[asterisk-users] Chan_sip max channels limit?
Hello, I have asterisk 1.4.4 running with anonymous sip calls enabled and I am testing the box for load using sipp with something like this - sipp -sn uac -s 10 -d 6 -i 192.168.1.49 -l 110 -r 5 -trace_err 192.168.1.50 Asterisk picks up the call and runs a test php-agi file that plays a .gsm file. As soon as the number of active calls reaches 99, asterisk starts Declining further calls. I thought this was a call-limit problem so created a type=friend entry for the sipp client's IP and it still does not change anything. Is there any way to change this limit? Any clues? Thanks, -- VaibhaV Sharma ___ --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] VoipSupply? [Semi-Urgent]
I don't think this is a problem because of the snow storm. I just got off the phone with them. The sales guy I used to deal with left a few months back and since then, its been a pain to get anything done with them. People I have dealt with had no clue. I called them this morning for a problem to be told that a technical support person will call me back within an hour. Then no one calls back for 5 hours. So when I call them back, I am told We don't do technical support on the phone. I don't know who told you that. The lady who I was speaking with had no clue of what I was asking for. She kept putting me on hold to ask someone for an answer. What was my question? Q. We purchased 25 polycom IP 601/501 from you a while back and one of them has a faulty power supply. How do I get a new one? A. Hold on Oh! You have to speak with RMA and not technical support. Go to our website / rma and submit an RMA. Q. Well, power supplies don't have serial numbers! A. Hold on. .. No you will have to obtain an RMA! Q. Well, what do I send to you? Can I speak with a technical support person? A. Hold on. .. Send us the power supply *and* the phone. Q. It will cost me the money for a power supply to ship the phone to you. Can you tell me somewhere else I can get just the power supply? A. If I had the answer I would have told you, sir. Gah! This is just one case. I am really disappointed with their service. I am worried about our technical support options for the polycom phones after the last few expereinces with Voipsupply. -- VaibhaV On 10/14/06 10:36 AM, Matt [EMAIL PROTECTED] wrote: Contact them again... they have always been very good... I'm chocking this up to the snow storm. On 10/13/06, Shaw Terwilliger [EMAIL PROTECTED] wrote: Matt wrote: Hi, Does anyone know what is going on with voipsupply? My sales guy hasn't been online in several days, their 800 number is fasy busy, as are their direct lines. And the canadian store website is down. What the heck is going on? If you search the archives from a few months ago you'll find a few unhappy voipsupply customers (including me). They never shipped what I ordered, didn't respond to any e-mail or calls. The president saw the list traffic and sent me a long apology (stating his commitment to service) and offered to send me an extra component that I had cancelled the order for--free of charge--as a show of good will. It's been two or three months since that promise, and I never received the part. He hasn't responded to my follow-up did you really mean it? e-mail either. -- Shaw Terwilliger [EMAIL PROTECTED] SourceGear LLC ___ --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] Polycom IP600 HTTP Provisioning problem
Hello, The latest Polycom firmware (1.6.x series) supports HTTP(s) provisioning that I have been trying to setup. The admin guide mentions that in the boot settings for the configuration server, URLs of this format can be used - http://user:[EMAIL PROTECTED]/dir/config.cfg But when I use that, the phone seems to be ignoring the sub-dir text and just tries to send requests like these - 172.16.19.160 - - [01/Aug/2006:18:24:54 -0400] GET /bootrom.ld HTTP/1.1 404 288 - Polycom-FileManager/1.0 (libcurl/7.12.1 OpenSSL/0.9.7d) (SIP-1.6.3:0067;SPIPPolycomSoundPointIP-SPIP_601) 172.16.19.160 - - [01/Aug/2006:18:24:54 -0400] GET /0004f3445566.cfg HTTP/1.1 404 294 - Polycom-FileManager/1.0 (libcurl/7.12.1 OpenSSL/0.9.7d) (SIP-1.6.3:0067;SPIPPolycomSoundPointIP-SPIP_601) 172.16.19.160 - - [01/Aug/2006:18:24:54 -0400] GET /.cfg HTTP/1.1 404 294 - Polycom-FileManager/1.0 (libcurl/7.12.1 OpenSSL/0.9.7d) (SIP-1.6.3:0067;SPIPPolycomSoundPointIP-SPIP_601) 172.16.19.160 - - [01/Aug/2006:18:25:07 -0400] GET /0004f2445566-phone.cfg HTTP/1.1 404 300 - Polycom-FileManager/1.0 (libcurl/7.12.1 OpenSSL/0.9.7d) (SIP-1.6.3:0067;SPIPPolycomSoundPointIP-SPIP_601) Which obviously generates a 404. Has anyone tried this with success? The only solution to this that I can think of is to configure a virtual host on the apache side and use a different URL. It would be more convenient if I don't have to create another virtual host on the machine just for the phone configs. Any clues? Thanks, -- VaibhaV Sharma Ishi Systems Inc. http://ishisystems.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ztmonitor values when zap channel is onhook
Hello, In my quest to figure out the source of the random echo on our shiny new asterisk install, I have been using ztmonitor on the TDM400p channels for the good part of today. I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to them (last 2 channels are unused but configured in zaptel). Even when the lines are onhook, the Tx values settle down to 0 but the Rx values still jump up and down. For some lines the values vary around 440, for others around 250 and for one of the unused FXO ports, its around 97 all the time. With txgain/rxgain set to 0.0, call volumes were considerably low. Hence, I have txgain set to 1.0 and rxgain set to 6.0. I was wondering if the ztmonitor Rx: values were normal for what they show, or are they too supposed to settle down to 0. If this is not normal, does this indicate any problems with the pots lines? -- VaibhaV ___ --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] Ztmonitor values when zap channel is onhook
On Fri, 2005-08-26 at 12:37 -0500, Eric Wieling aka ManxPower wrote: VaibhaV Sharma wrote: Hello, I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to them (last 2 channels are unused but configured in zaptel). Even when the lines are onhook, the Tx values settle down to 0 but the Rx values still jump up and down. For some lines the values vary around 440, for others around 250 and for one of the unused FXO ports, its around 97 all the time. With txgain/rxgain set to 0.0, call volumes were considerably low. Hence, I have txgain set to 1.0 and rxgain set to 6.0. why not txgain=-6 to start out with? With txgain set to 0.0, people on the other end complained about low volume from our side hence I started out with a positive value. Are the ztmonitor Rx: values of around 90 - 400 normal even when the zap channel is Onhook or do I have some problem on the pots lines that I should fix? Thanks, -- VaibhaV ___ --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] Small office setup/using analog lines w/ Ast erisk
On Tue, 2005-08-23 at 01:30 -0600, Colin Anderson wrote: Good luck and please keep posting so everyone can learn from your experience. Hmmm... we too moved our telephone system at work to asterisk just a few days back and since then, I have been following this thread. A few snippets from my experience (which BTW are very similar to the original poster of this thread). Old setup Avaya-Lucent Partner III with 15+ extensions and 8 voice lines Hardware: * Dell PowerEdge 1800 with a single Xeon 3.2 ghz * 2 SATA Hdds with H/W RAID (RAID mainly because we plan to do a lot of recording on conference calls + fault tolerance) * 2 TDM400P cards with 8 incoming pots lines * About 13 polycom IP600, one IP501 and 2 cisco 7960 phone extensions + a lot of remote sip clients * One TDM card is Rev I and the other one is a few months older. Even the red modules on the cards are a mixture of 3 revisions as we bought them in installments while testing the setup. Due to the short deadlines, we decided to go with [EMAIL PROTECTED] for now and then later on move to a custom programmed asterisk install. Running AAH 1.3 on the above hardware (darn! 1.5 is already out). Problems: 1. Poweredge 1800 only had 2 free PCI-X slots. A quick call to digium confirmed that the cards would work just fine on those slots. 2. After the install, users complained about low call volumes on both sides of the calls. This was quickly fixed by tinkering with the rxgain and txgain in zapata.conf. But this introduced another problem (point 4). 3. On one of the lines, there was a lot of disturbance. It was a loud/bad buzzing noise which would be heard randomly on any call through that line. Called up digium and the support technician suggested swapping the red modules to see if the particular module was faulty. I rebooted the machine at the end of the day and things were fine even without swapping the modules. I don't know what the problem was but it was only on that particular pots line. 4. Major echo and chit-chat noise issues on random calls. Some of our users have calls running for an average of 30 - 40 mins and they were experiencing random echo problems to the point that sometimes they would leave a voicemail on someone's cell phone and the user would hear only echoed voice and nothing else. Changing the txgain made the echo worse so I had to settle for 0.0 as its value. Tried increasing / decreasing echocancel from the default 128 to 256 or 64/32 but echo got worse both ways. So the only change I could make was the rxgain so that call volume was better on this side. Tried changing the echotraining values too but that had no effect on the echo. At this point, I had rxgain set to 10.5 and txgain set to 0.0. I figured that the echo was being generated maybe by the phones as a few users complained of echo on internal calls too. So I looked up polycom's config file and enabled the echo supression tags. That did not help much. Less, but still random echo problems. I found a page somewhere on voip-info.org which mentioned that some digium card revisions have compatibility issues with some motherboards. The list specifically mentioned the Dell SC series as being problematic. The newer Poweredge series seemed to be fine but I was not convinced, so I called up digium and the technician sent me a standard email with about 4 - 5 things that might be causing the random echo problems. Those included - - IRQ sharing issues. Assign separate IRQs to each digium card. - IDE drives have DMA mode set to on and that IDE drives are preferred instead of SATA / SCSI, specifically because IDE drives can be set to use UDMA2. - IRQ misses are bad and should be minimised for the digium cards. This can be checked using zttest application. Minimum acceptable value is 99.98% - Avoid running X-windows on the box and disable frame-buffer by using vga=normal on boot. - If you still see irq misses, try using acpi=off and/or noapic - Try disabling hyperthreading if it is enabled Except the IRQ sharing part, I tried everything in that list and nothing seemed to change. Still lot of random echo on calls. Changing the IRQ settings was a bit difficult as I only had 2 free PCI-X (now occupied) slots for the two digium cards, so changing slots was not an option. I disabled the USB subsystem, floppy drive and the onboard SATA controller. Even then, one of the digium cards is sharing the IRQ with the network controller and I see no way to be able to change that. I am thinking of calling dell if this situation persists. Day before yesterday, I took the TDM cards out and swapped a few modules between themselves. That did not change anything. Then just yesterday, I found this URL in one of the emails on this thread - http://lists.digium.com/pipermail/asterisk-users/2005-March/096754.html Out of this list, I tried the following just yesterday evening - Recompile zaptel with - MMX enabled - Enable the AGGRESSIVE_SUPPRESSOR with MARK2 Since this
Re: [Asterisk-Users] Small office setup/using analog lines w/ Ast erisk
On Tue, 2005-08-23 at 14:38 -0400, Andrew Kohlsmith wrote: On Tuesday 23 August 2005 13:25, VaibhaV Sharma wrote: * 2 SATA Hdds with H/W RAID (RAID mainly because we plan to do a lot of recording on conference calls + fault tolerance) What good does RAID give you on writes? None whatsoever. RAID only helps performance on reading. Fault tolerance aside, I'd get it working first, THEN play with RAID. I meant to point out that we use RAID not for performance reasons but to achieve some redundancy to start with, as a lot of other important systems are going to be integrated with this box. Changing the txgain made the echo worse so I had to settle for 0.0 as its value. Tried increasing / decreasing echocancel from the default 128 to 256 or 64/32 but echo got worse both ways. So the only change I could make was the rxgain so that call volume was better on this side. Tried changing the echotraining values too but that had no effect on the echo. At this point, I had rxgain set to 10.5 and txgain set to 0.0. Run ztmonitor and try to adjust them as described here: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html Yes, I forgot to mention that I used ztmonitor while tweaking the rx/txgain numbers. The call quality has acceptable for now. Also make these changes to your zaptel Makefile: add CFLAGS+=-march=pentium4 KFLAGS+=-march=penitum4 underneath the comments about zconfig.h. Also you could try defining XLAW as described in zconfig.h to optimize for a few zap channels (not sure if few is 8 or not). Finally, enable MMX in zconfig.h. Considering that we would be increasing the number of incoming lines in the future, do you suggest moving to a channel bank right away? Maybe something that will do echo canceling on the hardware level? - Enable the AGGRESSIVE_SUPPRESSOR with MARK2 Agressive mode turns your phones into half-duplex devices. When you're talking, you don't hear ANYTHING from the other side, and vice-versa. That's why you don't hear any echo. Ah! That explains things a bit. Thanks a lot, -- VaibhaV http://vsharma.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
Re: [Asterisk-Users] Small office setup/using analog lines w/ Ast erisk
On Tue, 2005-08-23 at 16:50 -0400, Andrew Kohlsmith wrote: On Tuesday 23 August 2005 16:11, VaibhaV Sharma wrote: I meant to point out that we use RAID not for performance reasons but to achieve some redundancy to start with, as a lot of other important systems are going to be integrated with this box. Problem #1. Trying to do more than JUST PBX with the box. Put all your integration stuff on a box in the same rack but don't put everything in your PBX box. Resource starvation is not a good thing to have. Na, none of the other stuff is running on this box. By integration, I meant integrating pbx services into other apps or maybe reporting based on call records. e.g. when and how many times was a particular customer contact called in the past few months. These features are mostly custom to our own apps and will not need any changes in asterisk. -- VaibhaV http://vsharma.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
