users are complaining of an echo and static on the inside end (the> internal side), but the outside end of the conversation doeesn't > notice anything. > > Does anyone have any suggestions on troubleshooting / fixing this > problem? >Hi Jeff, I recommend upgrading to Asterisk@home 2.0 which was just released. It uses Asterisk 1.2 and a 2.6.9 based kernel which handles i/o and interrupts much better. While the link below discusses issues with digium cards, in general the interrupts and IDE vs SATA drive discussions are of use no matter what. http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te40 5p_noise.html I have 2 Digium T400P cards connected to 8 POTS lines and 10 sipura spa-841 phones. I went through two PCs we had (2.9 GHZ celeron, 2.1HGZ Athlon XP) both with IDE drives. I finally declared war on echo and the Rice Krispies syndrome (Snap, Crackle, Pop) on the internal end of the conversations. I went and bought an ASUS P5LD2 motherboard with 1GB memory, 3.2GHz P4 with hyperthreading and a 2MB cache, and a SATA drive. I installed Asterisk@Home 2.0. After some minor magic to get the correct merlin gigabit ethernet driver for CentOS 4.2, everything came up perfect. This was my thanksgiving project and so far sound quality has been perfect. No echo and no rice krispies. I simulated network load on the system by copying multi-GBs of files through the net from another server with scp while I called out and back into the system on multiple lines. Even with the scp reporting 9.5MB/sec, the phone sound quality was fantastic. I then upped the ante by copying multi-gb files on the hard drive which when viewing stats with top (hyperthreading shows as 2 CPUs and you run the SMP kernel) both CPUs showed no idle time. IO- Wait state never greater than 10%. Phone calls were still perfect with no echo or noise. Out of the 10 vmails I left as part of the test, only one had 3 very faint pops in a 30 second message. They could have come from the POTS line for all I know. I ran extended phone conversations by calling the Asterisk system from our old phone system, picking up the extension on the called SIP phone and then playing Law & Order dvd episodes (lots of talking) and placing the handset near to the speaker and taking the old phone system handset and listening and talking back into it for 30 minutes at a time. Necessity is indeed a mother.... ;-) The calls were perfect. I'm amazed at how clear the dvd audio came through. I then reversed the process and played the dvd audio through the Asterisk system handsets while listening and talking back through the old phone handsets. After 30 minutes the quality was still excellent. Hope you find some of this ramble useful. Mike _______________________________________________ --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
Correct. The issue is that most of the echo is between internal stations. SIP -> SIP. The users with the system using the sipura's don't report any echo when calling outside the office or receiving a call. The users with the system using the audiocodes report an echo for the first 1 - 2 seconds of a conversation using the pots lines (this must be the echo cancellation process of the audiocodes) but then it clears up. Likewise, they are experiencing echo when talking back and forth between extensions. I also forgot to mention that the users of the audiocodes system report "static" on the line. Jeff Busch -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric "ManxPower" Wieling Sent: Wednesday, December 07, 2005 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Static on inside end of conversation The device that interfaces to the PSTN is the interface that must cancel echo. If I read your post correctly, that is the SAP-3000 and the Audiocodes boxes in your case. Jeff Busch wrote:> Update on this... And it is still not solved. > > This is actually fairly interesting. I have two installations at a > construction company. They are both running similar class machines (I> was wrong in my initial post) they are: > > System "A" > 2.4 ghz Celeron > 1 gb RAM > IDE Drives > Asterisk@Home 1.13 (Asterisk 1.0.9) > An Audiocodes MP-108 to interface with PSTN > 9 Polycom IP-500 > Phone system is on 16 port linksys switch. All PC's are on separate > 16 port switch. Both switches are plugged into our VPN firewall > (Netscreen > 5xp) > > > System "B" > 2.4 ghz Celeron > 1 gb RAM > IDE Drives > Asterisk@Home 2.1 (asterisk 1.2) > 5 Sipura SPA-3000 to interface with PSTN > 12 Polycom IP-500 > Phone and data on NEW Netgear 48 port Smart Switch. Voice is on Vlan > 02 with QoS set to HIGH. Data is on Vlan 01 with QoS set to LOW. > Switch is connected to Netscreen 5xp VPN firewall > > > Polycoms are running the newest possible firmware for this phone: > Bootrom 2.6.1 and SIP 1.5.2 > > > Everything is working fine on both systems, but BOTH systems are > experiencing echo on the IP side of the conversation. Some phones are> worse than others. > > System A is experiencing echo both on internal Station to station > calls, and on outbound calls. > System B is experiencing echo on internal station to station calls, > and no one has complained about echo on outbound calls yet. > > The Polycoms are running G.711u as their 1st configured codec (G711a > as > #2 and G.729AB as #3) and Asterisk is configured to disallow=all & > allow=ulaw. > > I have searched and searched concerning echo and Polycom phones and > haven't found anything that seems to be relevant. Any help would be > appreciated. > > One more note, I am NOT running TDM4xxp cards on either of these > machines, so Zaptel information concerning echo wouldn't be relevant, > correct? > > Thanks! > > > Jeff Busch > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mike > McMullen > Sent: Tuesday, November 29, 2005 6:26 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Static on inside end of conversation > > > ----- Original Message ----- > From: "Jeff Busch" <Jeff.Busch@lewisbuilds.com> > >> Hello, >> >> I am running the following configuration: >> >> 2.8ghz P4 with 1GB of RAM >> Audiocodes MP-108 connected to 5 POTS lines Polycom IP-500 phones >> Asterisk@Home 1.3 (this is Asterisk 1.0.9) >> >> End users are complaining of an echo and static on the inside end >> (the > >> internal side), but the outside end of the conversation doeesn't >> notice anything. >> >> Does anyone have any suggestions on troubleshooting / fixing this >> problem? >> > > Hi Jeff, > > I recommend upgrading to Asterisk@home 2.0 which was just released. > It uses Asterisk 1.2 and a 2.6.9 based kernel which handles i/o and > interrupts much better. > > While the link below discusses issues with digium cards, in general > the interrupts and IDE vs SATA drive discussions are of use no matterwhat.> > http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te > 40 > 5p_noise.html > > I have 2 Digium T400P cards connected to 8 POTS lines and 10 sipura > spa-841 phones. > > I went through two PCs we had (2.9 GHZ celeron, 2.1HGZ Athlon > XP) both with IDE drives. I finally declared war on echo and the Rice > Krispies syndrome (Snap, Crackle, Pop) on the internal end of the > conversations. > > I went and bought an ASUS P5LD2 motherboard with 1GB memory, 3.2GHz > P4 with hyperthreading and a 2MB cache, and a SATA drive. I installed > Asterisk@Home 2.0. After some minor magic to get the correct merlin > gigabit ethernet driver for CentOS 4.2, everything came up perfect. > > This was my thanksgiving project and so far sound quality has been > perfect. No echo and no rice krispies. > > I simulated network load on the system by copying multi-GBs of files > through the net from another server with scp while I called out and > back into the system on multiple lines. Even with the scp reporting > 9.5MB/sec, the phone sound quality was fantastic. > > I then upped the ante by copying multi-gb files on the hard drive > which when viewing stats with top (hyperthreading shows as 2 CPUs and > you run the SMP kernel) both CPUs showed no idle time. IO- Wait state > never greater than 10%. Phone calls were still perfect with no echo ornoise.> Out of the 10 vmails I left as part of the test, only one had > 3 very faint pops in a 30 second message. They could have come from > the POTS line for all I know. > > I ran extended phone conversations by calling the Asterisk system from> our old phone system, picking up the extension on the called SIP phone> and then playing Law & Order dvd episodes (lots of talking) and > placing the handset near to the speaker and taking the old phone > system handset and listening and talking back into it for 30 minutesat a time.> Necessity is indeed a mother.... ;-) > > The calls were perfect. I'm amazed at how clear the dvd audio came > through. > > I then reversed the process and played the dvd audio through the > Asterisk system handsets while listening and talking back through the > old phone handsets. > > After 30 minutes the quality was still excellent. > > > Hope you find some of this ramble useful. > > Mike > > > _______________________________________________ > --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