[Asterisk-Users] Problems registering Linksys SPA941 with * via SIP
Hi all, Having trouble registering Linksys SPA941 with our * box. We can't find an entry in the config screens to allow us to put the IP address of the SIP server (i.e. the * box) in. We can find an entry for a SIP proxy in the phones set up, but we're not using one (SIP connections are direct across the LAN). Phone using static IP and no DNS present on network. Appreciate any help. Cheers, Damian. ___ --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] 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 [EMAIL PROTECTED]: 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http
Re: [Asterisk-Users] ip phones
Polycom make some surprisingly good and reasonably priced SIP handsets too. Many of these support headsets and we've been quite impressed by them. 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 Stephen Arulraj [EMAIL PROTECTED]: Hi John, I know a good one and it's from Siemens. The new optiPoint 410 or 420 series are great phones with good quality speech and features. Contact me off list for pricing etc.. With regards, Stephen John Fraser wrote: Hi all, Does anybody have any info on a decent quality sip hard phone that is headset compatible? Thank you John ___ --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
[Asterisk-Users] Remote call pick-up
Hi, Does anyone have remote call pick-up working on * (either via SIP or otherwise)? If so then can you post your features.conf, sip.conf and/or zapata.conf? We can't seem to get this (seemingly simple) function to work. Cheers, Damian. ___ --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] Remote call pick-up
Hi, Does anyone have remote call pick-up working on * (either via SIP or otherwise)? If so then can you post your features.conf, sip.conf and/or zapata.conf? We can't seem to get this (seemingly simple) function to work. Cheers, Damian. -- 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 ___ --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] Variable in call parking
Hi Andrew, Not sure if I understand your question, but this may help - * has the following settings in features.conf that are related to parking: parkext = ;the extension that users xfer calls to in order to park them parkpos = - ;the extension range that * will use to park calls. It will tell the user which extension it has parked each call on when they are parked. Dialling this extension retrieves the parkpos. context = parkedcalls ;context that calls are parked in parkingtime = xxx ;number of seconds a call will be parked for before being sent back to the extension that parked it. Hope this helps, Damian. 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 Andrew Nowrot wrote: Hi, Can anyone tell me if Asterisk sets some variable when doing a call parking (when someone presses an exten set in features.conf). In can't find this information on a wiki. Cheers ;) Andrew ___ --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] zttest - 100% ?
Have you checked that the TDM400P isn't sharing an IRQ with anything else? Don't trust /proc/interrupts - run lspci -v to confirm this. We have * running on an x206 and found that the only way to stop the TDP400P sharing an IRQ with other devices was to juggle cards between slots. Hope this helps! Damian. 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 Marco Supino wrote: Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in the zttest testings, Thanks for any info. Marco. ___ --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] Remote call pickup
Hi, Has anyone got sample sip.conf, features.conf and zapata.conf files that they can send me that demonstrate a working remote pick-up config? We can't seem to get it working at one of our sites - have changed the remote pick-up extension so it doesn't conflict with the SIP phones redial function but to no avail. Appreciate any help. Cheers, Damian. -- 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 ___ --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] zttest
Hi Waldo, I would be money on your problem being related to the accuracy of zttest. One way of checking IRQ's is to run cat /proc/interrupts, but it is a lot more accurate to run lspci -v and lspci -vb. I would recommend Googling the lspci command, although the output is pretty self explanatory. The TDM appears as a TigerJet card, not sure what TE410P will list as. PCI devices have their IRQ's dictated by the BIOS of the host system. How (and if) you can configure these manually depends on the type of BIOS you have... in our IBM xSeries 206 we had to actually juggle cards between slots to get it to assign a unique IRQ to the TDM400P. Good luck! D. 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 Waldo Rubinstein wrote: Damian, Thanks for your input. Hyperthreading is in fact enabled and now that you mention this I will disable it. The reason I ask is because under some load (may be 40 simultaneous calls), voice quality degrades. We have audio problems where one party hears the other but not viceversa and then it all works fine. It's random audio quality problems in general. During these cases, I'm constantly running vmstat 1 and CPU utilization is always 85%+ idle. I will also look into setting the TE410P in its own IRQ. Do you know how I can do that? Is that a motherboard BIOS setting or is it something that needs to be done to the TE410P itself? Thanks, Waldo On May 16, 2005, at 12:59 AM, Damian Funnell wrote: Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). Do you have issues with your * box? If so then I would start worrying about zttest output (and thinking about disabling hyper threading on those dual Xeons), otherwise have a smile and a beer and pity us poor fools who have had problems due to poor results. Cheers, Damian. 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 Waldo Rubinstein wrote: I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% --- Results after 57 passes --- Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793 What does this mean? Should I have expected to get 100% across the board? This is from a TE410P running on Debian 2.6.11-1-686-smp on a dual Xeon 2.4GHz server. Thanks, Waldo ___ 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 ___ 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
Re: [Asterisk-Users] zttest
Hi Rich, This is always a BIOS setting - there is no O/S command to disable H/T. To date I have never heard of a BIOS that does not allow the user to disable H/T, but I have read that there are BIOS'es out there that don't offer this function. Go into your BIOS setup screen and you should find the option somewhere. D. 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 Rich Adamson wrote: Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). How do you disable hyper threading (what's the command and where is it placed)? ___ 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
Re: [Asterisk-Users] zttest
...Jens makes a liar out of me, although I read that the 'noht' switch stops the OS from using H/T but doesn't disable it completely. I make no warranties regarding the accuracy of this information, though. D. 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 Jens Vagelpohl wrote: On May 16, 2005, at 14:37, Rich Adamson wrote: Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). How do you disable hyper threading (what's the command and where is it placed)? If this is a Linux box, look at the kernel boot arguments in [lilo| grub].conf and append noht, that disables it. My grub.conf on one of my boxes looks like this: title CentOS (2.4.21-27.0.4.ELsmp) root (hd0,0) kernel /vmlinuz-2.4.21-27.0.4.ELsmp ro root=LABEL=/ noht initrd /initrd-2.4.21-27.0.4.ELsmp.img jens ___ 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
Re: [Asterisk-Users] zttest
Rich, did you check IRQ's? Our zttest results didn't improve markedly when we did either change (IRQ's or H/T), but the problem went away regardless. Other than that Digium have recommended throwing out our SCSI320 RAID hardware and replacing it with IDE (i.e. not SATA) kit, although thankfully we haven't had to make this retarded change (yet). I would also recommend trying to disable H/T in the BIOS (rather than via software) as I wonder if H/T still runs on your box (but is not accessed by the OS), so may still be causing you grief. D. 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 Rich Adamson wrote: On May 16, 2005, at 14:37, Rich Adamson wrote: Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). How do you disable hyper threading (what's the command and where is it placed)? If this is a Linux box, look at the kernel boot arguments in [lilo| grub].conf and append noht, that disables it. My grub.conf on one of my boxes looks like this: title CentOS (2.4.21-27.0.4.ELsmp) root (hd0,0) kernel /vmlinuz-2.4.21-27.0.4.ELsmp ro root=LABEL=/ noht initrd /initrd-2.4.21-27.0.4.ELsmp.img Thanks, I added the noht, rebooted, and still get zttest results that consistently at 99.987793%, both before and after the change. Guess hyper threading has nothing to do with it on this particular system. This is a new motherboard and it doesn't indicate anything in the bios relative to hyper threads either. Rich ___ 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
Re: [Asterisk-Users] Static on TDM Zaptel FXO
Hi Gregory, Have you checked that the card is on its own IRQ? There has been a bit of discussion about this type of thing recently on the list, do a search on the archive to find the various threads. Try running zttest and see what accuracy it is reporting - anything less than 99.99% is supposedly bad. Various 'fixes' (work-arounds) are to make sure the zaptel is on it's own IRQ, disable Hyperthreading and remove RAID controllers or any other high-performance devices that support DMA and other things that are likely to compete with the zaptel. Good luck. D. 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 Gregory Wiktor - ADCom Corp. wrote: Hello All, I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy static. Even with the pots line disconnected, if I do a dial I still get static. This way I know it's not the line, but rather something on the card. I tried alternate pci slots. This card has a power connector, does anyone know what the power requirements are? The unit is in a small case with a 2.4ghz p-4 and 512mb ram, on an intel board with 533fsb. All other functions are fine. I am using the latest CVS on Debian 2.6test Anyone experience this? Regards, Greg ___ 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
Re: [Asterisk-Users] zttest
Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). Do you have issues with your * box? If so then I would start worrying about zttest output (and thinking about disabling hyper threading on those dual Xeons), otherwise have a smile and a beer and pity us poor fools who have had problems due to poor results. Cheers, Damian. 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 Waldo Rubinstein wrote: I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% --- Results after 57 passes --- Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793 What does this mean? Should I have expected to get 100% across the board? This is from a TE410P running on Debian 2.6.11-1-686-smp on a dual Xeon 2.4GHz server. Thanks, Waldo ___ 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
[Asterisk-Users] Something every TDMP user should know
Hi team, Not long ago a bunch of us were posting reports of a strange phenomenon where voice quality would pack up completely from time to time, typically resulting in loud crackling on the line and/or the voice channel breaking up completely. With our installation it would occur from time to time, typically when the * server was at it's busiest. Most of the time this problem would result in all users having to terminate their calls and re-establish them. After a lot of (very frustrating) troubleshooting we have have now gone two weeks without a re-occurrence of the problem and we are hoping that we may have finally resolved it altogether. I wanted to post a quick summary of the steps that we have taken to resolve this issue and what we think the problem turned out to be, as (from the number of responses to my last posts about this issue), it sounds like a few people have been experiencing it, so hopefully our experiences will help. The * server in question is based on a single-processor IBM xSeries 205 with a gig of RAM, SCSI 320 HDD's (RAID 1) and Red Hat ES 3. It uses ISDN (via CAPI and a four port Eicon Diva Pro Server card) and a mixture of SIP and analogue extensions. A TDM400P with four FXS ports supports the four analogue extensions (all Uniden cordless phones) and the SIP handsets consist of a mixture of BT102's and SNOM190's. Our turning point with this issue came when we bit the bullet and purchased a support incident from Digium. By this stage we had spent dozens and dozens of hours trying unsuccessfully to research and diagnose the problem and still had no accurate idea of what was causing it. Several people replied to our posts to this list saying that they were having a very similar issue as well, but no one had a clue what was causing it. Digium support zeroed in on the issue fairly quickly and we got the *distinct* impression that they have seen this problem many times before. They instantly got us to look at the output of zttest and we found that this was (in their words) 'extremely low', with 'best' and 'worst' readings of 99.975586% and 99.963379% respectively. They told us that we needed to be getting at least 99.98% and recommended that we: Check that the TDMP is on it's own IRQ (much to our embarrassment our card wasn't at the time, so we had to play with it a bit to get it to occupy a unique IRQ). Disable hyper threading on the Xeon CPU. Uninstall our SCSI hardware and replace it with IDE hardware. Upgrade to the latest stable releases of Asterisk, Zaptel and Libpri. We made changes 1 and 2 in the above list and are prepared to make changes 3 and 4 if we find the problem hasn't gone away. It hasn't happened in over two weeks now (after occuring many times per day for a while), so we hopefully won't have to throw out our SCSI hardware. After we made each change (1 and 2 were made about two weeks apart from each other) we found that the quality improved, with the incidence of the issue halving after '1' and disappearing (hopefully for good) after '2'. Incidentally the results of zttest *did not* noticeably improve after making these changes (it is still below 99.98%). Apparently our problem is related to the fact that the TDMP generates massive amounts of IRQ requests and that it becomes extremely upset if a suitable number of those IRQ requests are not honoured. Dispite the fact that a PCI device has to be able to share an IRQ in order to meet the PCI specification, it appears that having a TDMP sharing an IRQ with *anything* is a really really bad idea. I haven't been able to get an explanation about why hyper threading is a bad thing, but apparently high-performance devices such as SCSI adapters can cause resource contention issues with the TDMP, resource issues that the TDMP becomes very upset about. So hopefully we have seen the back of this problem and I have to say that I have been pretty dissappointed to find out that this issue appears to be relatively well known by Digium, but seemingly not publicised in the slightest. We searched for days to find anything relating to our issue but to no avail. Hopefully the next time someone has this issue they might find this mail and save themselves some of the frustration that we had. When we challenged Digiums advice about retarding the CPU (i.e. disabling hyper threading) and slowing I/O (by throwing out our SCSI RAID controller and replacing with IDE) they fell strangely silent - after getting prompt and meaningful responses to our requests they suddenly stopped responding at all. I think that this issue constitutes a pretty major flaw in the design of the TDMP and we will strongly avoid putting these cards into any * servers from now on. This is a real shame, as we as a company really want to reward Digium for all of their good work by actually buying their products, but we no longer have any faith in the design and suitability for production use of this product. Maybe it's time for Digium to think about
Re: [Asterisk-Users] Best CPU config for dual-Xeon?
Hi Tony, check out my recent post regarding our experiences with Hyperthreading and * with Zaptel cards. We have a few machines in the wild that *do* run Hyperthreading but no Zaptel cards and these work absolutely fine. My understanding is that the Hyperthreading problems are purely related to HW interrupts with Zaptel. My advice would be to leave HT turned on and just turn it off if you have problems with it - something that takes are few seconds in the BIOS and doesn't require any software changes. HT does provide significant performance improvements over non-HT... performance that could come in handy if your * server has a lot of calls in progress (and hence a lot of CODEC's to process). Cheers, Damian. 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 Tony Mountifield wrote: I have some beefy dual-Xeon servers that I will be using for Asterisk VoIP applications (i.e. no Zaptel cards). Using 2.6.11-1.14_FC3smp as the kernel (Fedora Core 3), and currently with Asterisk STABLE. My question is concerning the CPU setup, as I've seen conflicting or out-of-date suggestions: given the above config, should I have hyper-threading turned on or off? Turned on appears like 4 CPUs, and turned off will, I assume, appear as 2 CPUs. It's not clear to me what the issues with HT are/were, and whether they only relate to the use of hardware, interrupts, etc. or what, so any advice would be much appreciated. Cheers Tony ___ 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] SNOM190 DTMF problem
Hi all, We've got a problem where a bunch of SNOM 190 phones that we have just installed are giving us problems with DTMF tones. Users of all phones reported that when they access voicemail the VM app is not recognising DTMF tones. One clever user figured out that they DO work if you hold the key down for a certain amount of time, but getting it right is very difficult. Have a feeling that there is something that we have misconfigured, but can't figure it out. Any help appreciated. Cheers, Damian. 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 ___ 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] [Fwd: Call forwarding]
Any takers? Sometimes the most basic questions yield the least replies, huh? Cheers, Damian. Original Message Subject: Call forwarding Date: Wed, 04 May 2005 08:40:41 +1200 From: Damian Funnell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com References: [EMAIL PROTECTED] [EMAIL PROTECTED] Hi team, Basic question I know, but I can't seem to find any obvious information about this: Does anyone know if * natively supports call forwarding from a given extension (i.e. call forwarding without having to write a macro)? My user wants to be able to dial a code plus a phone number to start diverting all calls to the given extension to that number. Call forwarding would then be disabled by dialling a code number again. I expected that * would support this type of feature natively, but can't find anything in the wiki. If responding please let me know if we need to enable anything in features.conf as well. Thanks in advance, Damian. 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 -- 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 ___ 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] Call forwarding
Hi team, Basic question I know, but I can't seem to find any obvious information about this: Does anyone know if * natively supports call forwarding from a given extension (i.e. call forwarding without having to write a macro)? My user wants to be able to dial a code plus a phone number to start diverting all calls to the given extension to that number. Call forwarding would then be disabled by dialling a code number again. I expected that * would support this type of feature natively, but can't find anything in the wiki. If responding please let me know if we need to enable anything in features.conf as well. Thanks in advance, Damian. 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 ___ 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] Delete voicemail
Hi all, Does anyone know what the easiest way is to delete voicemail for one extension? Had a search online but couldn't find anything. Cheers, Damian. ___ 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] US$200 bounty for * paging feature
I agree - you guys really shouldn't be wasting our bandwidth unless it's important. What were you thinking? trixter http://www.0xdecafbad.com wrote: On Wed, 2005-04-20 at 22:14 -0500, Dan Perik wrote: Michael D Schelin wrote: Ok you guys enough. The debate will go on forever. Agreed! At the risk of wasting bandwidth myself Please, guys stop wasting my precious bandwidth. If you want to private message your flames, great but leave this list to Asterisk, please. Thanks! - Dan Interesting that so many people are coming out to say stop, even to reply to others saying stop and holding precious bandwidth up as the reason. I love your logic. To jump on the bandwagon stop waasting my bandwidth telling people to stop wasting your bandwidth. Its only fair. ___ 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
Re: [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???
Hi Tim, Thanks for your post, it's most insightful. It certainly puts a pretty large dent in my confidence in the TDM for commercial use - imagine if there was more than one TDM in a system (especially with a RAID adapter). Running a PABX without hardware RAID 0 is not an option for us, as we don't want disk failure to result in the PABX dying, so I guess we are going to have to research ways of retarding it somehow. Cheers, D. 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 [EMAIL PROTECTED] wrote: Yes. It has to do with latency and bus contention. I've run a TDM board in an IBM Netfinity 5600 server with an IBM ServeRAID 3L controller (SCSI-U2W). The big difference, though, is that the RAID controller was on its own PCI bus, and the TDM card was on its own PCI bus. With both controllers on the bus, you can have latency issues. For example, if the RAID controller sets up a DMA of a big chunk of disk, it owns the bus for that transfer. If an Ethernet packet is delayed by 50us during that time, nobody cares. But if the TDM card is delayed, it most certainly cares: especially as its generating 1000 interrupts a second! That's the problem with the TDM cards. They do *nothing* on the CPU side. The CPU has to do *everything*, and it has to do it *immediately*. When you are using plain-jane IDE, you can tweak the kernel to put the IDE stuff at a low priority. But when you've got a fancy RAID controller, it tends to think it's the most important thing in the system. And as a rule, hard drive I/O usually *is* the most important I/O going on in a system. However, in this case, the TDM card trumps that. And Digium doesn't know how to tweak every last RAID driver in existence for low-priority operation--or even if it's possible. Hence, the recommendation for IDE. Yet they require PCI 2.2, which eliminates most Pentium III's and lower! :) I'm still in the midst of testing the TDM cards. So far, so good, in an EPIA-based solution and in the 5600. But I've been through at least half a dozen different systems before I've found these... Tim Massey ___ 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
[Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???
Hi everyone, For anyone who was following the earlier thread about noise problems on *, here's a little gem from Digium support. In a nutshell Digium told us that they thought that low accuracy results from zttest were the cause of noise problems that we have been experiencing. We re-configured the box to give the TDM400P its own IRQ, but found that the zttest results were still lower than what Digium recommended they should be. When I asked them for further information on how to improve this they replied: ** Extract begins ** SCSI RAID can cause the problem. If disabling hyper threading does not resolve your problem my next suggest would be to revert to a PATA IDE hard drive solution configured to UDMA level 2 using hdparm. SCSI or SATA causes problems on some systems from what I have seen. The problem increases when using a SCSI or SATA RAID. ** Extract ends ** I really hope that they are wrong, as I don't feel like throwing away my nice expensive Ultra320 SCSI RAID controller and hot plug drives and replacing them with some crusty old IDE config. Needless to say I'm not going to go and shell out on IDE controller drives until I'm a little more certain that this is actually a problem and have asked them for more information. Does anyone else find it odd that the TDM could possibly have a problem sharing a box (but not an IRQ) with a SCSI controller? Combined with the fact that they have also recommended that we turn off hyper threading (also causes problems with TDM, apparently), I'm wondering if these cards shouldn't come with a warning not to use anything with half decent performance in your * server! ___ 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] extension dialing resistivity
Hi Joseph, Let me take a guess - the problem only occurs when dialling four digit extensions? I think you will find that your dial plan is matching the three digit extension and then dialling it straight away - Asterisk won't wait for a timeout before trying to follow the dial plan, as soon as it finds a match it will try and dial whatever you've told it to (whether an extension context exists or not). This means, for example, that if you dialed extension '1234' then Asterisk will try and dial '123' if it finds a matching pattern in the dial plan - even if the extension '123' is invalid. There are two ways around this - either re-configure your dial plan so Asterisk won't get confused between three digit and four digit extensions (starting them in different numbers is a good idea) or configure your SIP phones (assuming you are using SIP phones) not to use forward dialling (i.e. to dial after a pre-set delay. We usually do the latter, as most SIP phones allow you to use the hash key to tell the phone to 'hurry up and dial now'. If you want to get really funky you can also write your dial plan so that it waits for 'n' seconds between each digit, but who could be bothered? 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 Joseph wrote: Which file control extension dialing responsivity / timing? When someone dial my extension, and is not fast enough, asterisk announces that the extension is not valid (it happened to me too). I have a mixed of two and three digit extensions in dial plan. Which setting controls this behavior. - ___ 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] Changing IRQ's on TDM
Hi Bryan, thanks, but we found a combination of slots that resulted in the TDM (and our BRI ISDN card) getting a unique IP address. IBM has a lot to answer for with their xSeries 206 server - looks like they've given the BIOS a lobotomy or something, as it doesn't allow you to configure much. Bryan Boatright wrote: Is the APIC and IO-APIC enabled? Send us 'cat /proc/interrupts' and your /var/log/boot.msg (or your distro's equivalent bootup log). Damian Funnell wrote: Hi all, I've found that a TDM400P card in our * box is sharing IRQ's with two other devices. The server doesn't support assigning IRQ's through the BIOS and the pig only has three PCI slots, so swapping cards between slots hasn't fixed the problem (it just ends up sharing IRQ's with other devices). Any ideas on how we can force the TDM to use a certain IRQ? Plenty of free IRQ's in the box, BIOS just doesn't want to use them. 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 ___ 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 ___ 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] Line Noise HELP!
Hi, For those that were having the same line noise problem that we were, an update: * Our TDM400P *was* sharing an IRQ, despite the output from 'cat /proc/interrupts' showing that it wasn't. Running 'lspci -v' showed that it was and we had to perform some card juggling to get it (and our ISDN card) to sit on different IRQ's. * It appears that this change has resulted in a better *average* accuracy via zttest (although as zttest does not provide an 'average' figure, this is anecdotal only). Certainly the 'best' and 'worst' output is no different (best = 99.975586%, worst = 99.963379%). * The users reported a significant improvement on the last full day of use - they experienced a couple of 'spikes' of noise, but no incidences of the prolonged noise that they were experiencing earlier. This was only over one day, though, so not sure if this means the problems licked or if Friday was just a good day. * I've asked Digium for further advice, as they have told me that we should try and get no less than 99.98% accuracy via zttest. * They've recommended disabling hyper threading and we are yet to give this a go. Will be pretty annoyed if we have to leave hyper threading turned off in order to get this to work, as performance is sure to suffer. Keen to hear if anyone else has managed to get anywhere with their noise problem. Not sure if we're on the right track or not, but will report back in a couple of days to see if the intermittent crackling has returned. Rich Adamson wrote: Excerpt of email from Digium support: Digium does not support CAPI or BRI, either freely or commercially. We can only provide support for you Digium TDM400P card. Your zttest output is extemely low. We are looking for nothing less than 99.98%. If you are receiving 99.96% this will cause major problems. Have a best output of 99.975% is really low. For the record, I ran zttest against a new TDM04b Rev H board and it consistently reported 99.975586%. There is no noise, crackling, etc. So, not sure what 99.975% is really low is based on. 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. Also, did the above, which had zero impact on the zttest results. If I try to use spandsp-pre11 for fax reception, it results in far less then usable output (*.tiff), and supposedly that is due to missed frames, missed interrupts, or something like that with the TDM04b card. Checked and double-checked share interrupts, and that isn't a problem. This is on a cvs-head RHv9 box with 2.2ghz processor, and nothing else running on the system. So, best guess is that 99.975586% is impacting fax but not voice. ___ 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
Re: [Asterisk-Users] Ring two extensions at the same time
The following dial string dials both extensions - this has worked for SIP and analogue extensions on our Asterisk machines. Both extensions ring until one is answered: exten = s,3,Dial(SIP/9295SIP/9287,,t,) 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 G.Marshall wrote: Hello, I can not find anything on this, so it may not be possible. I would like to dial one number which then rings at least two extensions at the same time. Not a hunt group, but ringing at the same time as if they were plugged into the same physical port. Does anyone know if this can be done, and if so how? Many thanks, Spencer ___ 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
Re: [Asterisk-Users] distribute outbound calls
Rotate or make sure the line you are dialling out on isn't in use before you try and use it? Lookup setgroup/checkgroup on the Wiki if it's the latter - http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup Allows you to create logical groups for just about anything, check whether the groups are full perform conditional steps, etc. Can post sample syntax if you're interested. D. 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 jltaylor wrote: Any ideas on how to rotate (evenly distribute) outbound calls over a number of 'trunks' or contexts? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 ___ 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
[Asterisk-Users] Has anyone had problems with Digium TDM400P and hyperthreading?
Digium have told us that a problem that we are having (with accuracy of zap interface as measured using zttest) may be due to the fact that we have a Xeon processor with hyperthreading and have suggested turning H/T off. Anyone else experienced a problem like this? No too keen about turning H/T off, as we're running the SMP RH kernel and don't really feel like replacing the kernel (and other kernel-specific bits) on the off chance that H/T is actually the problem. Thanks, Damian. 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 ___ 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] Busy line status and chan_capi?
Hi Kib, What exactly is it that you want to do? If you have a direct dial-in (DDI) number that goes to a certain extension then you can handle this pretty easily in your dial plan - check out this snippet below from one of our customers' machines. This example is pretty basic, but it works fine for their requirements: exten = 290,1,Answer exten = 290,2,Dial(SIP/9290,,t,) exten = 290,3,Voicemail(9290) exten = 290,4,Congestion The extension (290 in this example) is the DDI (known as a DID) number that the telco presents on the line (this customer has 10 DDI's) and that CAPI presents when the line rings. This is used to decide which extension to route the call to. By default SIP will use call waiting, which this particular customer doesn't like, so they dial *71 from each extension to cancel it. 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 Kib Eki wrote: What do i have to confiure so that a call comming in the * server through chan_capi recognizes a normal busy line beep if the SIP phone is busy? Kib ___ 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
Re: [Asterisk-Users] ISDN Fritz and TDM400
Hi Robson, We are using a 4 port Eicon Diva Pro BRI ISDN with a TDM400P with four FXO ports. We are using CAPI with the TDM400P and everything works fine most of the time. We are having periodic problems where the call quality completely falls apart for all in-progress calls and we have yet to diagnose this and resolve it, but at present we suspect that it may be a timing issue related to the TDM sharing and interrupt or something like that. We have found CAPI great so far, although there was a complete lack of help (through this list or through the IRC channel) when we were trying to set it up, seemingly because not many people use it. Let me know if you want copies of config files, etc. D. 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 Robson Ribeiro wrote: Is it possible to have on the same machine an ISDN Fritz Card and a TDM400 with two FXO ports? If so, is there any place I can find instructions to configure it? ___ 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
[Asterisk-Users] Changing IRQ's on TDM
Hi all, I've found that a TDM400P card in our * box is sharing IRQ's with two other devices. The server doesn't support assigning IRQ's through the BIOS and the pig only has three PCI slots, so swapping cards between slots hasn't fixed the problem (it just ends up sharing IRQ's with other devices). Any ideas on how we can force the TDM to use a certain IRQ? Plenty of free IRQ's in the box, BIOS just doesn't want to use them. 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 ___ 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] Line Noise HELP!
Hi Rich, Hear your point about the trace and all, will try and figure something out. Will also look at logging debug messages. We did the unthinkable and purchased a support incident through Digium and they have zeroed in on the zttest output, as per the info below (I've pasted in an excerpt from their email in case anyone else finds it useful). Not sure if this is the cause of our problem or not, as I don't understand whether the TDM card is used as a timing source for calls over CAPI, but will look at getting zttest output up regardless. Have to say that I am pretty impressed with Digium support so far - the engineer even rang me in New Zealand to follow up on their email and to inform me that I still have 45 minutes of my hour left to use. Cheers, Damian. Excerpt of email from Digium support: Digium does not support CAPI or BRI, either freely or commercially. We can only provide support for you Digium TDM400P card. Your zttest output is extemely low. We are looking for nothing less than 99.98%. If you are receiving 99.96% this will cause major problems. Have a best output of 99.975% is really low. Please ensure that you are running either Asterisk Release 1.07, Zaptel Release 1.07, and Libpri Release 1.07 (Libpri only required if using a PRI) from www.asterisk.org or Asterisk CVS Stable/HEAD, Zaptel CVS Stable/HEAD, and Libpri CVS Stable/HEAD (Libpri only required if using a PRI) as of the current date from Digium's CVS server. You may obtain instructions on downloading CVS Stable/HEAD from Digium's CVS server by visiting the download area on www.asterisk.org. Please verify that your Digium hardware is not sharing an IRQ on your system. You can accomplish this by running cat /proc/interrupts. Do not solely rely on cat /proc/interrupts to determine whether your Digium hardware is sharing an IRQ on your system. Make sure your Digium hardware is on its own IRQ by itself and that it is taking interrupts. You can determine whether it is taking interrupts from the 2nd column of output from cat /proc/interrupts. This should be something other than zero. You will also need to verify that your Digium hardware is not sharing an IRQ by examining the output after runninglspci -v and lspci -vb. Using lspci is the best way to determine whether or not your Digium hardware is sharing an IRQ on your system. Please verify that all Digium hardware is on its very own IRQ by itself. You may need to disable unnecessary hardware on your machine such as sound controllers, USB controllers, extra ethernet controllers, firewire, parallel ports, and/or serial ports. You should try moving and swapping our card to different PCI slots in order to get it on it's own IRQ. Some BIOS's will allow you to specify an IRQ for each PCI slot and/or onboard devices. 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. You will need to add the hdparm command you executed to one of your distribution's startup scripts. This way the IDE hard drive settings will be updated on each reboot. You can check whether or not your Digium hardware on your system is experiencing IRQ misses by using the zttest application which should be located in yourzaptel source directory. Do not solely rely on zttest to determine whether you are having IRQ misses with your Digium hardware on your system. Optimally,we are looking for output of 100% from zttest. Our cards will function properly as long as they do not report back less than 99.98%. Some people have reported no apparent problems with output as low as 99.975%, while others will have many apparent problems with an output as low 99.975%. You are almost guaranteed to have many apparent problems with an output lower than 99.975%. We strongly suggest doing everything possible in order to obtain atleast 99.98% output from zttest. I would watch the output over a 5 minutes period to check for spikes on intervals. You may also look for IRQ misses using the zttool application. Do not solely rely on zttool to determine whether you are having IRQ misses with your Digium hardware on your system. This application should be built while compiling zaptel. zttool requires the libnewt development package to be installed on your system in order to compile properly. IRQ misses with your Digium hardware can be due to I/O problems on your system. You may test if you are having I/O problems on your system by running hdparm -t /dev/[Hard Drive Device]. This will causes massive amounts of I/O on your system. The symptoms of an I/O problem on your system could be cracklingand/or static
Re: [Asterisk-Users] Running asterisk without special hardware
Hi Manish, Sure can, although you will need a timing source. If you don't plan to have any Digium hardware then you can use ztdummy (see http://www.voip-info.org/wiki-Asterisk+timer+ztdummy), which we have never used personally, but have yet to hear bad things about. D. 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 Manish Sapariya wrote: Hi List, Is it possible to setup asterisk in LAN only environment. I want to setup asterisk so that we use softphones for voice communication over LAN. I dont have access to any other special hardware and at this moment I dont think I need that too. Any pointer will be of great help. Thanks, Manish ___ 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
Re: [Asterisk-Users] Setgroup Checkgroup
Hi Ronald,
We use SetGroup/CheckGroup and your syntax appears to be fine (either
that or ours is broken too, but it seems to work ok!)
One question - what lines do you have in priority 1 - 3? We found that
our dial plan would not work unless the first priority (for all
extensions) was 1 and unless the priorities increased an integer at a
time (not sure if this is by design or not, but was the only way it
would work regardless).
Also (and this is really grasping at straws), have you tried using
different group names in case 'sip-1x' has any special meaning? These
names can be anything that is meaningful and we use the format 'line28x'
for our group names (which works fine).
Good luck!
Ronald Wiplinger wrote:
I have some troubles to use Setgroup / Checkgroup!!!
I setup a test (NoOP's are deleted): First caller should get first
line, second caller should get second line, third caller should get
busy and send an email. Note, that I used twice here to check the
first line!!!
[trunkint_A]
exten = _90N.,104,SetGroup(sip-13); increase Group counter
exten = _90N.,105,CheckGroup(1); check no more than 1
in this group
exten = _90N.,106,NoOp(Line 106)
exten = _90N.,107,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _90N.,108,hangup
;
exten = _90N.,206,SetGroup(sip-12)
exten = _90N.,207,CheckGroup(1)
exten = _90N.,208,NoOp(Line 208)
exten = _90N.,209,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _90N.,210,hangup
;
exten = _90N.,308,SetGroup(sip-13)
exten = _90N.,309,CheckGroup(1)
exten = _90N.,310,NoOp(Line 310)
exten = _90N.,311,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _90N.,312,hangup
;
exten = _90N.,410,Busy
exten = _90N.,411,SYSTEM(mail -s 'VPBX all lines in use'
[EMAIL PROTECTED])
I thought that 104 will set the Group counter sip-13 to 1 and will
use line 107 for the dial command
If another caller comes in that way, sip-13 would be 2 and because
Checkgroup allows only 1, the Group coutner would be setback to 1
and it will follow the jump to 206 and sets the Group counter sip-12
to 1
A third call should now find Group counter sip-12 and sip-13 set to
1 and give a busy signal and send an email.
HOWEVER, the log file show:
-- Executing SetGroup(Local/[EMAIL PROTECTED],2,
sip-13) in new stack
-- Executing CheckGroup(Local/[EMAIL PROTECTED],2, 1)
in new stack
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Line 106)
in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
so far so good!
-- Executing SetGroup(SIP/615-92c3, sip-13) in new stack
-- Executing CheckGroup(SIP/615-92c3, 1) in new stack
-- Executing NoOp(SIP/615-92c3, Line 106) in new stack
-- Executing Dial(SIP/615-92c3, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Ahh, it does not check Group counter sip-13, ... it checks
SIP/615-92c3 and Local/[EMAIL PROTECTED],2
How can I make it that it checks exactly the Group countersip-13
bye
Ronald
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Re: [Asterisk-Users] Setgroup Checkgroup
I would try setting the group name to something shorter and less likely
to conflict with any reserved system names (try something arbitrary like
'group1', 'group2', etc.
Our experience with SetGroup certainly doesn't indicate that using two
or more setgroups results in previous ones reverting to zero - we have a
total of four SetGroups and these are set and checked extensively
through our dial plan and they work fine.
Ronald Wiplinger wrote:
Damian Funnell wrote:
Hi Ronald,
We use SetGroup/CheckGroup and your syntax appears to be fine (either
that or ours is broken too, but it seems to work ok!)
One question - what lines do you have in priority 1 - 3? We found
that our dial plan would not work unless the first priority (for all
extensions) was 1 and unless the priorities increased an integer at a
time (not sure if this is by design or not, but was the only way it
would work regardless).
It starts correct with 1 (there are some NoOp and ENUM lookup, which
is not interesting for that case)
Also (and this is really grasping at straws), have you tried using
different group names in case 'sip-1x' has any special meaning?
These names can be anything that is meaningful and we use the format
'line28x' for our group names (which works fine).
The real group name is sip- (my phone number) I have
shorten it at the example.
I read somewhere, if you have TWO or more Setgroup than all previous
ones will be set to zero back!!! If that is the case than it makes not
much sense to use it that way I do. As you can see in the CLI outputs,
I copied, it uses a different name of the group variable.
bye
Ronald
Good luck!
Ronald Wiplinger wrote:
I have some troubles to use Setgroup / Checkgroup!!!
I setup a test (NoOP's are deleted): First caller should get first
line, second caller should get second line, third caller should get
busy and send an email. Note, that I used twice here to check the
first line!!!
[trunkint_A]
exten = _90N.,104,SetGroup(sip-13); increase Group counter
exten = _90N.,105,CheckGroup(1); check no more than
1 in this group
exten = _90N.,106,NoOp(Line 106)
exten = _90N.,107,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _90N.,108,hangup
;
exten = _90N.,206,SetGroup(sip-12)
exten = _90N.,207,CheckGroup(1)
exten = _90N.,208,NoOp(Line 208)
exten = _90N.,209,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _90N.,210,hangup
;
exten = _90N.,308,SetGroup(sip-13)
exten = _90N.,309,CheckGroup(1)
exten = _90N.,310,NoOp(Line 310)
exten = _90N.,311,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _90N.,312,hangup
;
exten = _90N.,410,Busy
exten = _90N.,411,SYSTEM(mail -s 'VPBX all lines in use'
[EMAIL PROTECTED])
I thought that 104 will set the Group counter sip-13 to 1 and
will use line 107 for the dial command
If another caller comes in that way, sip-13 would be 2 and
because Checkgroup allows only 1, the Group coutner would be
setback to 1 and it will follow the jump to 206 and sets the Group
counter sip-12 to 1
A third call should now find Group counter sip-12 and sip-13 set to
1 and give a busy signal and send an email.
HOWEVER, the log file show:
-- Executing SetGroup(Local/[EMAIL PROTECTED],2,
sip-13) in new stack
-- Executing CheckGroup(Local/[EMAIL PROTECTED],2, 1)
in new stack
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Line
106) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
so far so good!
-- Executing SetGroup(SIP/615-92c3, sip-13) in new stack
-- Executing CheckGroup(SIP/615-92c3, 1) in new stack
-- Executing NoOp(SIP/615-92c3, Line 106) in new stack
-- Executing Dial(SIP/615-92c3, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Ahh, it does not check Group counter sip-13, ... it checks
SIP/615-92c3 and Local/[EMAIL PROTECTED],2
How can I make it that it checks exactly the Group counter
sip-13
bye
Ronald
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Re: [Asterisk-Users] Line Noise HELP!
Hi Paul, there was a thread yesterday in regards to a few of us experiencing a very similar problem - a problem that (if the same for all of us), doesn't seem to have been properly diagnosed yet. One thing that appeared to be common to all of us was the version of Asterisk that we are running (1.0.6), is this the version that you are running? What FXO card are you using? Does the noise that you are complaining about occur on every call and, if so, always after exactly a minute, or is it more random? Starting to wonder if there isn't a problem with Stable, interested to hear what version you're running. Damian. Paul wrote: I recently hooked up my sipura IP phone and set it up as an SIP device to connect to asterisk. I am able to dial a number on the SIP phone, connect to an external number via the PSTN connected to asterisk and begin the conversation. At first, the audio quality is PERFECT, in both directions. I can hear the person clearly and they claim to hear me like im on a regular POTS line. After approximately 1 minute, the quality turns horrible and the person can no longer hear me, but I can faintly here them. There is a lot of static on the line, it almost sounds like an electronic device is interfering with it. I thought maybe it was a wireless phone or router, so I disconnected all those and put my cell phone in the other room. Still no change. Anyone have any ideas, this is really getting to be a problem. Paul ___ 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
Re: [Asterisk-Users] Line Noise HELP!
Hi mate, Interesting - you're using a different version of Asterisk than I am, but your problem sounds identical. We thought it was the SIP phones that we were using as well, but then it started occurring on the analogue phones as well. Post again when you've tried a new phone, will you? Let us know if the problem goes away. Cheers, Damian. Paul wrote: Damian, pbx*CLI show version Asterisk CVS-HEAD-03/23/05-00:44:07 built by [EMAIL PROTECTED] on a i586 running Linux There is my version info. Someone on the list has suggested that its my SIPura phone. It could very well be the phone, but it just seems unlikely that the conversation would be perfectly clear for some time before the static starts. I tried it today and was able to go approximately two minutes before it started. The FXO card is a generic x100P. Im going to try to get another IP phone and test it to see if its the phone. Let me know if you come up with any ideas. Paul From: Damian Funnell [mailto:[EMAIL PROTECTED]] Sent: Monday, April 11, 2005 12:49 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Line Noise HELP! Hi Paul, there was a thread yesterday in regards to a few of us experiencing a very similar problem - a problem that (if the same for all of us), doesn't seem to have been properly diagnosed yet. One thing that appeared to be common to all of us was the version of Asterisk that we are running (1.0.6), is this the version that you are running? What FXO card are you using? Does the noise that you are complaining about occur on every call and, if so, always after exactly a minute, or is it more random? Starting to wonder if there isn't a problem with Stable, interested to hear what version you're running. Damian. Paul wrote: I recently hooked up my sipura IP phone and set it up as an SIP device to connect to asterisk. I am able to dial a number on the SIP phone, connect to an external number via the PSTN connected to asterisk and begin the conversation. At first, the audio quality is PERFECT, in both directions. I can hear the person clearly and they claim to hear me like im on a regular POTS line. After approximately 1 minute, the quality turns horrible and the person can no longer hear me, but I can faintly here them. There is a lot of static on the line, it almost sounds like an electronic device is interfering with it. I thought maybe it was a wireless phone or router, so I disconnected all those and put my cell phone in the other room. Still no change. Anyone have any ideas, this is really getting to be a problem. Paul ___ 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 ___ 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] Line Noise HELP!
Can't help but wonder if this isn't a bug in Asterisk or one of it's modules, as there seems to be a lot of people experiencing the same problem, seemingly with different hardware and software configurations. Anyone know how (or if it's possible) to submit a bug report to Digium regarding this type of problem? I for one am going to have a customer return their Asterisk box for good if we can't get to the bottom of this soon. Andre Normandin wrote: Hi, I'm having very similiar problems.. However, I'm running a development version, and it happens on both SIP phones, and on Analog phones connected via Sipura SPA-2000's (I have 2 different SPA2000's, and 4 analog lines.. Seems to happen on all of them as well).. The problem seems to be EXACTLY as described. THe call seems fine at first, then within minutes the call degrades to the point that neither end can hear each other.. First, the volume seems to lower, and then static, breaking up, etc.. I have both DIGIUM X100 cards for my pots lines (3 of them), and BROADVOICE for outgoing calls. It seems to happen no matter if I'm on an analog line (I.E. someone called me), or if it was me that initiated the call (BROADVOICE outbound). I do have a 'remote' SIPURA SPA2000 located at a friends house in a different state -- he is an extension on my pbx so he can call me, and he can call his friends locally (He just moved away) via my POTS or BROADVOICE line.. He experiences the same problems as I described above, unless he calls me directly at my 'internal' extension, or I call him at his 'internal' extension.. I.E. If it doesn't touch POTS or BROADVOICE, the problem doesn't seem to occur..?? The other interesting thing that has happened of recent development is that some people are complaining that they are hearing the 'electronic beep' sound as if the call is being recorded, but I am not recording the call. This has occured with my friend as well as incoming and outgoing POTS/BROADVOICE calls. If anyone has an idea, I'd love to hear it.. The problem is driving me (and others who talk to me) crazy!!! - Andre -Original Message- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of *Damian Funnell *Sent:* Monday, April 11, 2005 3:08 PM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] Line Noise HELP! Hi mate, Interesting - you're using a different version of Asterisk than I am, but your problem sounds identical. We thought it was the SIP phones that we were using as well, but then it started occurring on the analogue phones as well. Post again when you've tried a new phone, will you? Let us know if the problem goes away. Cheers, Damian. Paul wrote: Damian, pbx*CLI show version Asterisk CVS-HEAD-03/23/05-00:44:07 built by [EMAIL PROTECTED] on a i586 running Linux There is my version info. Someone on the list has suggested that its my SIPura phone. It could very well be the phone, but it just seems unlikely that the conversation would be perfectly clear for some time before the static starts. I tried it today and was able to go approximately two minutes before it started. The FXO card is a generic x100P. Im going to try to get another IP phone and test it to see if its the phone. Let me know if you come up with any ideas. Paul From: Damian Funnell [mailto:[EMAIL PROTECTED] Sent: Monday, April 11, 2005 12:49 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Line Noise HELP! Hi Paul, there was a thread yesterday in regards to a few of us experiencing a very similar problem - a problem that (if the same for all of us), doesn't seem to have been properly diagnosed yet. One thing that appeared to be common to all of us was the version of Asterisk that we are running (1.0.6), is this the version that you are running? What FXO card are you using? Does the noise that you are complaining about occur on every call and, if so, always after exactly a minute, or is it more random? Starting to wonder if there isn't a problem with Stable, interested to hear what version you're running. Damian. Paul wrote: I recently hooked up my sipura IP phone and set it up as an SIP device to connect to asterisk. I am able to dial a number on the SIP phone, connect to an external number via the PSTN connected to asterisk and begin the conversation. At first, the audio quality is PERFECT, in both directions. I can hear the person clearly and they claim to hear me like im on a regular POTS line. After approximately 1 minute, the quality turns horrible and the person can no longer hear me, but I can faintly here them. There is a lot of static on the line
Re: [Asterisk-Users] Line Noise HELP!
Thanks for that Rich. Etheral trace is going to be almost impossible for various reasons, but will try the other two options. Can't find much online re. debugging - any chance of killing the box by turning this on? SIP show channels and the various CAPI show commands do not show anything out of the ordinary when the problem occurs. Cheers, D. Rich Adamson wrote: Can't help but wonder if this isn't a bug in Asterisk or one of it's modules, as there seems to be a lot of people experiencing the same problem, seemingly with different hardware and software configurations. Anyone know how (or if it's possible) to submit a bug report to Digium regarding this type of problem? I for one am going to have a customer return their Asterisk box for good if we can't get to the bottom of this soon. If I had the problem (which I don't with CVS-HEAD-04/07/05 and several different types of sip devices), I'd start with an ethereal trace that could be shared with those of us that can analyze it. Needs to include packets from when the audio goes to hell. Then, I'd turn on debugging (in logger.conf) and see if any messages are relative to the problem. If you can capture the results of 'sip show channels' and 'sip show channel ' for the bad conversation, that might be helpful to see as well. ___ 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
Re: [Asterisk-Users] Terrible crackling on analogue line and X100P card
Rich Adamson - would appreciate your advice as well, as your mail is the closest I have seen to a knowledgeable response so far in regards to this crackling issue. I have a customer who has a very similar crackling problem and to date we have suspected it to be the ISDN BRI adapter and/or CAPI, as it affects calls in progress over the genuine TDM400P card (which has 4 FXS ports) and SIP simultaneously. Although the nature of this problem seems to vary, the customer reports that the crackling usually starts when an external call is received at which time the crackling overwhelms all voice channels and everyone has to hang up and re-establish the calls (after which everything works fine again). Could it be possible that a problem with the TDM400P could affect SIP calls on Asterisk as well? I don't think I have an interrupt problem (see interrupt table pasted below) and the output of zttest appears to be ok (at least as good as 99.96% accuracy), so we are stumped. Linux, Asterisk and ISDN driver versions as follows. Appreciate any help you can offer. Cheers, Damian. 1) Linux localhost.localdomain 2.4.21-4.ELsmp #1 SMP Fri Oct 3 17:52:56 EDT 2003 i686 i686 i386 GNU/Linux 2) Asterisk 1.0.6 built by [EMAIL PROTECTED] on a i686 running Linux 3) divas4linux_2_4_21_4_RHEL3_i686_SMP-104.345-1 *** zttest output *** --- Results after 43 passes --- Best: 99.975586 -- Worst: 99.963379 *** End zttest output *** *** Interrupt Table *** [EMAIL PROTECTED] root]# cat /proc/interrupts CPU0 CPU1 0: 40674647 0IO-APIC-edge timer 1:969 0IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 6:282 0IO-APIC-edge floppy 8: 1 0IO-APIC-edge rtc 12: 7132 0IO-APIC-edge PS/2 Mouse 14: 6650 1IO-APIC-edge ide0 16: 0 0 IO-APIC-level usb-uhci 17: 406583244 0 IO-APIC-level wctdm 18:5450488 0 IO-APIC-level eth0 19: 0 0 IO-APIC-level usb-uhci 21: 11834725 0 IO-APIC-level DIVA 4BRI 15587 23: 0 0 IO-APIC-level ehci-hcd 27: 339738 0 IO-APIC-level a320raid NMI: 0 0 LOC: 40673196 40672862 ERR: 0 MIS: 0 *** End interrupt table *** 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 Rich Adamson wrote: I'm experiencing terrible trouble with crackling and noise on an analogue line connected to an X100P (compatible) card. I've checked the line with a normal analogue phone and it works fine, clear as a bell, but any outgoing or incoming calls to Asterisk are almost completely drowned out by loud crackling. I've attempted to adjust the RX and TX gains, but to no avail. There's also an echo, but only one way. I'm assuming this is a separate issue so I've not done much to investigate that, but I may be wrong so if it is related does anyone have any suggestions? I never had this trouble with ISDN, but then I wouldn't would I? :) If anyone can wave a magic wand, or at the very least point me to a website where I can get my own magic wand, please let me know. What country are you in, and does the chipset on the compat card support the telco standards in your country? If the chipset doesn't match your telco standards, there is a high probability you won't get rid of the echo. If it does match, then try echotraining=800 echocancel=yes Regarding the crackling noise, have you checked for shared interrupts (cat /proc/interrupts)? If you run cat /proc/interrupts every ten seconds, do you see calculated interrupt values of about 1,000? Go to /usr/src/zaptel directory and run ./zttest Do you get something close to 100% over some period of time? What version of asterisk are you running? ___ 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
Re: [Asterisk-Users] Terrible crackling on analogue line and X100P card
I have a very similar problem that I have been grappling with for a while. I've got a genuine TDM400P with four FXS ports and am using an Eicon Server quad BRI ISDN (using CAPI) for external calls. To date we have had no luck at all in diagnosing this problem as we too have periodic problems where the crackling occurs only sometimes, but affects all calls that are in progress (including those using the TDM400P and SIP calls that do not). Asterisk does not report any problems when this problem occurs, but it is sufficiently bad to force everyone to terminate in-progress calls (at which time everything works fine again). Appreciate hearing if you guys find a resolution to the problem that you are having, as we have had zero luck so far. dean collins wrote: I have a similar situation but it seems to vary from call to call sometimes. Using 2 digium genuine x100p's in a dell with riser card. I'm wondering if it is something to do with the riser because it doesn't seem to matter if I swap various cords, positions, etc. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Stuart Ford Sent: Saturday, April 09, 2005 9:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Terrible crackling on analogue line and X100P card Dear all ... I'm experiencing terrible trouble with crackling and noise on an analogue line connected to an X100P (compatible) card. I've checked the line with a normal analogue phone and it works fine, clear as a bell, but any outgoing or incoming calls to Asterisk are almost completely drowned out by loud crackling. I've attempted to adjust the RX and TX gains, but to no avail. There's also an echo, but only one way. I'm assuming this is a separate issue so I've not done much to investigate that, but I may be wrong so if it is related does anyone have any suggestions? I never had this trouble with ISDN, but then I wouldn't would I? :) If anyone can wave a magic wand, or at the very least point me to a website where I can get my own magic wand, please let me know. Thanks Stuart ___ 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 ___ 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] Terrible crackling on analogue line and X100P card
Forgot to mention - we are using an IBM xSeries 206 Server, so the Dell riser card may not be the issue if we are having the same problem. 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 Damian Funnell wrote: I have a very similar problem that I have been grappling with for a while. I've got a genuine TDM400P with four FXS ports and am using an Eicon Server quad BRI ISDN (using CAPI) for external calls. To date we have had no luck at all in diagnosing this problem as we too have periodic problems where the crackling occurs only sometimes, but affects all calls that are in progress (including those using the TDM400P and SIP calls that do not). Asterisk does not report any problems when this problem occurs, but it is sufficiently bad to force everyone to terminate in-progress calls (at which time everything works fine again). Appreciate hearing if you guys find a resolution to the problem that you are having, as we have had zero luck so far. dean collins wrote: I have a similar situation but it seems to vary from call to call sometimes. Using 2 digium genuine x100p's in a dell with riser card. I'm wondering if it is something to do with the riser because it doesn't seem to matter if I swap various cords, positions, etc. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Stuart Ford Sent: Saturday, April 09, 2005 9:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Terrible crackling on analogue line and X100P card Dear all ... I'm experiencing terrible trouble with crackling and noise on an analogue line connected to an X100P (compatible) card. I've checked the line with a normal analogue phone and it works fine, clear as a bell, but any outgoing or incoming calls to Asterisk are almost completely drowned out by loud crackling. I've attempted to adjust the RX and TX gains, but to no avail. There's also an echo, but only one way. I'm assuming this is a separate issue so I've not done much to investigate that, but I may be wrong so if it is related does anyone have any suggestions? I never had this trouble with ISDN, but then I wouldn't would I? :) If anyone can wave a magic wand, or at the very least point me to a website where I can get my own magic wand, please let me know. Thanks Stuart ___ 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 ___ 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
Re: [Asterisk-Users] How does asterisk know the did called on?
Hi Courtney,
This really depends on the type of channel you are using. We use CAPI
with one of our customers and they have ten DID's (DDI's in NZ) and use
four BRI lines. Here is a snippet from their extensions.conf that
shows how it dials certain extensions based on the number dialled. The
three digit number in the dialling rules are presented by the ISDN
channel/CAPI and will probably be slightly different based on your
telephony provider. We had to dial one of the DDI's and see Asterisk
reported it as being on the console to figure out how long the DDI
number was.
(Note that you would be much better off using macros rather than the
numerous entries that we have in extensions.conf, but we have
structured it like this upon customer request).
;[incoming]
exten = s,1,Answer
exten = s,2,Dial(SIP/9295,20,t,)
; No answer or busy, so dial other extension(s) - can add others to
this list if required
exten = s,3,Dial(SIP/9295SIP/9287,,t,)
;Queue not set up, do not use the following unless configuring it first.
;exten = s,3,Queue(receptionqueue)
;DDI's follow
;Manager (290)
exten = 290,1,Answer
exten = 290,2,Dial(SIP/9290,,t,)
exten = 290,3,Voicemail(9290)
exten = 290,4,Congestion
;Supervisor 1 (291)
exten = 291,1,Answer
exten = 291,2,Dial(SIP/9291,,t,)
exten = 291,3,Voicemail(9291)
exten = 291,4,Congestion
Hope this helps.
Damian.
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
Courtney Couch wrote:
If I were
to buy 20 did's how do I know within asterisk which number was dialed?
(like say I want a few of the did's to ring specific extensions if they
are dialed and others to go through the menu)
Is there any ${var} that has the number dialed in on? (that would be
optimum).
-Courtney.
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Re: [Asterisk-Users] Installing CAPI
Hi there, We recently did our first * install with CAPI and we found the levels of support (and general knowledge) within the community seriously wanting. In fact, we found things so bad that I would caution against using CAPI unless you are feeling particularly game and confident in your abilities to fix problems, as you are likely to find it very difficult to get help if you need it. Out of the half dozen or so help requests that I or my colleagues posted to this forum or to the #asterisk IRC channel, for example, we didn't receive a single helpful response. Not one. Not that there wasn't anyone who was willing to help, but there just didn't seem to be anyone around who was using CAPI in anger. We originally chose CAPI over ISDN4Linux because of the commercial support that was supposedly available through junghanns.net (CAPI also provides a better feature set than ISDN4Linux, but we don't use any of the additional features, so this wasn't a consideration for us), but when we called upon junghanns.net for support it took them so long to respond that we needn't have bothered (we had stumbled across a fix ourselves by the time we got a response from them). If this hasn't scared you off then check out the documentation at http://www.junghanns.net/asterisk/ and the sample files/readme that come with the CAPI source. There is also a fairly good configuration guide at http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI and the CAPI readme is reproduced at http://www.voip-info.org/wiki-Asterisk+CAPI+Readme. Drop me a mail at damian dot funnell at fff dot co dot nz if you would like me to send you a copy of our conf files so you can see how we're using it. Right now we are trying to diagnose a problem where the voice channels over CAPI fall apart a few times per day, resulting in all external calls having to be terminated. We don't know if this problem is CAPI related, but predictably we haven't been able to find anyone in the community who can help us figure it out. Best regards, Damian. [EMAIL PROTECTED] wrote: Hi! I can't find any instructions of installing capi and chan_capi. Do you know any site with instructions or can you give me step by step help with this. Thank you for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ 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
[Asterisk-Users] External voice channels pack up
Hi all, Having an intermittent problem with * where external voice channels suddenly pack up completely, resulting in large amounts of 'crackling' on the line (audible at both ends of the channel). The channel stays open and each end can still hear the other, but the crackling is loud enough to end the conversation pretty quickly. Problem only seems to occur when more than one call is in progress and also appears to occur on occasion when another extension is picked up or hung up. Problem appears to affect all in-use voice channels simultaneously. Haven't been able to confirm whether this problem also affects internal calls due to intermittent nature (and fact that customer makes very few internal calls). Problem can sometimes result in a single loud 'spike' when another line is picked up, but most often results in a loud, continuous crackling that won't go away until the channel is terminated. As soon as the user hangs up and re-dials the problem goes away (until next time). One user reports that problem even started when second line started ringing on their extension (i.e. when call waiting kicked in due to incoming call on their DDI), meaning that call in progress had to be terminated. Incoming call (i.e. call that was on call waiting) seemed to be fine, however. Running latest * and CAPI and using Eicon Server 4 port ISDN (BRI) card on RHES. Mixture of 5 x SIP phones (BT102) and 4 x analogue cordless phones (all connected to Digium TDM400P). Originally suspected problem was to do with BT102's (i.e. SIP handsets), but problem has since occurred with analogue phones as well. Appreciate any help with this, as source of problem not immediately apparent and customer patience wearing pretty thin. Cheers, Damian Funnell. ___ 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] CAPI questions
Hi all, I have two questions regarding CAPI. Excuse the fact that they are very 'newbie' in nature, but the CAPI documentation is wafer thin! Firstly I have four BRI adapters (all trunks and controlled by CAPI) in my * box and I would like to know whether I can group these together for dialling out in the same way that ZAP channels can be grouped together. Secondly I have a problem where * doesn't seem to recognise incoming calls when one of the B channels is in use. If someone is on the phone to an external number, for example, then incoming calls ring (for the caller, at least) but * doesn't seem to have any idea that the channel is ringing. Lastly, my capi.conf (as below) only defines one controller as this is what we are testing with. My understanding is that the interface block (starting with 'msn=470' and ending with 'devices=2') needs to be repeated for each of the four BRI adapters, but with the correct MSN for each. The documentation I have seen is ambiguous, can anyone confirm this is correct? Thanks in advance, I M Newbie. ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 musiconhold=random [interfaces] msn=470 incomingmsn=* controller=1 softdtmf=1 accountcode= context=incoming ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 -- 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 ___ 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] CAPI questions
Thanks Elmar. I assume it is up to the carrier to determine the MSN for each connection? D. 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 Elmar Haneke wrote: Lastly, my capi.conf (as below) only defines one controller as this is what we are testing with. My understanding is that the interface block (starting with 'msn=470' and ending with 'devices=2') needs to be repeated for each of the four BRI adapters, but with the correct MSN for each. If you have different MSN then you have to repeat it for each controller. If they are on the same MSN you can enter "devices=8" and "controller=1,2,3,4" or repeat which should also work. Elmar ___ 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
[Asterisk-Users] CAPI trunks
Hi all, Can anyone help me with a CAPI problem that I am having. I've got one BRI trunk (will have 4 when it goes into production) and when one of the B channels is in use (i.e. there is an incoming/outgoing call in progress) I can't get Asterisk to answer the other ringing B channel (Asterisk doesn't even seem to know that it is ringing). Incoming calls work fine when no channels are in use and Asterisk will still dial out on the second channel (if the first is in use). Thanks, Damian. capi.conf: ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 musiconhold=random [interfaces] msn=470 incomingmsn=* controller=1 softdtmf=1 accountcode= context=incoming ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 -- 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 ___ 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] Recommended Phone for beginner
Hi Ryan, I've used the BudgetTone 101 on several accounts and they certainly aren't the best phone on the market, but they have so far been reliable (touch wood) and are pretty straightforward to set up. Call quality would probably rate at a 8 out of 10 on these phones, but that's not much worse than the 3com and Cisco phones that we have installed. Cheers, Damian. 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 Ryan Burke wrote: Hello everyone, I've been watching this list for a while, but it is the first time I've posted. I'ved decided to setup a * server for my house and will need 3 phones (one main, one for my wife, and one for my office). I was wondering if there was a particular brand that people reommended? I'd like ot get an actual SIP phone, instead of an adapter like the Sipura SPA-2000. I've been looking at the Grandstream BudgetTone 100 series but after looking at the Wiki for setting up * with that phone it looks like it might be more trouble than its worth. Of course I would love a Cisco 79* but I'd like to keep the cost at a minimum but get a good amount of flexibility in tersm of features. Hopefully once I get over the learning hump I can start contributing to this list. Any input would be appreciated. Thanks, Ryan ___ 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
