Re: [asterisk-users] Problems using TE412P and TDM400B in a IBM x3650
James FitzGibbon wrote: Another day, another apparant unexplained hardware incompatibility. I have a TE412P and a TDM400B living quite happily in a whitebox using an Intel motherboard: http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm I tried to move to an IBM x3650 system. It uses a slightly newer chipset, but apparantly it's in the same family. The SE-7230 board has been EOL'd and the suggested replacement uses the same chipset as the x3650. I had to get a PCI-X riser cage to put the cards into, as the server only supports PCIe as shipped. http://www-03.ibm.com/systems/x/rack/x3650/specs.html When I just have the TE412P in the server, no problems. If I put both the TE412P and the TDM400B in, I get no end of errors. When I put the TE412P in the first PCI-X slot and the TDM400B in the second, then none of my PRI channels will get out of red alarm - they go red as soon as I load zaptel, and stay there through ztcfg, starting asterisk, restarting zaptel via the Asterisk CLI, etc. If I swap the cards, then only one of the ports (#4) stays in red alarm, while the other 3 seem to be fine. I checked /proc/interrupts, and both cards were getting their own interrupt (forgot to save the output unfortunately, and I'm back on the original hardware right now). Has anyone run this type of hardware combo successfully, or had similar problems on other hardware that they got around? Can you make sure Digium tech support hears about this, so that it can be addressed? Thanks, -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Platform Recommendations for Digium Card Compatability
James FitzGibbon wrote: On 8/10/07, Jason K. Carter [EMAIL PROTECTED] wrote: Could everyone that has a working production Asterisk server that uses a Digium telephony card as a BRI/PRI gateway let me know what motherboard/processor your server uses? Currently running a TE412P in a IBM x3650 Model 7979. I had some problems when I also had a TDM400B in the same system. I have also run this card successfully on a Intel SE7230NH-1 board (having the TDM400B installed as well was not a problem on this board) I had a reproduceable kernel panic under moderate load running this board on a HP DL380G5 with Zaptel 1.4. Zaptel 1.2 was just fine. Do you have any more information on this? (i.e. stacktrace and error associated with it)? Make sure you're testing with a current 1.2 or 1.4 version of the drivers. There have been a few bug fixes in the last few releases. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Digium TE120P card support MFCR2
[EMAIL PROTECTED] wrote: Hi, I have successfully configured DIGIUM card and successfully communicated through it to the another E1 card running application. Can anybody tell me does TE120P support MFC/R2 protocol. Zaptel drivers are designed to be protocol independent. The TE120P should work with any protocol that any other T1/E1 zaptel card supports. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2 TDM24xx and B410P
Florent Barbier wrote: Hi here, Did you get any solution ? I've quiet the same pb : http://forums.digium.com/viewtopic.php?t=17394 Thank you for your answer. flo_turc Sorry for the late reply :-( We are aware of that particular issue, and working on tracking it down. Very big sorry for the inconvenience in the mean time :-( -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
[EMAIL PROTECTED] wrote: Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the same jacks, and the line is crystal-clear. Below is my zapata.conf, if you guys have any ideas how I might resolve this, I'd appreciate it. I have installed from sources Asterisk 1.2.22 and zaptel-1.2.19 on a debian etch x86_64. Have you tried running fxotune on it? If you have (or haven't) make sure you try the zaptel-1.4 version of fxotune. It has improved significantly since 1.2. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel update locks up computer from 1.2.9.1 to 1.2.19
Tzafrir Cohen wrote: On Wed, Aug 15, 2007 at 09:22:45PM -0400, Jerry Geis wrote: I am trying to update a machine with a TE210P card setup as PRI. Running Centos 4.4. What is the output of: uname -r I stop asterisk, I do service zaptel stop. I look at lsmod and all zaptel modules are unloaded. I compile zaptel 1.2.19, I install zaptel. when I do the service zaptel start, the machine locks up. I reboot the machine and it locks up when loading zaptel. When exactly? If you manually run: modprobe zaptel # is that the right driver? insmod wct4xxp # you may need to wait for a while here. udevd takes its time on centos4 # to populate /dev/zap ztcfg What exactly does lock up mean, by the way? Does it output a stack trace on the screen? Can you not ssh into anymore? Or is does the current console you are on simply hang but the rest of the system behaves as normal. The answer to this is important. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
shadowym wrote: Please explain to me how FXO tune would fix popping and clicking sounds??? As mentioned by Stephen, if the echo canceler is improperly tuning that certainly might be possible. But moreover, if there is ambient line noise that is on the line, fxotune will try to pick the best settings on the line interface to either mitigate any line noise that it receives in the audio receive path. One other possibility is you could see if it the clicking and popping correlates to hard drive activity... if that's so, you might have a hard drive or raid controller disabling interrupts for too long. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compatibility of PRI Two B channel transfers TBTC/2BTC
Matt Florell wrote: Hello, A client has asked for Two B channel Transfer capability (known as TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG Path Replacement) in a new Asterisk system and so I researched the capability and came up with quite a few gaps in documentation. From what I've gathered, the official Digium statement is that is works with DMS100 only, and only in Asterisk 1.4.X : http://kb.digium.com/entry/26/140/ This definitely works. I wrote it and tested it myself. Although in a bugtracker posting with a patch from over two years ago, Matt Fredrickson from Digium says that it works with 5ESS under Asterisk 1.2.X: http://bugs.digium.com/view.php?id=3554 There's an implementation I scrubbed out a couple of years ago, but I think there was a bug in it that I was not able to fix. When push came to shove, and I needed a switch to debug it on (and when I had more time to work on it), nobody offered switch access so that I could debug it. So I don't think it is working right now. There are also bounties and claims of this feature working on NI2 protocol(although no patches posted) on the voip-info.org Wiki: http://www.voip-info.org/wiki/view/Asterisk+bounty+PRI+2B+channel+transfer+for+NI2+PRI+line http://www.voip-info.org/wiki/index.php?page=Asterisk%20bounty%20PRI%202B%20channel%20transfer Yeah, well, they're really old :-) Try getting a hold of the authors. As for actually using this feature, you apparently need to add the following lines to the zapata.conf section that you want to be able to use 2BCT: facilityenable = yes transfer=yes Yes, that is correct. To execute the transfer, you need to use the Transfer cmd within Asterisk: http://voipinfo.org/wiki/view/Asterisk+cmd+Transfer This is incorrect. If you have transfer=yes and facility=yes in zapata.conf for both channels, and both channels meet all the other criteria for TBCT (on the same PRI, and a few other switch dependent rules), when a native bridge is attempted, it automatically attempts to pass the calls up to the upstream switch. If it is successful, your calls will remain up, but you will get a hangup in asterisk on both calls. And according to this post, you can only do 2BCT transfers if the first call is inbound: http://www.mail-archive.com/[EMAIL PROTECTED]/msg25131.html That's a rule only for DMS100. Does 2BCT work with DMS100 and 5ESS right now? Last I heard (a couple of years ago) it doesn't. Are there people using this in production right now that can shed some more light on exactly how they are using it, and executing the transfers? I hope I answered your questions :-) -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.2.20.1 and 1.4.5.1 released
Steve Kennedy wrote: On Wed, Aug 22, 2007 at 01:19:14PM -0500, Asterisk Development Team wrote: The Asterisk.org development team has announced the release of Zaptel versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in the install target in the Makefile of the 1.4.5 and 1.2.20 Zaptel releases, as well as a handful of other issues. See the respective Changelogs for more details. Both releases are available as a tarball as well as a patch against the previous release. They are available for download from downloads.digium.com. Don't seem to be on www.asterisk.org (1.2.19 and 1.4.4) Sorry, I still have to get the powers that be to update the home page :-) -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2 + Zaptel 1.4 + HPEC = Crash?
Trevor Peirce wrote: Hello, Has anyone tried the combination of asterisk 1.2.24, zaptel 1.4.5 and HPEC 9.00.003? First of all... Why are you using zaptel 1.4.5 with asterisk 1.2? That is a red flag in itself. In particular, with a hardware configuration similar to: Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) I have two fully independent systems (both production, so I can't do further testing unfortunately) that crash anywhere between an hour and a day after booting under a minimal load. If HPEC is disabled, the problem is gone (but really bad echo). If I use zaptel 1.2.20.1, the problem is gone. The result is a kernel panic followed by an automatic reboot. Nothing is written to log files so I cannot provide any debug information. As mentioned this has happened on multiple production machines and I do not have any other wctdm cards to test with. I would be curious to hear if anyone else noticed the same problem or if they have it working. What are the common denominators? Thanks, Trevor -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 and TDM800 fxo stop answering
Stefano Arata wrote: Hi, I have two asterisk with the Digium TDM400 installed on the first and the TDM800 installed on the second. Both systems are linux Debian 4.0 whith kernel 2.6.18 and asterisk 1.2.24. Often the cards stop answering calls, and I can't make or receive calls; I need to reboot the system or manually reload the zaptel modules to restore it. I've tried zaptel versions 1.2.18, 1.2.19 and 1.2.20 too but the problem remains. I can't find any error in the asterisk log files nor in the syslog but I've found this suggestion http://www.voip-info.org/wiki/view/Asterisk+automatic+daily+restart on wiki, that suggests to set up a cron job to restart the driver daily, but this doesn't work for me. Are there other solutions to this problem? First off, could you try zaptel-1.2.20.1? I made a fix that could possibly be related to this and it was either in 1.2.20 or 1.2.20.1. If that doesn't fix, could you please contact Digium tech support so we can make sure your problem is fixed. Thanks :-) -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDL F10 brazilian doorbell device + TDM2400
gincantalupo wrote: Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not living there so I do not know if its a matter of signalling type or what. Is there anbody who tried this stuff or similar? It sounds like there might be an issue here related to not having disconnect supervision enabled. Can this device provide come sort of disconnect supervision? -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Tony Mountifield wrote: In article [EMAIL PROTECTED], Russell Bryant [EMAIL PROTECTED] wrote: If the TE110P will not work out for you, Digium will trade it for a TE120P. The 120 is the replacement for the 110 which uses a far superior PCI interface developed at Digium instead of the TigerJet, which has been the cause of compatability issues in the past. Very soon, the TigerJet part will no longer be in use in any of the Digium cards. Will there be non-TigerJet TE2xx and TE4xx cards that are regular PCI and not PCI Express? Those cards have never had a TigerJet on them. They have an FPGA on them, which makes it easier to make changes when there are PCI compatibility problems. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Arthur Miller wrote: The Digium cards are known to steal IRQ's. The Sangoma cards do not Not to appear defensive, but that is a technically inaccurate and also technically ambiguous statement. To correct it, there used to be a potential problem related to using the TE2xxP/TE4xxP cards relating to IRQ sharing which was fixed by a driver update. That is now resolved, and there shouldn't be any further issues. A considerable portion of the IRQ problems are an urban legend, a sort of scapegoat to point at. However, I would like to say that if anyone *does* have any problems relating to this, Digium and I personally are *very* interested in correcting them. We want to make sure that you trust our products, and want to stand behind our ability to support that. We have had some growing pains along the way, but we are *very* interested in making sure our hardware works to your and our other customers' satisfaction, and certainly stands up for itself in the face of competition. The Asterisk community is very important to us, and your perception of our products is crucial to our ability to afford to better support you and also forward the development of Asterisk. If you do have a problem, please contact technical support so that it can be fixed as soon as possible. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P (TDM22P) and aux power.
Thomas Kenyon wrote: Joe Acquisto wrote: I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to terminate POTS lines and use all SIP Phones? Yes, you only need to connect a power supply if you have FXS boards. Due to circumstances, I end up with a 1u server that has no aux power connectors available. I have to use this server, so am considering abandoning the analog phones and using all SIP. IIRC, the aux power *is* only to power ringers. I don't remember if it is also needed to provide the potential for the line as well, but I cat testify to the fact that you can comfortably run a TDM400P with 4 FXO boards on it and nothing plugged into the PSU header. That is correct. You *only* need the power connector plugged in for FXS modules. FXO modules do not need them. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztcfg error : TE110p error with CAS signalling on span 1 conflicts with HDLC with ...
Vidura Senadeera wrote: Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. cas signalling on span1 conflicts with HDLC with FCS on channel 16 It's fairly self explanatory. CAS stands for Channel Associated Signalling. That means signalling is passed on the same channel that the media is, like in robbed bit signalling protocols like FXO, FXS, EM, etc. Since you are using a PRI which does not contain inband signalling, but rather out of band signalling, you need to set it to `ccs` instead of `cas` (in your span= line) which stands for Common Channel Signalling. This is for signalling modes such as PRI or SS7 which use a dedicated channel to do call related signalling. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Dial tone came from fxs modules
Mojo with Horan Company, LLC wrote: Just to be clear, I thought that dialtone provision didn't require the power cable, just generating ring voltages? Can anyone say? The DC-DC converter on the FXS modules supplies both ringing voltage and line voltage. If the power connector is not plugged into the TDM card then the FXS module can't generate line current and the call will not be held up. (From Mickey Morris, hardware design engineer here at Digium) Matthew Fredrickson Moj Anthony Messina wrote: On Wednesday 05 September 2007 09:09:25 am wassim darwish wrote: Hi: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Can i get some help please. do you have the power cable attached to it. that's what you need to generate a dialtone. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
Steve Totaro wrote: I have had Digium tech support tell me to do the same thing I'm hoping that wasn't the final conclusion in the tech support debugging process. If it was, than I am very sorry to hear that, and will make note of it. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. Thanks, Steve G B wrote: Hi, I appreciate the help. I called the vendor of the card and they recommended removing all of the PCI cards on the system (including the video card), and moving the card to a new PCI slot. I did all of them together, ran the system headless, and ssh'ed in remotely. It worked! haha... This must be proof that I have purchased a real piece of @#$. Thanks for all of your help. Date: Sat, 8 Sep 2007 02:41:50 +0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] New Installed X100p On Fri, Sep 07, 2007 at 08:45:11AM -0700, G B wrote: Hi Tzafrir, I am not sure what to look for, so I haveattached both the contents of /var/log/kern.log as well as the outputof dmesg. If you are looking for something specific, I simply asked for a few lines around that message. Anyway, the relevant lines are: Relevant lines: [ 39.337207] Failed to initailize DAA, giving up... [ 39.337283] wcfxo: probe of :00:0c.0 failed with error -5 No more details. This may be a defective card. I have also seen some cases where some voodoo at the PCI layer was required (e.g: passing the boot option pci=noacpi). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Can you find the hidden words? Take a break and play Seekadoo! Play now! http://club.live.com/seekadoo.aspx?icid=seek_wlmailtextlink ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 56k modem configuration
Andrea Spadaccini wrote: Hello everybody, I've got a 56k usb modem, lsusb says: Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc. I'd like to let it work with Asterisk. I think that I should use chan_modem and/or chan_modem_bestdata, but I found little or no documentation. Can anybody please post some instructions? I would be very surprised if chan_modem actually works... I don't think I've *ever* seen it setup before. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
Steve Totaro wrote: Matthew Fredrickson wrote: Steve Totaro wrote: I have had Digium tech support tell me to do the same thing I'm hoping that wasn't the final conclusion in the tech support debugging process. If it was, than I am very sorry to hear that, and will make note of it. Thanks, yes, that was the final resolution. I have also heard That motherboard or that server is not supported. Again, this was quite some time ago and Digium has changed as a company as well as the product line, the whole entity has matured. Might be a non-issue now. I would hope so too as well. We're working to change a lot of things that have caused us problems in the past. Part of that problem was learning to deal with a tremendous amount of growth in a short period of time, which I would imagine is difficult for any small company. Let me ask you this, is using a T1 card for ISDN data supported now? I believe it should be working well now, a while ago I spent a bit of time making sure the zaptel portion of it functionally didn't have any problems across a range of kernels. I know that one of the reasons why that support did not support that was (IIRC) it sometimes involved recompiling a systems kernel, or upgrading a systems kernel, which is not an insignificant thing to do for a customer. Though I have not had to look at it in a while, I believe that at the very least it could be easier now, with the packaging of some of the hdlc utils in zaptel so that it works correctly across kernel versions. That one irked me since it was a selling point, but when calling for support I was told, It can do it but it is not supported. and info on the net was VERY sparse for accomplishing this (circa 2003) Sorry again about you trouble with that. I hope that somehow we can win you back :-) Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 56k modem configuration
Andrea Spadaccini wrote: Ciao Matthew, I would be very surprised if chan_modem actually works... I don't think I've *ever* seen it setup before. Well.. So there's no hope to make that modem work with Asterisk, right? Unless someone speaks otherwise, I would say that the most accurate answer is, your mileage may vary, but don't hope for a lot :-) -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P periodic sound clicks on FXS
Costa Tsaousis wrote: Hi, I am having periodic sound clicks (2-3 per second) on all FXS of a TDM400P when the remote end is my VoIP provider. However: - recording the conversation on the asterisk, does not have the glitches, although I can hear them on a real phone. - My VoIP provider to my VoIP phones through the same asterisk is OK. - TDM to TDM through the same asterisk is OK. If TDM to TDM is ok, then it would strongly point towards a problem with perhaps the VoIP provider. This is just shooting off of my hip, but maybe a jitterbuffer issue, like with the phones? I think when Asterisk bridges SIP-SIP calls, it doesn't do any jitter buffering. Matthew Fredrickson I tried with and without echocancel and different values of echotrain (including 'no'), without luck. The card is not sharing interrupts. Any ideas? Kernel is 2.6.9 asterisk is 1.2.19-BRIstuffed-0.3.0-PRE-1y-h # lspci 00:00.0 Host bridge: Intel Corporation 82945G/GZ/P/PL Memory Controller Hub (rev 02) 00:02.0 VGA compatible controller: Intel Corporation 82945G/GZ Integrated Graphics Controller (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev e1) 00:1f.0 ISA bridge: Intel Corporation 82801GB/GR (ICH7 Family) LPC Interface Bridge (rev 01) 00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 01) 00:1f.2 IDE interface: Intel Corporation 82801GB/GR/GH (ICH7 Family) SATA IDE Controller (rev 01) 00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 01) 01:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) 01:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 01:07.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) # cat /proc/interrupts CPU0 0:4262013 XT-PIC timer 1: 8 XT-PIC i8042 2: 0 XT-PIC cascade 3:4217220 XT-PIC qozap 5: 11979 XT-PIC eth0 8: 1 XT-PIC rtc 11: 29016 XT-PIC libata 15:4211433 XT-PIC wctdm NMI: 0 ERR: 0 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P periodic sound clicks on FXS
Costa Tsaousis wrote: Matthew Fredrickson wrote: Costa Tsaousis wrote: Hi, I am having periodic sound clicks (2-3 per second) on all FXS of a TDM400P when the remote end is my VoIP provider. However: - recording the conversation on the asterisk, does not have the glitches, although I can hear them on a real phone. - My VoIP provider to my VoIP phones through the same asterisk is OK. - TDM to TDM through the same asterisk is OK. If TDM to TDM is ok, then it would strongly point towards a problem with perhaps the VoIP provider. This is just shooting off of my hip, but maybe a jitterbuffer issue, like with the phones? I think when Asterisk bridges SIP-SIP calls, it doesn't do any jitter buffering. If it is a jitterbuffer, then why the recordings (of the same calls I hear the clicks, not other calls) do not have them? Well, I could be wrong since I haven't checked the code, but I believe that asterisk only enables jitterbuffering on a call if it terminates either at a non-rtp endpoint, such as a zaptel TDM interface or perhaps a recording to a file. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
shadowym wrote: Maybe his comments were taken out of context as they don't have the whole interview posted. Why is he talking about queue games, Biologicall and other extremely niche crap when there are huge holes in the basic offering (SLA and SCA)? Considering it is an open source project, anybody that has access to the source code (i.e. everybody) can work on whatever they want to, whether it be SLA, SCA, or queue games for the more light hearted. Matthew Fredrickson From: Al lists [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 11, 2007 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG) I liked the queue game concept! although it could be cruel! On 9/11/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up Seems the Adtran relationship goes way back... Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE405P intermittent yellow alarm
Richard van der Hoff wrote: Steve Totaro wrote: Richard van der Hoff wrote: [intermittent yellow alarm] At this point, I'd really like to know what a yellow alarm actually means. I've read that it indicates that that the other end of the E1 is in an alarm condition: however BT's terminating unit seems quite happy with no alarm conditions at all. Check your cabling. Replace it with new stuff. Re-punch everything. It is obviously somewhere in the line. If the above does not fix it, maybe you can get a lucky and get a good tech out that will stick around to see the issue. The only bit of cable I own here is the 2m length of cat-5 between the te405P and BT's line terminating unit. And yes, I've replaced that about 5 times now... Thanks for your help, but again I'd like to ask: what does a yellow alarm actually mean? From the driver source code I can see it is set when the FRS0 register has bit 4 set - but that doesn't help a lot... A yellow alarm means that the other end is seeing loss of signal (detected a red alarm from its perspective). When it detects LOS, it transmits yellow alarm to notify the other end. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
Scott Moseman wrote: Here's what I'm showing in the logs... [Sep 18 09:52:09] VERBOSE[2786] logger.c: == Registered file format g729, extension(s) g729 [Sep 18 09:52:09] VERBOSE[2786] logger.c: format_g729.so = (Raw G729 data) [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This module is supplied under a commercial license granted by Digium, Inc. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Please see the full license text supplied by the accompanying [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: register utility, or ask for a copy from Digium. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This product includes software developed by the OpenSSL Project [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Copyright (C) 1998-2006 The OpenSSL Project [Sep 18 09:52:09] VERBOSE[2786] logger.c: == G.729 Host-ID: x:x:x:etc [Sep 18 09:52:09] WARNING[2786] codec_g729.c: Failed to initialize G.729 copy protection! [Sep 18 09:52:09] VERBOSE[2786] logger.c: codec_g729a.so = (Annex A/B (floating point) G.729 Codec (optimized for i686)) Any ideas where this points me? I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. Matthew Fredrickson Thanks, Scott On 9/17/07, Scott Moseman [EMAIL PROTECTED] wrote: What's the best way to debug what's going on within Asterisk? I turned up the 'core debug', but that did not give me what I was hoping to find. I'm hoping to see some kind of error that explains why it will not pass through the g729 codec. Thanks, Scott On 9/14/07, Scott Moseman [EMAIL PROTECTED] wrote: I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and the phone in question. I even attempted to setup the phone with only the allow=g729, but in that instance it won't even complete the call. We had to add g711 support to the gateway in question for now to get it up and running, but we want to get it back to using only g729. CLI show modules like g729 Module Description Use Count format_g729.so Raw G729 data 0 codec_g729a.so Annex A/B (floating point) G.729 Codec ( 0 2 modules loaded I downloaded the pre-compiled g729 module from Digium. The sip.conf references g729 and the codec module is loaded. Unless there's anything else I need to do that I'm missing? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN data packets
Arpit Mehta wrote: I have a ISDN PRI(T1) line coming into my TE110 Asterisk card. When I use pri intense debug span 1 It is supposed to show every packet that is received on the PRI line. I wanted to know in ISDN Pri when a call connects how are the data (voice) packets (for PRI) shown in Asterisk. Or if there is some other command to see these kind of data packets ? pri intense debug is used to see signalling that happens on the PRI. There is not a visualizer for b channel voice data. The closest thing you could try to use is ztmonitor or a record() in your dialplan. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRI debug in Asterisk
Arpit Mehta wrote: Hi all, Does Asterisk contain a full fledged ISDN packet sniffer. By giving the command pri intense debug span 1 , does it debug every packet received (control and voice/data packets) ? No, like I said in response to your other question, the only thing you can directly see in pri intense debug is the signalling packets. Data with TDM is not packetized as its native format, so that is why there isn't a way to see tdm voice packets like you can see RTP packets. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRI debug in Asterisk
Erik Anderson wrote: On 9/18/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi all, Does Asterisk contain a full fledged ISDN packet sniffer. By giving the command pri intense debug span 1 , does it debug every packet received (control and voice/data packets) ? To get the equivalent of a packet sniffer, you'll need to go to a lower-level tool than asterisk. For sangoma cards, you can use the `wanpipemon` command to do a packet dump. I'm not sure what the equivalent for Digium cards is, but I'm sure it's possible. You can basically use ztmonitor to get a B-channel data dump. That should also work on the Sangoma cards. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN data packets
Arpit Mehta wrote: Thanks for the reply. I was not looking for a visualizer. I justed wanted to see the data packets flowing in the asterisk CLI (for example something similar to the rtp packets that flow when making a voip call). I can see the various messages like CONNECT, SETUP etc. I am a newbie regarding ISDN and I might be looking at things wrongly. Unfortunately, there isn't a way of seeing ISDN TDM data flowing into and out of asterisk like RTP. Matthew Fredrickson Thanks Regards Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: Arpit Mehta wrote: I have a ISDN PRI(T1) line coming into my TE110 Asterisk card. When I use pri intense debug span 1 It is supposed to show every packet that is received on the PRI line. I wanted to know in ISDN Pri when a call connects how are the data (voice) packets (for PRI) shown in Asterisk. Or if there is some other command to see these kind of data packets ? pri intense debug is used to see signalling that happens on the PRI. There is not a visualizer for b channel voice data. The closest thing you could try to use is ztmonitor or a record() in your dialplan. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. -- -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hfcmulti and B410P Digium Card
voip crazy wrote: Hello all, I am getting the following error in /var/log/syslog. I have got 2 B410P cards in this box. Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0153, z2=00d3) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0013, z2=0012) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0053, z2=0153) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0093, z2=0092) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=00d3, z2=0053) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0113, z2=0112) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0153, z2=00d3) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0013, z2=0012) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0053, z2=0153) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(2) reading 15 bytes (z1=004e, z2=0040) HDLC COMPLETE (f1=3, f2=2) got=15 Sep 19 17:13:31 localhost kernel: 02 01 06 06 08 01 63 5a 08 02 80 90 Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0093, z2=0092) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(2) has 383 bytes space left (z1=015d, z2=015d) sending 4 of 4 bytes HDLC I left untouched the /etc/init.d/misdn-init script to load the default values. Is needed the hfcmulti modules with this kind of cards? What is the menaing of this errors? Are something missconfigured? Unless you are having some sort of problem other than this, than I think that this is just standard debug output, which you can disable if you set the debug option in /etc/misdn-init.conf to 0. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is softswitch
Alex Balashov wrote: On Wed, 19 Sep 2007, Anthony Francis wrote: IMHO asterisk is a softswitch, it may not be a very high capacity one (right now) but it can be and if you don't mind splitting your physical trunk calls over multiple machines it works very well as a call routing engine, you just need to have carefully designed plans. It is far to easy to create call routing loops, but if you don't know what you are doing with a real telephony switch you can do the same. No SS7/ISUP support (and no TCAP, which is required for LNP and LIDB and traditional CNAM), poor/incomplete IMT support, can't take more than a few T1s per host - if that. No GR.303 support. Actually, I have been working on an SS7 stack for asterisk called libss7. SS7 support is already in trunk, and should be in the next stable release of Asterisk. Right now it only does ISUP/MTP3/MTP2, but with some work an effort, SCCP/TCAP/LNP support could be implemented as well. Asterisk has had GR.303 support for a while, though I don't think it's asymmetric (it only supports one particular function of it, or something like that IIRC). -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No audio on Zap (T1/PRI) channels
Steve Totaro wrote: Steve Edwards wrote: I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards. Each group of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback demo-congrats fine. The middle server (ts21) cannot -- just dead air. If I call via ZAP, dead air. If I call via IAX, I hear the file. I copied /etc/zaptel.conf, /etc/asterisk/*, /var/lib/asterisk/sounds/demo-congrats.gsm from ts20 to ts21 -- no joy. I have seen this in my system log file: Oct 2 18:41:49 WARNING[7477]: chan_zap.c:8087 zt_pri_error: [Span 0 D-Channel 0] PRI: !! Got reject for frame 95, but we have nothing -- resetting! I'm running asterisk-1.2.24, asterisk-addons-1.2.7, libpri-1.2.5, zaptel-1.2.20.1. show channel zap/?, zap show channel ? appear identical between working and non-working systems both on-hook and off-hook. Any clues or clues where to start looking? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Double check both zaptel.conf and zapata.conf and also call the telco to make sure they have they have the same NFAS scheme on all T1s setup correctly. Sometimes (let's face it, alot of times, the provider messes something up). Also check that all of your T1 cables are plugged into the correct T1 port. I have made that mistake myself when doing 28 T1s off a T3. I got dead air just as you described. Yes, if you are running NFAS, getting dead air on a call is a symptom of not having the logical span identifier correctly corresponding to the physical span you have plugged in (spanmap option in zapata.conf, IIRC). -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
Brian West wrote: Sangoma has contributed to Asterisk in the past and they still do. They also have contributed to Yate, FreeSWITCH and various other software that is capable of using their hardware. This argument of Digium vs Sangoma is very emotional for some. I see it as competition is good and drives innovation. Digium can't take every bit of credit for Asterisk, you have to remember the community has a large part in making Asterisk as popular as it is. I know their is hostility directed at anyone that uses non-Digium hardware by some folks and their shouldn't be. Its an open market and an open platform. Rhino makes hardware that plugs into zaptel but yet I don't see their drivers in the zaptel repo... I don't see many of the third party hardware drivers in the zaptel repo. Not to ignite any fires, but I don't think I've *ever* knowingly received a patch to libpri or chan_zap from them. And I've fixed a few protocol related bugs in libpri for people with Sangoma cards. It'd be nice if they at the very least supported the protocol stacks and zaptel channel driver they use to make money off their cards. Matthew Fredrickson /b On Oct 5, 2007, at 7:51 AM, Steve Murphy wrote: Oh, Julian, I'd imagine what I'm about to say will fuel some flames! Here's a fairly powerful argument for all you asterisk users, as to why you should purchase a Digium product vs. a Sangoma: Because Digium uses a chunk of the purchase money to support Asterisk. And Sangoma DOES NOT. Digium employs several developers specifically to maintain and improve Asterisk. Sangoma DOES NOT. While they may maintain and improve their own versions of the various drivers, THEY DO NOT SHARE THEIR SOFTWARE. Matt F. told me last week we haven't seen ANYTHING from them for a LONG TIME, with respect to the zaptel drivers. If they have been contributing patches, they are disguising their association with Sangoma. Don't get me wrong. I AM a Digium employee! A software Developer to be specific, an Asterisk developer to be precise. So, I AM highly biased towards Digium! Digium has a harder job than Sangoma with respect to Asterisk. While Digium takes a chunk of its revenue, and uses it to maintain and improve Asterisk (not just the drivers), Sangoma doesn't, and it gives them a competitive edge. So, for all you folks who have bought Digium, I personally thank you! You have helped Asterisk, and you have personally helped ME. If you have long-range business or interest in Asterisk, you are indirectly contributing to its growth and improvement when you buy Digium products services. murf -- Steve Murphy Software Developer Digium ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
Brian West wrote: I think the horse has been long dead! /b Yeah, and while we're on such things, I think that vi beats the pants off of emacs :-) -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] really sorry about this - E1 vs T1
Julian Lyndon-Smith wrote: I am *really* sorry about hijacking this thread, but the only way I can post to the -user list is by replying to another thread. (btw, this is getting really annoying. Please, Digium, sort the filters out!) I installed my super-duper new TE412P card today, without remembering to check the settings for T1/E1. As the server is now a hundred miles away, is there a) Any way of checking what setting is in place b) Changing that setting without having to physically remove the card and see ? Yes, see the t1e1override module parameter in wct4xxp/base.c. IIRC, it's 0xff to hard code to E1 mode, and set it to 0 for T1 mode. -1 is to use the jumper settings. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] really sorry about this - E1 vs T1
Andreas van dem Helge wrote: On 10/11/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: Yes, see the t1e1override module parameter in wct4xxp/base.c. IIRC, it's 0xff to hard code to E1 mode, and set it to 0 for T1 mode. -1 is to use the jumper settings. Seems like a bad design. Why not just make it a software choice?? That is a software choice. So you can either use the jumpers, or your can override it in software. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First Time T1 Questions
[EMAIL PROTECTED] wrote: On 10/19/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: In addition to my below question, i was wondering if anyone had a problem with installing zaptel on debian sarge. its a udev problem, make install thinks i am running udev, but when i fix the makefile to be like 1.4.4 which works, when i load ztcfg it still says 1.4.4. so something is not right... Not sure what to tell you but certainly it works without problems in CentOS/RHEL SuSE Linux. About the cards personally I like the sangoma cards. As you can see they have a better warranty than the digium cards. Also I feel they aren't as tied to a platform (Asterisk) as the Digium cards are. And some people claim some Digium cards have IRQ issues or problems with certain big-name server components (mainboards mainly) of which I haven't heard similar complaints for the Sangoma cards. I know I've said this time and time again, but just for the purpose that this will be archived somewhere on the net, there should not be any more problems related to interrupts and specific servers. If there are, *please* let me know so that we can fix it. We have spent much of the last year or so getting rid of these problems, and we are very much committed to having 100% compatibility, and getting rid of our former reputation of having IRQ/motherboard problems. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX
[EMAIL PROTECTED] wrote: Hi, I'm running some Asterisk-machines, and on one of them i get this errors in the CLI, but i don't know what that means. Hardware: Digium 4-Port E1 Card with HWEC Intel Pentium D 3 GHz 2 GB RAM SATA Harddisk Supermicro Mainboard Software: latest libpri/zaptel/asterisk of version 1.2 I tried also asterisk version 1.4.x, but there the problem occurs every 10 calls, on asterisk 1.2 its about every 100 calls. Did this recently start, like after you upgraded or is this something that has always been a problem for you since you installed? If it has always been a problem, can you post a `pri debug span x` trace of a call when this happens? That will help to know more about what is going on here. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First Time T1 Questions
Michael J. Liberatore wrote: Well this is the bug I am having with the make install of 1.4.5.1: http://bugs.digium.com/view.php?id=10156 Even though I got it to install ztcfg -vvv still says 1.4.4 also. Mike We just made a new zaptel release (1.4.6) in which there were many fixes. Tzafrir (from Xorcom) made a significant number of Makefile changes between 1.4.4 and 1.4.5 (and 1.4.5.1 too) that may or may not have introduced this problem. Please retest with latest zaptel and update the bugnote so that we know if this problem has been fixed. Matthew Fredrickson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Friday, October 19, 2007 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] First Time T1 Questions [EMAIL PROTECTED] wrote: On 10/19/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: In addition to my below question, i was wondering if anyone had a problem with installing zaptel on debian sarge. its a udev problem, make install thinks i am running udev, but when i fix the makefile to be like 1.4.4 which works, when i load ztcfg it still says 1.4.4. so something is not right... Not sure what to tell you but certainly it works without problems in CentOS/RHEL SuSE Linux. About the cards personally I like the sangoma cards. As you can see they have a better warranty than the digium cards. Also I feel they aren't as tied to a platform (Asterisk) as the Digium cards are. And some people claim some Digium cards have IRQ issues or problems with certain big-name server components (mainboards mainly) of which I haven't heard similar complaints for the Sangoma cards. I know I've said this time and time again, but just for the purpose that this will be archived somewhere on the net, there should not be any more problems related to interrupts and specific servers. If there are, *please* let me know so that we can fix it. We have spent much of the last year or so getting rid of these problems, and we are very much committed to having 100% compatibility, and getting rid of our former reputation of having IRQ/motherboard problems. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First Time T1 Questions
[EMAIL PROTECTED] wrote: On 10/19/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I know I've said this time and time again, but just for the purpose that this will be archived somewhere on the net, there should not be any more problems related to interrupts and specific servers. If there are, *please* let me know so that we can fix it. We have spent much of the last year or so getting rid of these problems, and we are very much committed to having 100% compatibility, and getting rid of our former reputation of having IRQ/motherboard problems. Unless you've updated the hardware and replaced the cards (even those already out of warranty) I don't understand how a software patch is going to fix hardware compatibility problems that have come up time and time again relating to the Digium cards. There have been different parts to this work, hardware as well as software. Both have been updated and improved for compatibility and increased performance. I know there have been countless (never really understood why) hardware revisions/models and I wouldn't doubt the newest ones have resolved most of the compatibility issues, but I never heard of any recall program for the older cards. As I already mentioned above, if you have any compatibility problems with a card, please notify technical support so that we can get it resolved. If replacement is needed, it will be part of the technical support process. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX
[EMAIL PROTECTED] wrote: On this machine its the first install, but i get this error 3 month before on an other machine also. I think the debug will bring t much data, cause there is any half second a call try, and its really hard to find this error in the debug file. The only thing i know is if i use a Sangoma card, the problem went. Can i send you a BIG debug file where some of this errors happened? If you can post a debug of a call when it happened and a call when it doesn't happen, that would help the most. Thanks Nico On Fri, 19 Oct 2007, Matthew Fredrickson wrote: [EMAIL PROTECTED] wrote: Hi, I'm running some Asterisk-machines, and on one of them i get this errors in the CLI, but i don't know what that means. Hardware: Digium 4-Port E1 Card with HWEC Intel Pentium D 3 GHz 2 GB RAM SATA Harddisk Supermicro Mainboard Software: latest libpri/zaptel/asterisk of version 1.2 I tried also asterisk version 1.4.x, but there the problem occurs every 10 calls, on asterisk 1.2 its about every 100 calls. Did this recently start, like after you upgraded or is this something that has always been a problem for you since you installed? If it has always been a problem, can you post a `pri debug span x` trace of a call when this happens? That will help to know more about what is going on here. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] White noise from TDM2400
Stephen Kratzer wrote: Hello. We recently replaced a channel bank in favor of a TDM2400E. After doing so, users began complaining that they could barely hear the remote parties. We increased gain appropriately for each channel which increased the volume of the voices but has also increased the volume of any line noise. It sounds like white noise which goes away when either party talks and returns during silence. Is there any remedy to this? Thanks. Do you have the new TDM2400E with the VPMADT032 on it? Also, what version of zaptel are you running? -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card
Joseph Begumisa wrote: Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950? I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. The Dell hardware owners manual states that it means the system BIOS has reported a PCI parity error on a component that resides in PCI configuration space at bus 0, device 4, function 0 and advises that the PCI expansion card be removed and reseated. Any suggestions on what exactly might be causing this are welcome. This sounds like something worthy of notifying tech support of. Can you try contacting them so that they can further diagnose this problem? Thanks. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card
Steve Totaro wrote: Joseph Begumisa wrote: Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950? I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. The Dell hardware owners manual states that it means the system BIOS has reported a PCI parity error on a component that resides in PCI configuration space at bus 0, device 4, function 0 and advises that the PCI expansion card be removed and reseated. Any suggestions on what exactly might be causing this are welcome. Thanks. Joseph My guess would be that the Digium card is causing the issue although you would probably be led to believe that the Dell is not compatible with the card and not visa versa. It would be interesting to see if a Sangoma board would have that same issue. I have not had any of these compatibility issues since going Sangoma. Is this an older card or one with the New and Improved Bus thing? The TE110P has the old style PCI interface on it. The TE120 is the newer one based on Voicebus. In any case, he should contact tech support about this so we can resolve it. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE210P issues
Jerry Geis wrote: I have a box with a TE210P. Things work for a while then stop when making call files. I get NOANSWER as the return code (right away). I am running asterisk 1.2.12.1, libpri 1.2.3 and zap 1.2.9.1 When I try to update to newer zaptel the machine locks when loading the zaptel drivers. I tried to manually load the wct1xxp module (I think that is the one for the dual T1 card???) and the machine locks. I am in a remote location so I cannot see if anything is on the console. I tried jumping to 1.4 and the same thing happens. I have updated quite a few asterisk boxes remotely and never had this issue before. Last thing I tried was chkconfig zaptel off, reboot, then try loading in new version and the same thing happened. It locked up. After rebooting I put back the old zaptel and it works again for awhile. What shall I try? Could you contact tech support about this? When you purchased your card, you also purchased support for issues like this. And please give us a chance to diagnose and fix this problem. I suspect that they will be able to resolve this. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE210P issues
Steve Totaro wrote: Calling Digium. Post your /var/log/messages and /var/log/asterisk/full (just anything that looks relevant). Try a Sangoma card. Or better yet, give us an opportunity to fix it. Sangoma cards have problems too and I'm sure they have been going through a trial of fire trying to eliminate compatibility-type problems with their boards, but when you have a problem, you still have to give them (and us) a chance to fix it. It's the nature of the PC world, there are a lot of different platforms to interoperate with. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compatibility Issues with dell poweredge 195 and TE110P card
Brian Hutchinson wrote: You guys are scaring me! I'm building a 2950 with SAS RAID 5 on the new PERC and it will have 2 TE420P's. I hope it works or my bacon will fry. You shouldn't see any problems with those boards. The 2950 is a common environment. If I remember correctly, there used to be a problem (and I think it was localized to the TE110P as well, IIRC), but it was fixed a while ago. On 10/25/07, Joseph Begumisa [EMAIL PROTECTED] wrote: Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950?I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. Yes, I have had this problem with a dell PE1650, 1850, SC1400, and PE650. I have a TE410P that does it. It may not be wise, but I just ignore the orange blinking LCD display (or light, depending on the model). I did try reseating the card, and it works for a few weeks, and then goes back to the same old thing. Yes, that happened too. Digium has graciously offered to send me a TE120P with the Digium VoiceBus technology which I will test out on the Dell 1950. Will post my findings thereafter. Joseph. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI span configuration - span remains down
Rony Ron wrote: Hello, Quoting Digium Support: The TE110P has been discontinued and replaced in our product lineup with the TE120P, which features many overall improvements and does not suffer from the HDLC Abort/Bad FCS problems that the TE110P did. Although this is true ( :-) ) I think that it is likely his problem is not related to this. Can you post a pri intense debug span x for the span in question? Matthew Fredrickson On 10/25/07, David Kennedy [EMAIL PROTECTED] wrote: Hi, I'm trying to connect to Telewest/Virgin Media with a TE110P using asterisk 1.4.13/zaptel 1.4.6. No matter what I try, my span always appears as PRI span 1/0: Provisioned, Down, Active My zapata.conf is currently --- [channels] echocancel=yes echocancelwhenbridged=no echotraining=yes switchtype=euroisdn contect=from-pri signalling=pri_cpe group=1 channel = 1-15 channel = 17-31 --- zaptel.conf is --- span=1,1,0,ccs,hdb3,crc4 dchan=16 bchan=1-15,17-31 loadzone=uk defaultzone=uk --- I'm in London and the server is in Manchester, so I can't look at the server directly, but when we first started setting it up, apparently a pair of cables were the wrong way round, so the card was in a RED alarm state. We've switched the cables and now the card is OK. We did have a lot of IRQ misses, so we've upgraded the kernel and now the accuracy reported by zttest is about 99.98%. Telewest have checked the line for faults and have reported that it's fine, but I just can't get it working. Does anyone have any ideas/suggestions? Thanks, Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to dial out over Zap - span 1 got hangup, cause 44
David Kennedy wrote: Hi I posted earlier about having issues connecting to Telewest's ISDN, only to find out later Telewest had forgotten to turn it on - hopefully I'm not having a similar silly problem. My PRI span is now up and operational, but when I try to send a call out over it, I just get congestion tones. Occasionally, I get about one second of ring tones, only for it to cut out and play congestion. Here's a bit of output (I've taken out the phone number) -- Executing [EMAIL PROTECTED]:6] Dial(SIP/charlie59-082bc890, Zap/my phone number|3600) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/my phone number -- Channel 0/4, span 1 got hangup, cause 44 -- Forcing restart of channel 0/4 on span 1 since channel reported in use -- Hungup 'Zap/4-1' [Oct 25 17:35:22] NOTICE[20503]: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/4-1' not posted == Everyone is busy/congested at this time (1:0/0/1) Additionally, once a zap channel has been used like this, it seems to end up in stuck in this state: PRI Flags: Resetting Previously, someone mentioned that the TE110P card installed had a few issues and I should be using a TE120P instead - could that be the cause? If your span is up ok, and you are actually getting a valid cause code back (as you mentioned) your card should be just fine. It sounds like protocol related problems. Are you sure you are sending the correct digit format out on the line? PRIs can be very picky about it. Some like the area code, some don't, and a number of other things. Also, can you get an inbound call on the PRI? That's usually the easiest first case to get working. From looking at the specs, it looks like 44 is cause requested channel unavailable. Maybe they haven't unbusied the channels yet or something like that. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to dial out over Zap - span 1 got hangup, cause 44
David Kennedy wrote: Hi I posted earlier about having issues connecting to Telewest's ISDN, only to find out later Telewest had forgotten to turn it on - hopefully I'm not having a similar silly problem. My PRI span is now up and operational, but when I try to send a call out over it, I just get congestion tones. Occasionally, I get about one second of ring tones, only for it to cut out and play congestion. Here's a bit of output (I've taken out the phone number) -- Executing [EMAIL PROTECTED]:6] Dial(SIP/charlie59-082bc890, Zap/my phone number|3600) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/my phone number -- Channel 0/4, span 1 got hangup, cause 44 -- Forcing restart of channel 0/4 on span 1 since channel reported in use -- Hungup 'Zap/4-1' [Oct 25 17:35:22] NOTICE[20503]: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/4-1' not posted == Everyone is busy/congested at this time (1:0/0/1) Additionally, once a zap channel has been used like this, it seems to end up in stuck in this state: PRI Flags: Resetting Previously, someone mentioned that the TE110P card installed had a few issues and I should be using a TE120P instead - could that be the cause? Oh yeah, and could you also post a pri debug span x of the call as well? That should tell a lot too. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI span configuration - span remains down
David Kennedy wrote: Hi While I have fixed the problem from this post, I do have another problem, and you have asked for a debug output here, so I'll go against my better instinct and reply here :) I just looked through your debug and can't see any obvious problems. It's likely you'll need to ask your telco why the other switch is complaining about the channel selection. Matthew Fredrickson -- Making new call for cr 32774 -- Requested transfer capability: 0x00 - SPEECH [ 00 01 0e 06 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 007 0: 0 N(R): 003 P: 0 44 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=44 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 86] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 6 ] [6c 0c 21 83 38 34 35 38 39 39 31 30 30 31] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '8458991001' ] [70 0c 80 30 32 30 38 36 35 39 32 32 39 31] Called Number (len=14) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) 'My Phone Number' ] [a1] Sending Complete (len= 1) q931.c:2881 q931_setup: call 32774 on channel 6 enters state 1 (Call Initiated) -- Called g0/My Phone Number -- T200 counter expired, What to do... -- Retransmitting 48 bytes voip1*CLI [ 00 01 0e 07 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ] voip1*CLI Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 007 0: 0 N(R): 003 P: 1 44 bytes of data -- Rescheduling retransmission (1) voip1*CLI [ 00 01 01 11 ] voip1*CLI Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 008 P/F: 1 0 bytes of data -- ACKing all packets from 6 to (but not including) 8 -- ACKing packet 7, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter voip1*CLI [ 02 01 06 10 08 02 80 06 5a 08 03 82 ac 18 ] voip1*CLI Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 003 0: 0 N(R): 008 P: 0 10 bytes of data -- ACKing all packets from 7 to (but not including) 8 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: RELEASE COMPLETE (90) [08 03 82 ac 18] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Requested channel not available (44), class = Network Congestion (resource unavailable) (2) ] Cause data 1: 18 (24) -- Processing IE 8 (cs0, Cause) q931.c:3503 q931_receive: call 32774 on channel 6 enters state 0 (Null) Sending Receiver Ready (4) voip1*CLI [ 02 01 01 08 ] voip1*CLI Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 004 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter -- Channel 0/6, span 1 got hangup, cause 44 -- Forcing restart of channel 0/6 on span 1 since channel reported in use voip1*CLI [ 00 01 10 08 08 02 00 00 46 18 03 a9 83 86 79 01 80 ] voip1*CLI Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 008 0: 0 N(R): 004 P: 0 13 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 86] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 6 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1
Re: [asterisk-users] PRI span configuration - span remains down
David Kennedy wrote: Is there some part of the debug output I need to tell the telco about? When I was on to them earlier today, the engineer only seemed to know how to turn bits of their network on and off, not much about settings I need my end etc. Just tell them when you try to make a call, you get cause code 44 back (channel unavailable). They can look at their switch to figure out what's going on. Matthew Fredrickson Dave On 10/25/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: David Kennedy wrote: Hi While I have fixed the problem from this post, I do have another problem, and you have asked for a debug output here, so I'll go against my better instinct and reply here :) I just looked through your debug and can't see any obvious problems. It's likely you'll need to ask your telco why the other switch is complaining about the channel selection. Matthew Fredrickson -- Making new call for cr 32774 -- Requested transfer capability: 0x00 - SPEECH [ 00 01 0e 06 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 007 0: 0 N(R): 003 P: 0 44 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=44 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 86] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 6 ] [6c 0c 21 83 38 34 35 38 39 39 31 30 30 31] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '8458991001' ] [70 0c 80 30 32 30 38 36 35 39 32 32 39 31] Called Number (len=14) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) 'My Phone Number' ] [a1] Sending Complete (len= 1) q931.c:2881 q931_setup: call 32774 on channel 6 enters state 1 (Call Initiated) -- Called g0/My Phone Number -- T200 counter expired, What to do... -- Retransmitting 48 bytes voip1*CLI [ 00 01 0e 07 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ] voip1*CLI Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 007 0: 0 N(R): 003 P: 1 44 bytes of data -- Rescheduling retransmission (1) voip1*CLI [ 00 01 01 11 ] voip1*CLI Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 008 P/F: 1 0 bytes of data -- ACKing all packets from 6 to (but not including) 8 -- ACKing packet 7, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter voip1*CLI [ 02 01 06 10 08 02 80 06 5a 08 03 82 ac 18 ] voip1*CLI Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 003 0: 0 N(R): 008 P: 0 10 bytes of data -- ACKing all packets from 7 to (but not including) 8 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: RELEASE COMPLETE (90) [08 03 82 ac 18] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Requested channel not available (44), class = Network Congestion (resource unavailable) (2) ] Cause data 1: 18 (24) -- Processing IE 8 (cs0, Cause) q931.c:3503 q931_receive: call 32774 on channel 6 enters state 0 (Null) Sending Receiver Ready (4) voip1*CLI [ 02 01 01 08 ] voip1*CLI Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 004 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter -- Channel 0/6, span 1 got hangup, cause 44 -- Forcing restart of channel 0/6 on span 1 since channel reported in use voip1*CLI [ 00 01 10 08 08 02 00 00 46 18 03 a9 83 86 79 01 80 ] voip1*CLI Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 008 0: 0 N(R): 004 P: 0 13 bytes of data
Re: [asterisk-users] PRI span configuration - span remains down
: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 004 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter -- Channel 0/6, span 1 got hangup, cause 44 -- Forcing restart of channel 0/6 on span 1 since channel reported in use voip1*CLI [ 00 01 10 08 08 02 00 00 46 18 03 a9 83 86 79 01 80 ] voip1*CLI Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 008 0: 0 N(R): 004 P: 0 13 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 86] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 6 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] voip1*CLI [ 00 01 01 12 ] voip1*CLI Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 009 P/F: 0 0 bytes of data -- ACKing all packets from 7 to (but not including) 9 -- ACKing packet 8, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Restarting T203 counter NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/6-1' [Oct 25 18:01:46] NOTICE[20956]: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/6-1' not posted == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:7] ResetCDR(SIP/charlie59-082bc890, w) in new stack -- Executing [EMAIL PROTECTED]:8] NoCDR(SIP/charlie59-082bc890, ) in new stack -- Executing [EMAIL PROTECTED]:9] Answer(SIP/charlie59-082bc890, ) in new stack -- Executing [EMAIL PROTECTED]:10] PlayTones(SIP/charlie59-082bc890, congestion) in new stack == Auto fallthrough, channel 'SIP/charlie59-082bc890' status is 'CHANUNAVAIL' -- Executing [EMAIL PROTECTED]:1] Hangup(SIP/charlie59-082bc890, ) in new stack == Spawn extension (route-ext-ycmcr, h, 1) exited non-zero on 'SIP/charlie59-082bc890' As I say, I've asked a separate question on this, so I don't really want to end up with two thread on the one problem :) Thanks Dave On 10/25/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: Rony Ron wrote: Hello, Quoting Digium Support: The TE110P has been discontinued and replaced in our product lineup with the TE120P, which features many overall improvements and does not suffer from the HDLC Abort/Bad FCS problems that the TE110P did. Although this is true ( :-) ) I think that it is likely his problem is not related to this. Can you post a pri intense debug span x for the span in question? Matthew Fredrickson On 10/25/07, David Kennedy [EMAIL PROTECTED] wrote: Hi, I'm trying to connect to Telewest/Virgin Media with a TE110P using asterisk 1.4.13/zaptel 1.4.6. No matter what I try, my span always appears as PRI span 1/0: Provisioned, Down, Active My zapata.conf is currently --- [channels] echocancel=yes echocancelwhenbridged=no echotraining=yes switchtype=euroisdn contect=from-pri signalling=pri_cpe group=1 channel = 1-15 channel = 17-31 --- zaptel.conf is --- span=1,1,0,ccs,hdb3,crc4 dchan=16 bchan=1-15,17-31 loadzone=uk defaultzone=uk --- I'm in London and the server is in Manchester, so I can't look at the server directly, but when we first started setting it up, apparently a pair of cables were the wrong way round, so the card was in a RED alarm state. We've switched the cables and now the card is OK. We did have a lot of IRQ misses, so we've upgraded the kernel and now the accuracy reported by zttest is about 99.98%. Telewest have checked the line for faults and have reported that it's fine, but I just can't get it working. Does anyone have any ideas/suggestions? Thanks, Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc
Re: [asterisk-users] PRI span configuration - span remains down
Rony Ron wrote: Hi, in meantime if you have another type of digium pri card you can plug it into your box to confirm that it's not related to that card! Better eliminate any doubt about that card... it made me suffer ! Well, if signalling didn't work on the D-channel, that might be a more plausible option. When the D-channel comes up, you've (for 99.% of cases) usually eliminated the card being a problem. It looks like his D-channel is up, if he's passing call signalling data back and forth like this. Matthew Fredrickson BR, On 10/25/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: David Kennedy wrote: Hi While I have fixed the problem from this post, I do have another problem, and you have asked for a debug output here, so I'll go against my better instinct and reply here :) I just looked through your debug and can't see any obvious problems. It's likely you'll need to ask your telco why the other switch is complaining about the channel selection. Matthew Fredrickson -- Making new call for cr 32774 -- Requested transfer capability: 0x00 - SPEECH [ 00 01 0e 06 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 007 0: 0 N(R): 003 P: 0 44 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=44 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 86] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 6 ] [6c 0c 21 83 38 34 35 38 39 39 31 30 30 31] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '8458991001' ] [70 0c 80 30 32 30 38 36 35 39 32 32 39 31] Called Number (len=14) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) 'My Phone Number' ] [a1] Sending Complete (len= 1) q931.c:2881 q931_setup: call 32774 on channel 6 enters state 1 (Call Initiated) -- Called g0/My Phone Number -- T200 counter expired, What to do... -- Retransmitting 48 bytes voip1*CLI [ 00 01 0e 07 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ] voip1*CLI Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 007 0: 0 N(R): 003 P: 1 44 bytes of data -- Rescheduling retransmission (1) voip1*CLI [ 00 01 01 11 ] voip1*CLI Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 008 P/F: 1 0 bytes of data -- ACKing all packets from 6 to (but not including) 8 -- ACKing packet 7, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter voip1*CLI [ 02 01 06 10 08 02 80 06 5a 08 03 82 ac 18 ] voip1*CLI Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 003 0: 0 N(R): 008 P: 0 10 bytes of data -- ACKing all packets from 7 to (but not including) 8 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: RELEASE COMPLETE (90) [08 03 82 ac 18] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Requested channel not available (44), class = Network Congestion (resource unavailable) (2) ] Cause data 1: 18 (24) -- Processing IE 8 (cs0, Cause) q931.c:3503 q931_receive: call 32774 on channel 6 enters state 0 (Null) Sending Receiver Ready (4) voip1*CLI [ 02 01 01 08 ] voip1*CLI Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 004 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter -- Channel 0/6, span 1 got hangup, cause 44 -- Forcing restart of channel 0/6 on span 1 since channel reported in use voip1*CLI [ 00 01 10 08 08 02 00 00 46 18 03 a9 83 86 79 01 80 ] voip1*CLI Informational frame: SAPI
Re: [Asterisk-Users] Looking for Q.Sig success story
On Jan 24, 2006, at 2:05 PM, Patrick Zwahlen wrote: Hi all, Did anyone had success with Q.Sig on * 1.2, especially with Alcatel 4400 (which seems to only support Q.Sig) ? I am thinking about interconnecting 15 sites together with asterisk (probably using IAX or SIP). I have a very heterogeneous environement using both PRI and BRI, but my pilot will start with the Alcatel at the central site. Any help/example is most welcome. You should be able to make/receive calls just fine. It's just we don't support some of the more cooler supplementary features (most of the reason people like to use Q.SIG) though. Matthew Fredrickson ___ --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] How many TDM2400P's will a server take?
On Jan 30, 2006, at 4:16 PM, James Harper wrote: Juan Carlos Castro y Castro wrote: How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Depends on the specs of the server. For example, a quad Xeon will be able to service many more interrupts/card/channels than a 500 mHz Celeron. :-) Given that one TDM2400P (or even the old 4 port one) generates 1000 interrupts/second, do two cards together have to generate 2000 interrupts/second? Is there, or could there be, a way to synchronise them so that both cards can be serviced by the one interrupt. Or is it more the work that needs to be done per interrupt rather than the number of interrupts that is the problem? Exactly... you're getting it. Doing 1000 things a second is not a lot of things to do for a processor that's clocked at 2,000,000,000 hertz (2 billion somethings per second :-) ). It's more of what has to occur during the interrupt handler that causes problems. The TDM2400P busmasters just about everything (including commands to registers and such) so it doesn't have to spend a lot of time in the interrupt handler waiting for PCI accesses. Your biggest problem that you'll probably worry about is power consumption and heat generation in worst case ringing scenario, as someone else mentioned. Matthew Fredrickson ___ --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] TE411P Really Bad Echo
On Feb 5, 2006, at 9:36 PM, Stagg Shelton wrote: I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are only using the 1st span of this card right now, and we have a tdm400p with 4 fxs modules installed as well. If anyone has experience with this card, can you tell me if I am missing something. 1 to 2 seconds?! That's ridiculously huge. I don't think you'll find a echo canceler anywhere that can fix your echo problem. If it gets better with the VPM disabled, then definitely contact Digium tech-support about it. Matthew Fredrickson ___ --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] TE405p -- loopback for the phone company?
On Feb 6, 2006, at 3:57 PM, Tim Connolly wrote: I wonder if Digium has any intentions of fixing this. I brought this to their attention shortly after purchasing a pair of TE411's. You can issue a loopup on span 2 only to get a message saying looping span1 which is to say, a bit scary when you only have two active PRI and one is already down for testing... This is fixed in trunk. It was but only recently implemented. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] more cpu intensive echo cancellers ?
On Feb 8, 2006, at 1:03 PM, Gerard Saraber wrote: Hi, I've had some decent luck with the mark3 echo canceller from the zaptel driver, echos on about 20% of the calls, people I've called say I sound great now, but our side hears echos. I was wondering if there was any way to tweak the current software cancelers into using more CPU (and hopefully doing a better job, close to a hardware canceler), I only have 10 lines, and a single call takes 0.5% cpu, I would have no problem if it went up to 5-10% if they would work better. Or should I just give up now and buy the channel bank, tellabs hardware echo canceler and a T1 pci card? (hope TDM400P cards have decent resale value ;) Yeah there is, upgrade to trunk and use the new echo canceller there (MG2). It's supposed to rock, at least from what I've heard. All the MEC cancellers are _OLD_. At least switch to 1.2 and the KB1 echo canceler before giving up. --- Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] more cpu intensive echo cancellers ?
On Feb 9, 2006, at 10:50 AM, Gerard Saraber wrote: Aha! good to know, I am running asterisk 1.2.4 and zaptel-1.2.3, should I switch to CVS ? I've tried the MG2 canceler with the above versions, each time I tried it, I had a constant echo, where with the mark3 it went away after a second or two at the beginning of the call. (which I can live with, but some of the calls are completely unusable due to continuous or returning echos) I'll go play with the mg2 and kb1 again and see what happens Try MG2 with trunk and KB1 with 1.2. KB1 is supposed to be fairly reliable in 1.2, and MG2 in trunk has a good possibility of outperforming KB1 from 1.2. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] more cpu intensive echo cancellers ?
On Feb 10, 2006, at 1:21 PM, Gerard Saraber wrote: Found it, going to go test it right now :) thanks! So far reports have been positive on the echo, but its a slow day ;) We're using cisco 7960 phones, they're pricy, but they work great and sound good, if it wasn't for the echo issue, I would have been able to roll the whole setup out already. Actually that's not quite true, I still have to make the 7914 addon module work with the 7960 phone, but that's not a show stopper. Either way, so far big thumbs up for the MG2 echo can, and if any developers read this, feel free to add a compile flag to make it more cpu intensive ;) and do more canceling. Does latest MG2 behave better than KB1 on your analog lines? I heard in the past that in some cases (primarily with analog lines) that KB1 worked better. Also, have you tried the echotraining=800 (in zapata.conf) tweak as well? --- Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] more cpu intensive echo cancellers ?
On Feb 10, 2006, at 10:25 PM, Steve Underwood wrote: Matthew Fredrickson wrote: On Feb 10, 2006, at 1:21 PM, Gerard Saraber wrote: Found it, going to go test it right now :) thanks! So far reports have been positive on the echo, but its a slow day ;) We're using cisco 7960 phones, they're pricy, but they work great and sound good, if it wasn't for the echo issue, I would have been able to roll the whole setup out already. Actually that's not quite true, I still have to make the 7914 addon module work with the 7960 phone, but that's not a show stopper. Either way, so far big thumbs up for the MG2 echo can, and if any developers read this, feel free to add a compile flag to make it more cpu intensive ;) and do more canceling. Does latest MG2 behave better than KB1 on your analog lines? I heard in the past that in some cases (primarily with analog lines) that KB1 worked better. Also, have you tried the echotraining=800 (in zapata.conf) tweak as well? A lot of the variability is probably due to thr lack of a DC blocker at the front of the echo canceller. As far as I remember, none of the cancellers in * has a DC blocker. Where can one find out more information on writing a DC blocker? I google'd around a bit, but couldn't find a definitive overview of what one was, and how to write one. Thanks! --- Matthew Fredrickson ___ --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] TE411P Really Bad Echo
On Feb 12, 2006, at 6:25 PM, Mike Pollitt wrote: Hi Rob –  Is it possible to disable the onboard echo canceller so that one may try the software cancellers instead?  I have the TE110P and am experiencing the same bad echo problems that I can’t seem to effect by fiddling with the echo canceller settings in zconfig.h  Cheers, Mike.  The TE110P doesn't have an onboard echo canceler; ere go, you can try whatever options you want and they should work. Matthew Fredrickson ___ --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] Problem: Digium TDM400 with XOptionsFlex - Solved
Thomas Klettke wrote: On Sat, 2008-03-22 at 12:48 -0400, John Novack wrote: Assuming you have also checked the obvious possible defects regarding cords from the XO device to the Digium card, what happens if you reverse tip and ring? John, you were right on the money: I've found that the two lines that gave me problems had the polarity reversed. Correcting it solved the problem. I wish I had checked that last week - before spending hours on troubleshooting ... Not certain even if the Digium FXO circuit is even sensitive to line polarity, Apparently it is - unlike the Sangoma A200 which worked with either polarity. Thanks for your help I can't say how much I appreciate it. Let me know if you're ever in the Houston area: I'll buy you a beer, or two ;-) Cheers, Thomas John Novack Just to let you guys know, we're looking into this to see why this might be happening. We'll keep you posted when we find out what's wrong. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem about voice when using TDM2400p with VPMADT032 echo canceller module
Vu AnhTuan wrote: hi you, I'm having problem with voice quality on my trixbox using TDM2400B.The trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'background crackle/buzz' coming back when they talk. anyone have the same problem? pls help me. thanks a lot. my trixbox and config file: trixbox version 2.4 (Linux kernel 2.6.18, Zaptel 1.4.7) This is definitely a technical support issue. Please contact them about this so that we can help you get it resolved as soon as possible :-) ! Matthew Fredrickson Digium, Inc. zaptel.conf # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0 Wildcard TDM2400P Board 1 fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 fxsks=9 fxsks=10 fxsks=11 fxsks=12 fxsks=13 fxsks=14 fxsks=15 fxsks=16 fxsks=17 fxsks=18 fxsks=19 fxsks=20 # channel 21, WCTDM, no module. # channel 22, WCTDM, no module. # channel 23, WCTDM, no module. # channel 24, WCTDM, no module. # Global data loadzone = us defaultzone = us zapata.conf -- ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;default ;echotraining=800 ;default rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=0 relaxdtmf=yes ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-channels.conf group=1 ;Include AMP configs #include zapata_additional.conf zapata_additional.conf --- ; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; ; Span 1: WCTDM/0 Wildcard TDM2400P Board 1 ;;; line=1 WCTDM/0/0 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 1 context=default ;;; line=2 WCTDM/0/1 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 2 context=default ;;; line=3 WCTDM/0/2 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 3 context=default ;;; line=4 WCTDM/0/3 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 4 context=default ;;; line=5 WCTDM/0/4 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 5 context=default ;;; line=6 WCTDM/0/5 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 6 context=default ;;; line=7 WCTDM/0/6 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 7 context=default ;;; line=8 WCTDM/0/7 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 8 context=default ;;; line=9 WCTDM/0/8 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 9 context=default ;;; line=10 WCTDM/0/9 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 10 context=default ...more... [IP-PBX ~]# ztcfg -vv -- Zaptel Version: 1.4.7-3259 Echo Canceller: OSLEC Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Channel 05: FXS Kewlstart (Default) (Slaves: 05) Channel 06: FXS Kewlstart (Default) (Slaves: 06) Channel 07: FXS Kewlstart (Default) (Slaves: 07) Channel 08: FXS Kewlstart (Default) (Slaves: 08) Channel 09: FXS Kewlstart (Default) (Slaves: 09) Channel 10: FXS Kewlstart (Default) (Slaves: 10) Channel 11: FXS Kewlstart (Default) (Slaves: 11) Channel 12: FXS Kewlstart (Default) (Slaves: 12) Channel 13: FXS Kewlstart (Default) (Slaves: 13) Channel 14: FXS Kewlstart (Default) (Slaves: 14) Channel 15: FXS Kewlstart (Default) (Slaves: 15) Channel 16: FXS Kewlstart (Default) (Slaves: 16) Channel 17: FXS Kewlstart (Default) (Slaves: 17) Channel
Re: [asterisk-users] IAXy device
Mojo with Horan Company, LLC wrote: Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Just to put out some official word on the matter, the IAXy does indeed have some echo cancellation built in. It has to since it interacts with a phone via a 2 wire to 4 wire conversion with a hybrid. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call deflection on ISDN PRI in Sweden
Hanna Wallin wrote:
Hello List!
We're having trouble making call deflection on ISDN PRI. We would like to
transfer a call to an external extension but keeping the callerid of the
caller so it can be presented to the receiver of the transferred call.
At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware
TE420B. We've ordered the service (CD) from the phone company.
The zapata.conf file inlcludes:
Transfer= yes
Facilityenable=yes
Callerid=asreceived
In extensions.conf we try to transfer a call to an external extension as:
Transfer(ZAP/g0/ ) but that fails with the ${TRANSFERSTATUS} =
UNSUPPORTED.
Ideas anyone? We would really appreciate it!
That supplementary service (CD) is not supported in libpri right now, so
that would be the reason why it doesn't work. The Transfer()
application is for analog lines, IIRC.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
Ruben Zamora wrote: Hi, I have a same problem, last week i was working with TE120 with a little echo in some call, I replace the card with a TE122B ( Included Echo Cancelation VPMADT032) and there was no more echo in my call. But know i have de same probelm with my incoming audio stream gets clipped / dropped when you speak. Please contact Digium technical support about this. This is definitely something that we need to work with the vendor of the echo canceller IP about. Matthew Fredrickson Thanks Ruben Lex Lethol escribió: Hi, I've used all kinds of digium cards without troubles. My last installation is using a TDM2400p with VPMADT032 echo cancel module and after a week of use we noticed that any incoming audio stream gets clipped / dropped when you speak or when ambient noise is high. The call basically feels as in a half-duplex channel, but only to the person behind our asterisk. I found a quick way to recreate by placing a call using zapata channel, someplace that has an audio stream (ie. music on hold from another pbx). When one talks into the phone, one can notice the incoming audio getting muted until you stop talking. First I thought it had to do with polycom configuration although we use the same setup for all installations (VAD, etc), but the same happens with other sip phones and after more tests I can only recreate this using the TDM2400p's FXO trunks. I have an older TDM2400p with no VPMADT032 in production (without this problem), this leads me to believe there maybe something wrong with VPMADT032 module or with my card in particular. Today I rebuilt everything from scratch using latest asterisk 1.2 release, rechecked with the TDM2400p manual zapata configs just to make sure I wasn't missing something. As the manual suggests, I am just using echocancel=yes and this should set 128 default value for the card. In the general zapata options there we have echocancelwhenbridged=yes. I have played with all yes/no combinations without luck. Interrupts and timing stuff are OK, we have good incoming and outgoing audio quality (as long as its not at the same time). Anyone else using this card showing the same problems? Any zaptel/asterisk gurus wanna take a shot at this? Thanks in advance for your feedback/comments. Lex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
Faraz R. Khan wrote: The newer zaptel (1.4.10) says it includes firmware 1.16 from the CHANGELOG: firmware/Makefile, kernel/wctdm24xxp/base.c, kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update wctdm24xxp's VPMADT032 firmware to version 1.16 However there seems to be no way to get this firmware and it does not seem to be included. It checks my firmware and says 1.07 is okay. We had to back that version of the firmware out due to release related problems. As for all problems related to the VPMADT032, if you have any issues, please contact technical support. They will be able to help you with whatever issue you may have. Matthew Fredrickson The URL provided does not contain firmware for the VPMADT032 I* have logged a query with digum. Is there a URL to get this firmware from? On Tue, 2008-04-08 at 11:12 -0500, Ruben Zamora wrote: Lex Thanks, I all ready download the last svn branches from zaptel And i am going to test these afternoon. My phone number es 81-83481611. Thanks Ruben Lex Lethol escribió: Ruben, I am also in Monterrey and have used digium hardware on R2 and PRI. MFC/R2 is not supported by digium but the zaptel driver requirement is the same.. what changes is using libpri vs unicall. Just go ahead and ask them for the firmware update or as Tzafir says use a newer zaptel that should include the updated firmware. If in trouble add me to gtalk I'll try to help out any way possible, Lex On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote: Lex Thanks a lot. These morning i call Digium Support. One issue that i miss in my before e-mail is that i have my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my MFC/R2. Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall. They told me they can help me because they dont have UNICALL support. So... I need to investigate more or wait for a new zaptel or anything else. Generally you can always use a newer zaptel. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm410p w/ echo - no full duplex
Michael J. Liberatore wrote: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. It sounds like you need the new revision of the firmware. Please contact technical support and they should be able to get it to you. Matthew Fredrickson thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm410p w/ echo - no full duplex
Michael J. Liberatore wrote: Matthew, I have just emailed support. Do you know what the latest revision is? Also, is it ok for mg2 to be in zconfig.h and echocancel=yes ? It will Yes. Chan_zap and zaptel know how to automatically use the hardware echo canceller. The configuration options like echocancel and echocancelwhenbridged apply the same to hardware and software echo cancellers. Matthew Fredrickson Digium, Inc. know automatically to use the hw ec rather than the software one? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Friday, April 11, 2008 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm410p w/ echo - no full duplex Michael J. Liberatore wrote: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. It sounds like you need the new revision of the firmware. Please contact technical support and they should be able to get it to you. Matthew Fredrickson thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem TDM01B
troxlinux wrote: hI list, I have some problems with a TDM01B , when I am talking on the phone with another person it cuts himself the call, this alone I am presented when I make calls to the pstn, with internal extensions I don't have problems I show them the log in the CLI -- Nobody picked up in 68000 ms -- Hungup 'Zap/4-1' -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/113-081cf588, ) in new stack == Spawn extension (cyber, 2768000, 2) exited non-zero on 'SIP/113-081cf588' Some person of the list that has presented the same problem with this card, and it finds it solved Please contact technical support. You need to get the new version of the firmware for that card, and they will be able to give it to you. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem TDM01B
troxlinux wrote: hI list, I have some problems with a TDM01B , when I am talking on the phone with another person it cuts himself the call, this alone I am presented when I make calls to the pstn, with internal extensions I don't have problems I show them the log in the CLI -- Nobody picked up in 68000 ms -- Hungup 'Zap/4-1' -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/113-081cf588, ) in new stack == Spawn extension (cyber, 2768000, 2) exited non-zero on 'SIP/113-081cf588' Some person of the list that has presented the same problem with this card, and it finds it solved Sorry, I may have misinterpreted what hardware you have. If you have the new TDM410 card with a hardware echo cancellation module on it, you can get help with a problem similar to that with the new version of the firmware from technical support. If that is not the board that you have, you may have some other issue that you are dealing with. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
around for host bridges that generate fast back to back transactions which the current version of the quad span cards do not advertise support for. 2008-03-14 16:39 + [r3983-3990] Matthew Fredrickson [EMAIL PROTECTED] * firmware/Makefile, kernel/wctdm24xxp/base.c, kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update wctdm24xxp's VPMADT032 firmware to version 1.16 * kernel/wct4xxp/base.c: When doing the ISR rewrite, forgot to include the vpmdtmfcheck when doing DTMF polling causing it to check for DTMF events even when it was told not to (+others) I need to have this system running in about a week and a half. What do you guys say ? The softlockup indicator should be benign. It gets called when loaded the firmware for the part since the firmware image is so large and it takes a long time to load. However, I might have a fix for you. Can you try my stack reduction branch at: https://origsvn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup If that does not work, please contact me directly and I will work with you to get a resolution. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Wed, Apr 16, 2008 at 3:26 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: The softlockup indicator should be benign. It gets called when loaded the firmware for the part since the firmware image is so large and it takes a long time to load. However, I might have a fix for you. Can you try my stack reduction branch at: https://origsvn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup If that does not work, please contact me directly and I will work with you to get a resolution. Matt, Thanks for your feedback. We've already tested the following branch as per Shaun's suggestion, without getting a different behaviour (see today's earlier email to the list): http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Question: - The url you suggest is very similar, are we talking about a different stackcleanup branch ? We are now in the middle of rebuilding a non 4K stack page kernel so as to give it a try with 1.4.10, the branch Shaun suggested, 1.4.9.2 and the branch you mention, if it is in fact different from Shaun's. We wait your confirmation and will post non 4K stack kernel results later today. One thing also I would like to see is your kernel .config file. Another thing that would for sure remove that warning is to disable the kernel softlockup detector which is giving a false lockup warning in this case. I belive it's under the KERNEL HACKING configuration menu if you are using menuconfig. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Shaun Ruffell wrote: Hi Al, Al Baker wrote: Shaun - Could you clarify your post a bit ? 1 - Is the 4 K stacks a Known Problem ? a) If so is it known to be problem on any specific Linux distro ? b) Should ALL installation Check for this PRIOR to doing an Asterisk Install ? I wouldn't really say a known *problem*, since it really depends on what other code is running in the system at the time. I just mentioned that because I've seen 8K stacks help in certain situations. 8K stacks are still the default configuration option in the vanilla kernel. Some distributions (CentOS / Fedora) have switched to 4K by default because they help with memory consumption in highly threaded environments like web servers. For the most part, kernel panics and oops are best handled on a case by case basis with Digium's tech support department since each case is unique. In this case, it looks like his kernel is compiled with the softlockup detector code and it is falsely triggering. Disabling that should remove the warning message at the very least. 2) The branch you mention below - are fixes from it in Any current * release ? They will be in the next Zaptel release. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Wed, Apr 16, 2008 at 4:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: One thing also I would like to see is your kernel .config file. Another thing that would for sure remove that warning is to disable the kernel softlockup detector which is giving a false lockup warning in this case. I belive it's under the KERNEL HACKING configuration menu if you are using menuconfig. Up till now we're running stock CentOS kernel: 2.6.18-53.1.14.el5 The .config is publicly available but we can fwd it to you should you prefer. The kernel we're now building (it is taking quite a while... but it also has been quite a few years since we've built custom kernels... since the 2.0.3x days ?) is based on the stock CentOS kernel with only the 4K stacks option disabled. Please confirm if the SVN branch you suggested is the same or different from the one Shaun suggested yesterday which we already tested. It's the same. Sorry, I sent you that email before I saw his message. I just got an idea for a clever way to make the softlockup detector not complain. I'll let you know when I have a patch to try. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: update with no 4K stack kernel: - The kernel was build from stock centos 5 kernel 2.6.18-53.1.14.el5 - The only .config change was to disable the CONFIG_4KSTACKS Tested zaptel-1.4.10, 1.4.9.2 and the stackcleanup svn branch as suggested by Shaun and Mathew. Short: Results are about the same (stack traces are different). 1.4.10 and the stackcleanup lead to soft hangups, 1.4.9.2 does not. 1.4.10 dmesg snippet: One thing you can also do is pass the nosoftlockup kernel parameter into the kernel from the bootloader. That should disable the softlockup detector. Matthew Fredrickson Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 154 wcte12xp: Setting up global serial parameters for T1 wcte12xp: Found a Wildcard TE122 ACPI: PCI Interrupt :18:08.0[A] - GSI 19 (level, low) - IRQ 162 Found TE2XXP at base address fdff, remapped to f893e000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x3613a400 Reg 1: 0x3613a000 Reg 2: 0x Reg 3: 0x Reg 4: 0x3101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff0031 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12xp: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 2: Primary Sync Source VPM400: Not Present wcte12xp: Setting yellow alarm VPM450: echo cancellation for 64 channels wcte12xp: Clearing yellow alarm BUG: soft lockup detected on CPU#1! [c044d480] softlockup_tick+0x96/0xa4 [c042de00] update_process_times+0x39/0x5c [c04196ef] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [c0605c30] _spin_unlock_irqrestore+0x8/0x9 [f8e82d57] Oct6100UserDriverWriteBurstApi+0x1d/0x27 [wct4xxp] [f8e95de0] Oct6100ApiLoadImage+0x1b5/0x289 [wct4xxp] [f8e9afc4] Oct6100ChipOpen+0x166/0x25e [wct4xxp] [f8e83050] init_vpm450m+0x196/0x306 [wct4xxp] [f8e6ab11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8e6eee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042624e] release_console_sem+0x1b0/0x1b8 [c042680e] printk+0x18/0x8e [f8af6fe4] t1_configure_t1+0xc10/0xc18 [wcte12xp] [f8ac65ef] zt_rbs_sethook+0x102/0x13b [zaptel] [f8acdf6a] zt_ioctl+0x273/0x144f [zaptel] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [c0483cb3] __d_lookup+0x98/0xdb [c047b32c] do_lookup+0x53/0x166 [c047d9ec] do_path_lookup+0x20e/0x25e [c0471053] get_empty_filp+0x99/0x15e [c047b5a5] permission+0xa2/0xb5 [c04e1a36] kobject_get+0xf/0x13 [c046ea1e] __dentry_open+0xea/0x1ab [c046eb43] nameidata_to_filp+0x19/0x28 [c046eb7d] do_filp_open+0x2b/0x31 [c047f4a7] do_ioctl+0x47/0x5d [c047f707] vfs_ioctl+0x24a/0x25c [c0470de6] __fput+0x13f/0x167 [c047f761] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 3: Secondary Sync Source Completed startup! 1.4.9.2 dmesg snippet: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.9.2 Zaptel Echo Canceller: MG2 PCI: Enabling device :12:01.0 (0150 - 0153) ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 154 wcte12x[p]: Setting up global serial parameters for T1 wcte12x[p]: Found a Wildcard TE122 Found TE2XXP at base address fdff, remapped to f893e000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x3571b400 Reg 1: 0x3571b000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x010200ff Reg 9: 0x00fd0001 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12x[p]: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 SPAN 2: Primary Sync Source timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 64 channels VPM450: hardware DTMF disabled
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Wed, Apr 16, 2008 at 6:51 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: One thing you can also do is pass the nosoftlockup kernel parameter into the kernel from the bootloader. That should disable the softlockup detector. Tested with no 4K stack kernel and stackcleanup svn branch zaptel version. Correct, the kernel no longer complains about the soft hangup. However the system still hangs (console inoperative, etc) while ztcfg'ing... That is normal while the firmware is loading. It should go away after the firmware has loaded. Can you answer my previous questions ? - If going live would you recommend zaptel 1.4.9.2 or 1.4.10 ? I recommend 1.4.10 by default. However, from what you said it would appear that you are having problems with 1.4.10 so you might stay with 1.4.10 if you are not having any issues with it. - Does the current behaviour from 1.4.10 prevent firmware uploading ? No. There is nothing that is happening that prevents firmware uploading. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Wed, Apr 16, 2008 at 7:18 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ex Vito wrote: Tested with no 4K stack kernel and stackcleanup svn branch zaptel version. Correct, the kernel no longer complains about the soft hangup. However the system still hangs (console inoperative, etc) while ztcfg'ing... That is normal while the firmware is loading. It should go away after the firmware has loaded. Ok. So here is our reasoning according to collected info. Please correct us where appropriate: 1. The system is supposed to hang while the firmware loads into the DSPs under any zaptel version 2. zaptel 1.4.10 leads to a soft hangup detected, zaptel 1.4.9.2 does not (assuming softhangup detection active in kernel) 3. zaptel 1.4.10 takes much longer ztcfg'ing than 1.4.9.2, that's why the soft hangup is detected under zaptel 1.4.10 (difficult to time, but let's say 1.4.10 takes 10s, 1.4.9.2 takes 3s) Now, back to the original question: - Should this be considered a regression ? - Next steps: a) file a bug and move this analysis to the bug tracker b) don't file bug and move analysis to the dev list c) don't file bug, keep on working on the users list I recommend 1.4.10 by default. However, from what you said it would appear that you are having problems with 1.4.10 so you might stay with 1.4.10 if you are not having any issues with it. I just realized where this is coming from. I was attempting to patch this from a different angle, but as soon as you mentioned the drastic difference in load time I realized what had happened. I'm going to make another update to my stack reduction branch to see if I can fix this. I'll let you know when it's done. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Fri, Apr 18, 2008 at 4:15 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: I just realized where this is coming from. I was attempting to patch this from a different angle, but as soon as you mentioned the drastic difference in load time I realized what had happened. I'm going to make another update to my stack reduction branch to see if I can fix this. I'll let you know when it's done. Great. We'll be right here... Since the bug has been closed, we post the timing results we did within this context. I just updated the branch. Wait about 5-10 minutes in case for the changes to get mirrored, and then try updating and doing it again. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Fri, Apr 18, 2008 at 9:36 PM, Ex Vito [EMAIL PROTECTED] wrote: On Fri, Apr 18, 2008 at 8:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: I just updated the branch. Wait about 5-10 minutes in case for the changes to get mirrored, and then try updating and doing it again. Looks better, no more soft lockup and ztcfg time is comparable to 1.4.9.2's: Matthew, ...is there any specific test you'd like us to perform on this revision ? (considering that currently we have no PSTN line to attach to... we can cross-connect the spans and generate traffic or, cross-connect with another lab system) Not really from me specifically. You already tested what I wanted to be tested, and that was to see if I could fix the load time issue and softlockup warning. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Fri, Apr 18, 2008 at 10:12 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ex Vito wrote: Matthew, ...is there any specific test you'd like us to perform on this revision ? (considering that currently we have no PSTN line to attach to... we can cross-connect the spans and generate traffic or, cross-connect with another lab system) Not really from me specifically. You already tested what I wanted to be tested, and that was to see if I could fix the load time issue and softlockup warning. Ok. So, since the bug we logged was closed and these tests weren't registered along with it, when can one expect to have your new code available in a zaptel release ? In the next one or maybe later because the branch you're working on has lots of different things to merge ? It should be in the next release. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quality problems with ISDN PRI
Carles Pina i Estany wrote: Hello, We have an Asterisk server with a TE410P Quad-Span togglable E1/T1/J1 card, 3 SPANs configured and OK and one SPAN unconfigured. In our tests it works fine, but when it has a big laod of calls (say, from 40 to 60) we have quality problems: some calls has the sound cut-off (during the call, voice was not stable) The IRQ card is alone, CPU load was not high, network was fine for sure. This server is receiving the calls from SIP channels and routing to the primaries. It's a HP server, multicore, multiCPU. I'm wondering if someone has had these kind of problems (quality problems, sound cut off) with 40 and 60 calls but not with 2 or 3, using Digium cards. Bit later I will call to Digium but I thought that here there is lot of people with lot of experience with these cards. There are a number of factors that can contribute to this type of problem, but probably the best solution is to call support and talk to them about this. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quality problems with ISDN PRI
linuxian iandsd wrote: i have HEARD asterisk wasn't made with the idea to run on multi-core processors in mind .. the result was that it uses one core all the time ..so one single P4 3.4 GHZ would perform better than a far more newser quad one. but i might be wrong. but one thing for sure check hardware compatibility before you buy anything. For the purposes of making sure list records are accurate, this in not true. Asterisk was indeed written with the intention to run on multi-core systems, and should utilize extra cores just fine. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)
Steve Totaro wrote: My question is does ANYONE do ANY testing on these releases? It would seem that this bug is so paramount to the purpose of the code that had anyone taken a MINUTE to TEST, it would have been discovered IMMEDIATELY. Not if you already had a zaptel udev rules script installed on the system that's used as the test machine. This was a regression do to recent Makefile changes. A test for this problem has now been added to our pre-release regression testing. Matthew Fredrickson sigh. Thanks, Steve Totaro On Wed, Apr 30, 2008 at 8:41 AM, Steve Totaro [EMAIL PROTECTED] wrote: Sean Bright to Asterisk show details 4:47 PM (15 hours ago) There is a bug in 'make install' in Zaptel 1.4.10 that causes the devices to not be installed correctly. You can either install 1.4.9 or wait for 1.4.11 to be released. On Wed, Apr 30, 2008 at 8:22 AM, Zoran Milenkovic, Datatek d.o.o. [EMAIL PROTECTED] wrote: Hi list! I have RHEL Server rel5 (Tikanga) 2.6.18-8.el5 #1 SMP Fri Jan 26 14:15:21 EST 2007 i686 i686 i386 GNU/Linux with installed digium packets 1. Asterisk 1.4.19 2. Zaptel 1.4.10 3. Libpri 1.4.3 My Digium hardware is [EMAIL PROTECTED] ~]# zaptel_hardware pci::04:00.0 wctdm+ e159:0001 Wildcard TDM400P REV I ...and I have 2 FXO and 2 FXS ports installed inside the TDM400P card The problem is the asterisk doesn't recognize the Zap channels at all. The error is No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) and there is the original output form Astersik console: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/zoran-09f1bf90, Zap/3|20) in new stack [Apr 30 13:17:48] WARNING[2845]: channel.c:3001 ast_request: No channel type registered for 'Zap' [Apr 30 13:17:48] WARNING[2845]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/zoran-09f1bf90, ) in new stack == Spawn extension (local, 12, 2) exited non-zero on 'SIP/zoran-09f1bf90' And everything was working quite fine when I was on asterisk 1.2.13, previously installed on this very same server, same Digium card etc. The configurations are totaly the same, also. What could be the resolution of this problem? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.10.1 Released
Matt Watson wrote: Does anybody know if this version fixes the soft lockup during ztcfg using a TE200B? http://bugs.digium.com/print_bug_page.php?bug_id=12468 No, continue to use the stackcleanup branch. That is going to be merged in for the next major release (1.4.11). Matthew Fredrickson -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Development Team [EMAIL PROTECTED] Sent: Thursday, May 01, 2008 1:07 PM Subject: [asterisk-users] Zaptel 1.4.10.1 Released The Asterisk.org development team has announced the release of Zaptel version 1.4.10.1. This release is a bug fix release for a regression in which the Zaptel udev rules were not installed correctly, as well as a few minor fixes in the xpp drivers. This release is available as a tarball as well as a patch against the previous release. It is available for download from downloads.digium.com. Thank you for your support! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI debugging ...
Gordon Henderson wrote: On Fri, 16 May 2008, Gordon Henderson wrote: Have a problem with an ISDN30 line in the UK. So following up my own post.. I've not solved this issue, but I think I know what causes it. This was my experiment to put 2 cards in one 1.3GHz system - a TDM400 with 2 x FXO and 2 x FSX and a TE120P - E1 card. The PRI card loses interrupts, so I'm guessing it loses a frame of data when it loses an interrupt, and eventually it gives up and does a reset. The TDM card was rock solid. The system is using oslec too FWIW. When I unloaded the wctdm module the PRI performend flawlessly. So I'm suspecting the 1.3GHz processor and underlying IO is marginal for this application. The Mobo doesn't have an APIC, just old PIC hardware, although both cards were on separate IRQs - the TDM card had the higher priority IRQ though - didn't have time to test it with the cards swapped over, but loading the modules in a differnt order didn't make any difference. Turning off the USB hardware didn't help either. The processor does seem to have a highish high-priority interrupt load (as seen by top). I'll be trying a newer kernel when I get a chance though (this is 2.6.18, compiled to match the motherboard exactly) Making calls through the TDM card just made it worse. However when it was working, it was working very well indeed, but the occasional time when it dropped all calls (about once an hour) wasn't good. You might try turning off echo cancellation to see if your D-channel performance improves. That would be a good test to tell if you should look into perhaps getting either a faster CPU or a hardware echo canceller. It's possible that you may be saturating your poor 1.3 Ghz CPU by doing echo cancellation for too many channels on it. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722 over ISDN PRI/BRI
Simon Hyde wrote: Hi, G.722 is heavily used by Broadcasters worldwide for wideband voice communications over ISDN. I'd like to be able to receive these G.722 over ISDN calls into an Asterisk exchange (with mostly a view to routing the calls to a Voicemail box where material can be recorded). I have been examining source code for the 3 different ISDN Channels in Asterisk and they all seem to be hard- codec to aLaw/uLaw G.711. It looks as though chan_capi *might* support bridging of G.722 data from one ISDN port to another, but not routing to any other source/transcoding/passing to voicemail. So I guess my question is, am I correct in the belief that all Asterisk's ISDN channels currently don't support anything other than G.711? How easy would it be to extend one of the ISDN channels to support G.722? Your belief is correct. Right now, the ISDN channels (at least in chan_zap) G.711 is the only voice codec that is supported. I'm not sure what is going to be necessary to get G.722 working there. If it's as simple as changing the bearer capability, the chan_zap work on top of that should be fairly easy. If you have to implement any of the H.* specs to get it working, that will be a bit more trouble. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune vs rxgain/txgain
Noah Miller wrote: Well, that clears it up a little. I think where I get confused is that sometimes using fxotune is called balancing the hybrid and some times using ztmonitor and adjusting the txgain/rgain settings is called balancing the hybrid. Perhaps they both try to achieve the same goal, but through different means? Not quite. Gain adjustment affects volume levels of the respective direction you are adjusting (echo and all). Balancing the hybrid via fxotune attempts to balance the hybrid in a manner so that the hybrid will remove as much of the echo as possible. This leads me to my other question - Are these two techniques mutually exclusive? In some posts from Matthew Frederickson, it seems that they are, and that if you use fxotune, you should set your gains back to zero. Some other people seem to suggest using both fxotune and adjusting gain levels. I note that Stephen Bosch asked just this question some time back, and nobody was able to answer him. These techniques are not mutually exclusive, I usually want people to use gain modification as the last step in trying to eliminate echo (after balancing the hybrid and making sure you are using a good echo canceller). In the case of running fxotune, your zapata.conf software gain levels should not affect its operation. If you are using any of the hardware gain settings (wctdm24xxp module parameters) you should normalize those to 0 beforehand so that they do not interfere with the calibration process. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
John Morey wrote: Tilghman, Thanks for the pointer. I'll check this tomorrow and let you know. Also, I would like to see the output without the -d flag and with the -v flag. This will output a lot of data (the echo ratio for every possible coefficient setting it has tried per port). Matthew Fredrickson John On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune vs rxgain/txgain
Noah Miller wrote: Hi Matthew - These techniques are not mutually exclusive, I usually want people to use gain modification as the last step in trying to eliminate echo (after balancing the hybrid and making sure you are using a good echo canceller). In the case of running fxotune, your zapata.conf software gain levels should not affect its operation. If you are using any of the hardware gain settings (wctdm24xxp module parameters) you should normalize those to 0 beforehand so that they do not interfere with the calibration process. Thanks for your responses! I actually didn't realize there are hardware gain settings available for wctdm24xxp (is there any documentation on this? I can't seem to find any). I assume the hardware gains default to 0 if left unset? Correct. They are set as module parameters, and actually only apply to fxo modules. Just two more questions: 1) I think we were experiencing ECFO with an rxgain setting of +10db (after having balanced the hybrid using fxotune). I'm guessing this is because that rxgain value amplifies the echo a bit too much. I know this is a bit of a loaded question, but is there a certain range of values for rxgain/txgain that we should stay within in order to avoid exacerbating any echo issues? I couldn't give you exact numbers off the top of my head. It's not hard to notice though if it's happening :-) 2) Are rxgain/txgain values applied before or after hardware echo cancellation? rxgain is pre-hardware echo canceller and txgain is post hardware echo canceller. (zapata.conf rxgain and txgain). -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
John Morey wrote: I switch the wires in lines 5-8 (i.e. reversed tip and ring) and reran fxotune to tune the lines. fxotune.conf ended up looking exactly the same as before the change. Since I was expecting/hopping to see a change but did not I switched everything back to the way it was. Is there a way to test the lines, using a multi-meter maybe, to tell if the tip and ring are correct or reversed? After putting things back I reran fxotune to get the verbose output. It, foxtune.out.gz, is attached. fxotune seems to have had a better time with It seems that one way or another the attachment didn't go through. Can you email the tarball to me directly or post it to a website? Thanks, Matthew Fredrickson line 7 during this run. fxotune.conf now contains: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=4,0,0,0,0,0,0,0,0 8=7,255,251,251,2,255,255,1,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 I tried calling directly into the lines above and it seems lines 5,6,8 have much more echo than lines 7,9,10. So just for fun I edited fxotune.conf to the following and reloaded (fxotune -s) it: 5=5,0,0,0,0,0,0,0,0 6=5,0,0,0,0,0,0,0,0 7=4,0,0,0,0,0,0,0,0 8=5,0,0,0,0,0,0,0,0 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 Unless I am just spacing out the echo on 5,6,8 seems less now. I really have no idea what is going on. John On Fri, Jun 6, 2008 at 1:31 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: John Morey wrote: Tilghman, Thanks for the pointer. I'll check this tomorrow and let you know. Also, I would like to see the output without the -d flag and with the -v flag. This will output a lot of data (the echo ratio for every possible coefficient setting it has tried per port). Matthew Fredrickson John On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
John Morey wrote: I switch the wires in lines 5-8 (i.e. reversed tip and ring) and reran fxotune to tune the lines. fxotune.conf ended up looking exactly the same as before the change. Since I was expecting/hopping to see a change but did not I switched everything back to the way it was. Is there a way to test the lines, using a multi-meter maybe, to tell if the tip and ring are correct or reversed? After putting things back I reran fxotune to get the verbose output. It, foxtune.out.gz, is attached. fxotune seems to have had a better time with line 7 during this run. fxotune.conf now contains: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=4,0,0,0,0,0,0,0,0 8=7,255,251,251,2,255,255,1,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 I tried calling directly into the lines above and it seems lines 5,6,8 have much more echo than lines 7,9,10. So just for fun I edited fxotune.conf to the following and reloaded (fxotune -s) it: 5=5,0,0,0,0,0,0,0,0 6=5,0,0,0,0,0,0,0,0 7=4,0,0,0,0,0,0,0,0 8=5,0,0,0,0,0,0,0,0 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 Unless I am just spacing out the echo on 5,6,8 seems less now. I really have no idea what is going on. Ok, I looked at the output of you running fxotune. Basically, the lines that have numbers in them besides 0 (after the first two terms x=y,...) are the complex line simulation line models. The output you gave me demonstrated that they gave the best return loss characteristics using the built in test frequencies. It's possible that your setup is not performing well with these line models, which is why you might notice less echo using the second set of settings you listed above. Which echo canceller are you using with this, by the way? (Hardware, software, if software, which software echo canceller). Matthew Fredrickson John On Fri, Jun 6, 2008 at 1:31 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: John Morey wrote: Tilghman, Thanks for the pointer. I'll check this tomorrow and let you know. Also, I would like to see the output without the -d flag and with the -v flag. This will output a lot of data (the echo ratio for every possible coefficient setting it has tried per port). Matthew Fredrickson John On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list
Re: [asterisk-users] Disconnect on PRI ignored?
Alexander Zielke wrote: Hi List, i recently set up a system with a TE410P. Everything works, except that disconnects don't seem to be processed. Here is what i get: -- SIP/2025-08245ac8 is ringing -- SIP/2025-08245ac8 is ringing -- SIP/2025-08245ac8 is ringing Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 23819/0x5D0B) (Originator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) q931.c:3779 q931_receive: call 23819 on channel 6 enters state 12 (Disconnect Indication) -- SIP/2025-08245ac8 is ringing -- SIP/2025-08245ac8 is ringing ... I just made a call from the outside to a local SIP-Phone, but when the outside call hangs up, the Phone keeps ringing. The call will only hangup, if i take the call, or wait for the call to time out. The only similar thing i found is the bug at http://bugs.digium.com/view.php?id=9588, but that seems fixed in 1.4.21.1. Did anyone else experienced something like that? If you are using libpri-1.4.4, you should either downgrade to 1.4.3 or upgrade to 1.4.5. A new default behavior was introduced in 1.4.4 (which should have been optional, not default) which causes a channel to be left open until the RELEASE timer expires when a DISCONNECT is received with Inband progress information avaiable. Matthew Fredrickson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
emist wrote: My best guess from looking at that is that its a driver bug. The last thing that happens before the lockup seems to be an ioctl call to the device. That was a bug that should have been resolved by 1.4.11 (he subsequently updated and it was resolved). Matthew Fredrickson Digium, Inc Hope it helps, Igor H. Lee, John (Sydney) wrote: This time, I am trying to remotely install Asterisk in China. I was told that an E1 line has been installed and so I plug it into port 1 of a TE412P. On the box, first of all, I just installed Zaptel 1.4.10.1. # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] # vi zaptel.conf [...] span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 *** However, I received a red alarm in zttool and the LED on the TE412P card is also red. *** I have made sure that the jumper is closed for port 1 on the TE412P card and so it could not be the jumper problem. ### Because this is the first time I install Asterisk in China and I was wondering if their E1 is different from the Euro E1. ### However, I went into dmesg and I discovered the following. ### Could it really be a zaptel bug? I saw on a similar few on the digium bug list but I cannot be 100% sure. Any thoughts? About to enter spanconfig! Done with spanconfig! Registered tone zone 33 (China) About to enter startup! TE4XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 128 channels BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681] Pid: 4681, comm:ztcfg EIP: 0060:[f8cba1df] CPU: 2 EIP is at init_vpm450m+0x32d/0x34a [wct4xxp] EFLAGS: 0286Tainted: G (2.6.18-92.1.6.el5 #1) EAX: EBX: f76ae8f0 ECX: 0019 EDX: ESI: f7997400 EDI: 0286 EBP: f76838c0 DS: 007b ES: 007b CR0: 8005003b CR2: b7f01007 CR3: 37bc1000 CR4: 06d0 [f8ca1b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8ca5ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042609c] release_console_sem+0x17e/0x1b8 [c046d53a] cache_alloc_refill+0x14b/0x450 [f8956f61] zt_ioctl+0x273/0x144f [zaptel] [c04d7d45] generic_make_request+0x248/0x258 [c045ae3c] __do_page_cache_readahead+0x69/0x1c6 [c0484a5b] __d_lookup+0x98/0xdb [c047c110] do_lookup+0x53/0x166 [c047e7e4] do_path_lookup+0x20e/0x25e [c047c389] permission+0xa2/0xb5 [c04e2d06] kobject_get+0xf/0x13 [c046f7fa] __dentry_open+0xea/0x1ab [c046f91f] nameidata_to_filp+0x19/0x28 [c046f959] do_filp_open+0x2b/0x31 [c048029b] do_ioctl+0x47/0x5d [c04804fb] vfs_ioctl+0x24a/0x25c [c0471bbe] __fput+0x13f/0x167 [c0480555] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Completed startup! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Lee, John (Sydney) wrote: The test for that is simple: head -n 1 /proc/zaptel/* Let's look at all four spans. Not just the first one. Thanks Tzafrir. # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 So I am quite sure that port 1 is plugged in properly. As I am dealing with telecom in China, I think I might have stepped onto the MFC R/2 bombshell but I have no idea whether the signalling is ISDN or R2. I tried the suggestion on http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 but the red alarm is still on. If it is really R2, then maybe I need to buy an E100P card instead of TE412P. No, you should be fine with a TE412. Just make sure that your line is plugged in correctly and your span= line is correct for the line settings. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
Tilghman Lesher wrote: On Friday 15 August 2008 13:45:11 Jay R. Ashworth wrote: On Fri, Aug 15, 2008 at 02:37:46PM -0400, Matt Florell wrote: Most carrier sales people don't know what TBCT is unfortunately, and even if a carrier is capable of doing it, it is a possiblity that not all of their equipment is capable of doing it. One client of mine tried to get TBCT working across all 16 of their PRIs(all on the same carrier) and it only worked on 4 of them, supposedly because not all of the telco equipment was capable of the feature. I expect to fight this battle, yes. :-) This actually depends on the kind of PRI service you have. For instance with DMS100 circuits you can only do TBCT with calls that come in to your circuit, not with outgoing calls. As for connecting two incoming calls, since that is not possible in Asterisk(to natively bridge two incoming calls together) I can't see how you would get that to work even if it is possible in TBCT. To be more clear, what I'm after is to have *someone else besides me* place calls out their PRI, and then TBCT those placed calls to my DN. By the time the calls get to me, they should just be standard phone calls. So I expect the call-placing-party to need TBCT, but not me. I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are capable of TBCT with the current zaptel code-base. Also, the two B channels involved in the TBCT have to use the same D channel. And I'm probably not concerned with whether Asterisk can deal with TBCT, because Asterisk probably won't be involved at that stage; just once the call's transferred to me. But before I inquire of said second party whether they *can* do that, I wanted to confirm it was possible. 2BCT works when the telco originates the call and Asterisk is hairpinning the call back out the same PRI circuit. However, Asterisk does not support the opposite direction. That is, a call originated from Asterisk that comes back in via the same PRI circuit cannot be 2BCT. I'm not certain whether this is a limitation of Asterisk alone or of the protocol, but it cannot be done. Similarly, Asterisk cannot complete a 2BCT request, if Asterisk is on the NET side of the PRI circuit. That might could be added in the future, but it is not supported now. So in summary, Asterisk can request 2BCT, but it cannot perform a 2BCT if requested from the other side. Let me clarify some of this. Under no circumstances can Asterisk receive a TBCT request. We just ignore them. We can initiate them however. There are different TBCT implementations, dependent on which switch type is used, with different restrictions associated with each switch type selected. For true TBCT (on switchtypes of NI2 and 5ESS, AFAIK), you can have any combination of inbound and/or outbound channels (one inbound/one outbound, two inbound, two outbound) and transfer them to the upstream switch. The protocol doesn't care. For DMS100's version of TBCT, called RLT, one leg *must* be inbound and the other *must* be outbound. No other combination is going to work. This is explicitly mentioned in the protocol in RLT. Hope that helps a bit. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DSS1 vs SS7
Kevin P. Fleming wrote: Alex Balashov wrote: Some carriers now do offer private SS7 instead of ISDN. But there is absolutely no reason why you should be doing this with Asterisk. Asterisk-SS7 is quite tenuous at best. Unless you have some specific reason to be using it, don't. Actually, SS7 support in Asterisk 1.6.0 appears to be quite solid, and it is being used in a quite a number of production deployments. Thanks for the plug Kevin! :-) Yeah, actually, if you guys want to know more there's an asterisk-ss7 mailing list. Asterisk-1.6.0 with libss7 is being used in many successful and high traffic installations around the world. The current record (that I have been told of) is an installation doing over 100,000 calls per day. So try to beat that ;-) Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
