Re: [asterisk-users] Stable-Stable Asterisk
Jay R. Ashworth wrote: Just how easy is it to roll back to the older release when the feces hit the fan? Seems like making that simple would be pretty important? Well, my opinion is that it is very easy. If it's not, I'd be happy to hear it so that we can make it easier. The only thing you have to clean up, really, is deleting old modules. Then, even if you forget, the Makefile will give you a huge warning about modules being there that it is not about to overwrite when installing Asterisk, meaning that they are probably too old or too new. The other thing to do is to review the contents of the UPGRADE.txt file to be sure that the configuration doesn't use any options that were deprecated in 1.2 and removed in 1.4. -- Russell Bryant Software 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] Stable-Stable Asterisk
On 8/24/07, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Fri, Aug 24, 2007 at 04:00:23PM -0400, Matt Florell wrote: With all of that said, I do have a testing setup that allows me to run tests at high loads on Asterisk, but not all scenarios can be checked in a testing setup. I ran a mid-volume test on 1.4.10 and it worked without crashing. I wanted to test a new feature in 1.4 so I put the server into production. It worked fine for a few hours under small load, but once the load increased there were several issues(mostly relating to stuck locks I am guessing) and the server would crash every few hours and also have some weird Manager API issues. So after a few days I rolled the server back to 1.2.X and all was well again. Running the tests again later at a higher call volume and on servers with more horsepower revealed the same crashes and other issues as I noticed in production. Here's a secondary question (and Matt, I *do* plan to get around the damned corner to one of your meetups one of these days :-): Just how easy is it to roll back to the older release when the feces hit the fan? Seems like making that simple would be pretty important? (Context: my boss is about to tip on playing with Asterisk, finally..) Cheers, -- jra It's really just a matter of recompiling Asterisk. Libpri-1.4 and zaptel-1.4 seem to work fine with Asterisk 1.2 as well so in a pinch you may not have to recompile those. Most 1.2 dialplans will work just fine with 1.4, so as long as you delete the modules between builds: rm -f /usr/lib/asterisk/modules/* You should be fine. MATT--- ___ --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] which OS would be fine for asterisk
I concur, Centos 4.4 FTW. ^^ -- Original Message -- From: Edgar Guadamuz [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Fri, 24 Aug 2007 23:50:51 -0600 I have used CentOS and it works fine and it is easy to install. I know that Debian is a little more complicated to install Asterisk and some teatures on Debian. I'd choice CentOS 4.2 or 4.4, as my personal preference. ___ --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 Sent via the WebMail system at rockynet.com ___ --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] Stable-Stable Asterisk
Tony Mountifield wrote: In article [EMAIL PROTECTED], Julian Lyndon-Smith [EMAIL PROTECTED] wrote: We have been running 1.4 since July 06 (it was trunk then), and have upgraded often with the 1.4 branch (Currently on SVN-branch-1.4-r77571). We have 100+ extensions (SIP) and 30 ISDN channels. We often have 50+ agents available for outbound calls and queues (20+ queues). We are making / receiving approx 5000+ calls per day. We use jabber and odbc heavily (updating / reading / Creating) as well as using odbc for cdr records. All calls are recorded (monitor at the moment). We use SMS inbound and outbound. This is on a dell 2850 with 2gb ram (top - 21:31:11 up 246 days). Asterisk has System uptime: 3 weeks, 4 days, 7 hours, 57 minutes, 44 seconds Whilst nowhere near the levels of some other people, for our purposes, 1.4 is working very very well for us, and the development guys have our gratitude and respect. It's a damn fine piece of work that has saved my company a lot of money in the 2 years we've been using asterisk. But it appears that you are not using IAX. I suspect until extremely recently it is IAX that has been the weak link in 1.4, because of the change from single-threaded to multi-threaded. The latest work on IAX with astobj2 looks like it should solve this at last! This is also true - there have been a lot of fixes for IAX recently. We simply have no need for it at the moment. Julian Cheers Tony ___ --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
[asterisk-users] Polycom firmware download
Hi, I'm trying to use Polycom 330 and apparently it needs latest firmware (SIP 2.2.0). I dont have access to polycom site to download and was wondering if any of you guys have it. Thank you! ___ --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 as a softswitch
Le Fri, 24 Aug 2007 20:50:05 +0400, Mark Quitoriano [EMAIL PROTECTED] a écrit: What is a good softswitch that is also open source rather than asterisk? You may want to check out freeswitch. ___ --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
[asterisk-users] IAX2 sofphones in use with dyndns (iax.conf)
Hi, I'm using asterisk 1.4.11 and try to configure asterisk and iax phones with dyndns. I'm using ZoIPer (and ePhone) as softphones that support the IAX2 protocol. The softphone on Side A is at: phone-a.dyndns.org The server is a Debian Linux Etch and dials up, makes dhcp, dnsmasq and ddclient for itself. The server on side A is: server-a.dyndns.org The softphone on side B is at phone-b.dyndns.org What parameters do I have to use for the iax.conf on server-a.dyndns.org? Both phones don't register. If I do set defaultip=phone-x.dyndns.org both phones register but we can't hear each other. Thanks for your help! Regards, Michael - - - - - - - sip.conf [general] bindport=4569 bindaddr=server-a.dyndns.org [phone-a] username=phone-a type=friend secret=verysecret auth=md5 host= dynamic ; or do I have to use host=phone-a.dyndns.org ; maybe I need to use defaultip=phone-a.dyndns.org? context=my-iax-phones dtfmmode=rfc2833 ; trunk=yes|no qualify=no ; some softphones don't like qualify=yes? callerid=phone-a 3000 [phone-b] username=phone-b type=friend secret=verysecret auth=md5 host= dynamic ; or do I have to use host=phone-a.dyndns.org ; maybe I need to use defaultip=phone-b.dyndns.org? context=my-iax-phones dtfmmode=rfc2833 ; trunk=yes|no qualify=no ; some softphones don't like qualify=yes? callerid=phone-b 3001 extensions.conf: [my-iax-phones] exten = 3000,1,Dial(IAX2/phone-a,20,mt) exten = 3000,2,VoiceMail(3000,u) exten = 3000,3,Hangup exten = 3001,1,Dial(IAX2/phone-b,20,mt) exten = 3001,2,VoiceMail(3001,u) exten = 3001,3,Hangup ___ --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] which OS would be fine for asterisk
On Sat, Aug 25, 2007 at 12:31:15AM -0600, Anthony Francis wrote: I concur, Centos 4.4 FTW. ^^ Centos 4.4, as in not the latest, and already hald the packages are not in the repositories? Any specific reason you avoid Centos 4.5? Centos5? Any specific reason to keep using something that is still labled kernel 2.6.9, that has quite a buggy udev implementation, for once? Debian++, BTW. -- Tzafrir Cohen icq#16849755jabber:[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
Re: [asterisk-users] Gizmo revisited
On Fri, Aug 24, 2007 at 10:05:44PM -0500, Carlos Leal wrote: Launched the OS X version of Gizmo after about a year of inactivity, downloaded the update and discovered the new improved Giszmo features Asterisk interoperability by allowing a secondary SIP account to be registered simultaneously. It also allows you to make the routing choice for outgoing calls; your own server or via Gizmo. So far, this is the best SIP softphone I've come across for OS X. It comes in other flavors and I thought I'd mention it as it can be free and I haven't seen it mentioned recently. Can it be used with any service which is not sipphone.com? (The phone itself is based on xten's phone, IIRC, and is thus certinly non-free) -- Tzafrir Cohen icq#16849755jabber:[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
Re: [asterisk-users] which OS would be fine for asterisk
. . . Personally I recommend SuSE Linux. OpenSuSE without the GUI installed will do just fine. If you want to buy SLES that's fine, but I really don't see the value in it. The value would be live support and access to online updates. Courtesy (for the price) of Novell. There are, of course, some differences between OpenSuse and SLES. I've run Asterisk on SLES 9 and SLES 10 without problems. Your View/Mileage May Vary. joe a. ___ --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
[asterisk-users] asterisk and vad/cng
Hi List, i've set up a cisco 7912 for my asterisk box. I've had problems with VAD and CNG. After googling a bit, i've found an article about asterisk not supporting these two protocols, therefore it's better to turn them off. Since then i did not found answer to my two questions, maybe somebody here could help me: a) am i even able to turn off vad/cng on cisco 7912? SIP image 8.0 version. I've been through the cisco admin guide, but it did not help. b) the article mentioned above was dated 2006, does asterisk still not support VAD/CNG? I'm using 1.4.8, the log says it does not, but maybe a patch, or 1.4.11? I encounter no big problems, only with MOH: i have to make some noise (breathing loud etc) to hear the music. Not a big deal after all. And some other questions not really related: anybody has experience with the following phones: linksys 921/941 and cisco 3911? Do these have the same problem? According to their datasheet, linksys supports VAD, but the user guide says nothing about turning it off. Thank you for any replies, it would help me a lot! Regards Adam ___ --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
[asterisk-users] Asterisk 1.2 + Zaptel 1.4 + HPEC = Crash?
Hello, Has anyone tried the combination of asterisk 1.2.24, zaptel 1.4.5 and HPEC 9.00.003? 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 -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more! ___ --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] Restart status
Ron Joffe wrote: If I issue a restart gracefully command, the system will wait until all channels are idle before restarting. During the time the system is waiting for idle activity, is there a command that can let me know it is in graceful restart wait mode ? Thanks, Ron Good question. I have wondered that a couple of times and assumed that there was not, but maybe there is. Probably not much help, but if you rarely issue commands such as this, hit the up arrow a few times. Thanks, Steve ___ --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] Asterisk 1.2 + Zaptel 1.4 + HPEC = Crash?
Matthew Fredrickson wrote: 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 Matt Florell seems to think it should be OK./ It's really just a matter of recompiling Asterisk. Libpri-1.4 and zaptel-1.4 seem to work fine with Asterisk 1.2 as well so in a pinchyou may not have to recompile those. Most 1.2 dialplans will work just fine with 1.4, so as long as you delete the modules between builds: rm -f //usr/lib/asterisk/modules//* / /You should be fine. / MATT--- ___ --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?
On Sat, Aug 25, 2007 at 12:20:56PM -0400, Steve Totaro wrote: Matt Florell seems to think it should be OK./ It's really just a matter of recompiling Asterisk. Libpri-1.4 and zaptel-1.4 seem to work fine with Asterisk 1.2 as well so in a pinchyou may not have to recompile those. Most 1.2 dialplans will work just fine with 1.4, so as long as you delete the modules between builds: rm -f //usr/lib/asterisk/modules//* If you want to be able to build asterisk 1.2 with zaptel 1.4 you'll need: ln -s ../zaptel/zaptel.h /usr/include/linux/zaptel.h ln -s zaptel/tonezone.h /usr/include/tonezone.h Note: I'm not sure exactly what happens if you run a 'make install' of zaptel 1.2 on top of that later. I guess you should remove those two symlinks manually before you do that. -- Tzafrir Cohen icq#16849755jabber:[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
Re: [asterisk-users] asterisk and vad/cng
On 8/25/07, Adam KOSA [EMAIL PROTECTED] wrote: Hi List, i've set up a cisco 7912 for my asterisk box. I've had problems with VAD and CNG. After googling a bit, i've found an article about asterisk not supporting these two protocols, therefore it's better to turn them off. Since then i did not found answer to my two questions, maybe somebody here could help me: a) am i even able to turn off vad/cng on cisco 7912? SIP image 8.0 version. I've been through the cisco admin guide, but it did not help. This isn't a Cisco mailing list. b) the article mentioned above was dated 2006, does asterisk still not support VAD/CNG? I'm using 1.4.8, the log says it does not, but maybe a patch, or 1.4.11? VAD is supported, CNG is not. I encounter no big problems, only with MOH: i have to make some noise (breathing loud etc) to hear the music. Not a big deal after all. Bug in the MOH code And some other questions not really related: anybody has experience with the following phones: linksys 921/941 and cisco 3911? Do these have the same problem? According to their datasheet, linksys supports VAD, but the user guide says nothing about turning it off. Linksys 921/941 work fine with Asterisk. Just don't enable CNG. ___ --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] Polycom firmware download
On 8/25/07, Al lists [EMAIL PROTECTED] wrote: Hi, I'm trying to use Polycom 330 and apparently it needs latest firmware (SIP 2.2.0). I dont have access to polycom site to download and was wondering if any of you guys have it. Thank you! Best idea is to ask your reseller. I am not aware of a community site with Polycom firmwares. If you wish I think I know the person who created http://spc.pifiu.com and if anyone has any Polycom firmwares I could pass them on. ___ --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] which OS would be fine for asterisk
On 8/25/07, Joe Acquisto [EMAIL PROTECTED] wrote: . . . Personally I recommend SuSE Linux. OpenSuSE without the GUI installed will do just fine. If you want to buy SLES that's fine, but I really don't see the value in it. The value would be live support and access to online updates. Courtesy (for the price) of Novell. There are, of course, some differences between OpenSuse and SLES. I've run Asterisk on SLES 9 and SLES 10 without problems. Your View/Mileage May Vary. joe a. With OpenSuSE you get free updates. The support is of no value to me. ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk
On 8/20/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Motherboard with SATA RAID1 support That's a mulit-port SATA controller with RAID in the driver (software). 256 MB RAM Use a little more RAM. digium PRI/E1 card Is there any reason you aren't using Sangoma cards? 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) None. 2. If I use Hardware based RAID 1, what would be the impact to the system? A PCI slot. 3. According to your practical experiance what is the ideal solution among both options? Software RAID works fine. ___ --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] Heavy duty environment - Is TDM2400P suits?
On 8/21/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote: Steve Totaro wrote: You should have no problems. Make sure you put surge protection and ground your POTS lines. It is a small investment. I have had SEVERAL FXO modules die or behave strangely after thunderstorms. I cannot prove it was a surge, but logic would indicate that was the issue. Steve, How are you providing surge protection? I have lost a couple of cards to storms also. I am wondering, did you try to RMA these modules? How did Digium handle it? ___ --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] which OS would be fine for asterisk
. . . The value would be live support and access to online updates. Courtesy (for the price) of Novell. There are, of course, some differences between OpenSuse and SLES. I've run Asterisk on SLES 9 and SLES 10 without problems. Your View/Mileage May Vary. joe a. With OpenSuSE you get free updates. The support is of no value to me. As stated YMVMV. For some people, the ability to have support and to have updates downloaded and installed automatically, (if desired) might be of value. For others, it would have no value or even a negative value. joe a. ___ --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] Polycom firmware download
Thats just sad, I got SIP 2.2 from trixbox now, but still we need to have some sort of place at least for ourselves to download this stuff. Looking for boot loader now. On 8/25/07, Andrew Joakimsen [EMAIL PROTECTED] wrote: On 8/25/07, Al lists [EMAIL PROTECTED] wrote: Hi, I'm trying to use Polycom 330 and apparently it needs latest firmware (SIP 2.2.0). I dont have access to polycom site to download and was wondering if any of you guys have it. Thank you! Best idea is to ask your reseller. I am not aware of a community site with Polycom firmwares. If you wish I think I know the person who created http://spc.pifiu.com and if anyone has any Polycom firmwares I could pass them on. ___ --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
Re: [asterisk-users] which OS would be fine for asterisk
Joe Acquisto wrote: . . . The value would be live support and access to online updates. Courtesy (for the price) of Novell. There are, of course, some differences between OpenSuse and SLES. I've run Asterisk on SLES 9 and SLES 10 without problems. Your View/Mileage May Vary. joe a. With OpenSuSE you get free updates. The support is of no value to me. As stated YMVMV. For some people, the ability to have support and to have updates downloaded and installed automatically, (if desired) might be of value. For others, it would have no value or even a negative value. joe a. But in all reality, value added features such as support and automatic updates aside, is there really a mainstream flavor of Linux that is better or worse for running Asterisk (or other apps for that matter)? I have had equal luck with all that I have played with (but not heavy load tested). I am bringing up several Fedora Core 7 boxen into production now. Besides a knee jerk reaction that Fedora Sucks, can someone give a real argument as to why I should or should not use it for production? (besides the several MB of yum updates daily, which to me is a good thing). Besides naming a flavor and saying It is the best, can someone add a few statements as to why, which will obviously have to compare the other flavors. Thanks, Steve Totaro ___ --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?
Matthew Fredrickson wrote: 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. References: http://lists.digium.com/pipermail/asterisk-users/2007-January/177404.html There was another thread once upon a time where it was also indicated that there would not be any problems. Possibly the one Steve Totaro mentioned in another reply. The motivation behind this was to resolve echo issues using the newer echo can. When I finally was able to get some HPEC licenses it was easier to just keep using what had proved stable (which was zaptel 1.4 with asterisk 1.2). Unfortunately the system started crashing often. I tested this on different hardware at the same facility and it crashed less often, but it still crashed. Now if it's a miracle that asterisk 1.2 and zaptel 1.4 could ever play nice together, that could explain it. But if there is nothing wrong with it, then I'd say this is a bug, hence why I asked if anyone else was using it in a similar manner. Thanks, Trevor -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more! ___ --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] Gizmo revisited
On Aug 25, 2007, at 3:58 AM, Tzafrir Cohen wrote: Can it be used with any service which is not sipphone.com? Yes IF you use it in a fashion similar to what I describe. All calls go though the Gizmo proxy including calls to/from your secondary account, an extension from my asterisk box in my case. I keep it free by not making any outgoing calls using Gizmo minutes by selecting my asterisk box as the default outbound route, and using my own DID which rings on two * extensions simultaneously so I can pick up the call on a regular handset or on the laptop. I could use any softphone for this functionality in conjunction with asterisk but I like their interface better than the other's I've tried including X- Lite, idefisk, etc. Info on using asterisk for this is at http://www.gizmoproject.com/asterisk.html (The phone itself is based on xten's phone, IIRC, and is thus certinly non-free) It's free to Gizmo subscribers whether they are paying subscribers or not. ___ --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
[asterisk-users] Avaya IPOffice and a SIP trunk to Asterisk
Has anyone successfully setup the Avaya IPOffice 500 with a sip trunk to Asterisk. If so can someone give some config examples? Thanks Rick ___ --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] which OS would be fine for asterisk
. . . Besides naming a flavor and saying It is the best, can someone add a few statements as to why, which will obviously have to compare the other flavors. Thanks, Steve Totaro I'd have to review the entire thread to see if anyone actually claimed any flavor was best, but can point to the subject that just asked for something fine. For my part, I offered my comments without an axe to grind, no skin in the game. But it certainly might be interesting to see if someone has a best and reasons for it. joe a. ___ --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] Gizmo revisited
Tzafrir Cohen wrote: On Fri, Aug 24, 2007 at 10:05:44PM -0500, Carlos Leal wrote: Launched the OS X version of Gizmo after about a year of inactivity, downloaded the update and discovered the new improved Giszmo features Asterisk interoperability by allowing a secondary SIP account to be registered simultaneously. It also allows you to make the routing choice for outgoing calls; your own server or via Gizmo. So far, this is the best SIP softphone I've come across for OS X. It comes in other flavors and I thought I'd mention it as it can be free and I haven't seen it mentioned recently. Can it be used with any service which is not sipphone.com? (The phone itself is based on xten's phone, IIRC, and is thus certinly non-free) I don't believe the current GP phone is based on anything from Counterpath/X-ten. When they originally launched SIPphone, they licensed the X-Pro phone from X-ten, but the Gizmo Project softphone has always been a completely different beast. Later versions include the ability to sign up for one other service provider AS WELL AS Gizmo Project/SIPphone, however I don't believe you can ONLY use it through another provider, and GP does some odd things with the Contact header that are... nonstandard. When REGISTERING with another provider, it uses a Contact header with a GP username/ip, requiring that all calls be initially routed back through GP's servers. They say this is for logging purposes, but it certainly makes for an ugly data path. We actually had to reconfigure our servers just to allow for the GP users who would occasionally register directly. N. ___ --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] Polycom firmware download
Andrew Joakimsen wrote: On 8/25/07, Al lists [EMAIL PROTECTED] wrote: Hi, I'm trying to use Polycom 330 and apparently it needs latest firmware (SIP 2.2.0). I dont have access to polycom site to download and was wondering if any of you guys have it. Thank you! Best idea is to ask your reseller. I am not aware of a community site with Polycom firmwares. If you wish I think I know the person who created http://spc.pifiu.com and if anyone has any Polycom firmwares I could pass them on. Doesn't contain 2.2.0 yet but maybe an email to the admin of the site or waiting a little bit, it will magically appear. http://www.freedomphones.net/polycom/files/spip_ssip_sip_2_0_1_release_note.pdf http://www.freedomphones.net/polycom/files/spip_ssip_sip_2_0_1.zip http://www.freedomphones.net/polycom/files/ Thanks, Steve Totaro ___ --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] Heavy duty environment - Is TDM2400P suits?
Andrew Joakimsen wrote: On 8/21/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote: Steve Totaro wrote: You should have no problems. Make sure you put surge protection and ground your POTS lines. It is a small investment. I have had SEVERAL FXO modules die or behave strangely after thunderstorms. I cannot prove it was a surge, but logic would indicate that was the issue. Steve, How are you providing surge protection? I have lost a couple of cards to storms also. I did not ground them properly (they declined that option in the original sales process) and they were from installations over one or two years ago. Since it was not really Digium's fault, I did not even bother with the RMA process. I may have tried if they were only a couple of months old. I just bought new modules and billed the customer parts and labor. I also sold them and installed proper grounding and surge supression. Thanks, Steve Totaro ___ --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: Matthew Fredrickson wrote: 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. References: http://lists.digium.com/pipermail/asterisk-users/2007-January/177404.html There was another thread once upon a time where it was also indicated that there would not be any problems. Possibly the one Steve Totaro mentioned in another reply. The motivation behind this was to resolve echo issues using the newer echo can. When I finally was able to get some HPEC licenses it was easier to just keep using what had proved stable (which was zaptel 1.4 with asterisk 1.2). Unfortunately the system started crashing often. I tested this on different hardware at the same facility and it crashed less often, but it still crashed. Now if it's a miracle that asterisk 1.2 and zaptel 1.4 could ever play nice together, that could explain it. But if there is nothing wrong with it, then I'd say this is a bug, hence why I asked if anyone else was using it in a similar manner. Thanks, Trevor Just to make sure you saw this reply quoted from Tzafrir, seems to really know his stuff: / If you want to be able to build asterisk 1.2 with zaptel 1.4 you'll need: ln -s ../zaptel/zaptel.h /usr/include/linux/zaptel.h ln -s zaptel/tonezone.h /usr/include/tonezone.h Note: I'm not sure exactly what happens if you run a 'make install' of zaptel 1.2 on top of that later. I guess you should remove those two symlinks manually before you do that. / Thanks, Steve Totaro/ / ___ --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] which OS would be fine for asterisk
Joe Acquisto wrote: . . . Besides naming a flavor and saying It is the best, can someone add a few statements as to why, which will obviously have to compare the other flavors. Thanks, Steve Totaro I'd have to review the entire thread to see if anyone actually claimed any flavor was best, but can point to the subject that just asked for something fine. For my part, I offered my comments without an axe to grind, no skin in the game. But it certainly might be interesting to see if someone has a best and reasons for it. joe a. Joe, My intention was not to imply anything about anybody or anything. I just really want to know if there are solid, definable differences or comparisons. I see so many threads with fanboys who swear by this or that but never provide any objective support of their statements. I figured this thread would wind up going in the same direction, thats all. I have the same questions you do, I am just trying to preempt the fanboys. Thanks, Steve Totaro ___ --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?
Steve Totaro wrote: Just to make sure you saw this reply quoted from Tzafrir, seems to really know his stuff: / If you want to be able to build asterisk 1.2 with zaptel 1.4 you'll need: ln -s ../zaptel/zaptel.h /usr/include/linux/zaptel.h ln -s zaptel/tonezone.h /usr/include/tonezone.h Note: I'm not sure exactly what happens if you run a 'make install' of zaptel 1.2 on top of that later. I guess you should remove those two symlinks manually before you do that. Yup I saw that. I got to the point where I made a script that'll wipe out tonezone.h and zaptel.h out of the various directories they could end up in, and set up the symlinks so asterisk 1.2 would build the zaptel modules. This all for the express purpose of making sure I had the right versions and nothing conflicting. -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more! ___ --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?
Tzafrir Cohen wrote: If you want to be able to build asterisk 1.2 with zaptel 1.4 you'll need: ln -s ../zaptel/zaptel.h /usr/include/linux/zaptel.h ln -s zaptel/tonezone.h /usr/include/tonezone.h Note: I'm not sure exactly what happens if you run a 'make install' of zaptel 1.2 on top of that later. I guess you should remove those two symlinks manually before you do that. It seems to wipe out the symlinks and replace them with actual files... although I've been in the habit of removing them regardless. -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more! ___ --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] Avaya IPOffice and a SIP trunk to Asterisk
Azfhasterisk wrote: Has anyone successfully setup the Avaya IPOffice 500 with a sip trunk to Asterisk. If so can someone give some config examples? Thanks Rick Haven't done it but I have suggestions. Plug in values that seem to make sense on both sides. Turn on SIP debugging on Asterisk, try rebooting the phones and making calls and see what kind of output you get. That debug info has made past integrations go so much faster for me than trying to find someone to hand me the answer. Anyways, once you do the above, if you still cannot figure it out, post the SIP debug stuff to the list. Many more people will be able to make sense of it and help that have never touched an Avaya IPOffice system. Thanks, Steve ___ --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
[asterisk-users] [Fwd: [MailServer Notification]Content Filtering Notification]
This email has violated the PROFANITY. and Quarantine entire message has been taken on 25/08/2007 21.00.49. Message details: Server:MAIL1RELAY Sender: [EMAIL PROTECTED]; Recipient:asterisk-users@lists.digium.com; Subject:Re: [asterisk-users] which OS would be fine for asterisk WTH?!? I see no profanity. Trevor Peirce wrote: Matthew Fredrickson wrote: 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. References: http://lists.digium.com/pipermail/asterisk-users/2007-January/177404.html There was another thread once upon a time where it was also indicated that there would not be any problems. Possibly the one Steve Totaro mentioned in another reply. The motivation behind this was to resolve echo issues using the newer echo can. When I finally was able to get some HPEC licenses it was easier to just keep using what had proved stable (which was zaptel 1.4 with asterisk 1.2). Unfortunately the system started crashing often. I tested this on different hardware at the same facility and it crashed less often, but it still crashed. Now if it's a miracle that asterisk 1.2 and zaptel 1.4 could ever play nice together, that could explain it. But if there is nothing wrong with it, then I'd say this is a bug, hence why I asked if anyone else was using it in a similar manner. Thanks, Trevor Just to make sure you saw this reply quoted from Tzafrir, seems to really know his stuff: / If you want to be able to build asterisk 1.2 with zaptel 1.4 you'll need: ln -s ../zaptel/zaptel.h /usr/include/linux/zaptel.h ln -s zaptel/tonezone.h /usr/include/tonezone.h Note: I'm not sure exactly what happens if you run a 'make install' of zaptel 1.2 on top of that later. I guess you should remove those two symlinks manually before you do that. / Thanks, Steve Totaro/ / ___ --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] rfc3680, reginfo+xml
Olivier, In principle, Registration/Presence and Call-Processing are separate logical functions but for cost or other reasons one could combine them in one software implementation or one physical box. For most parts, Asterisk is the Registrar in a SIP network and therefore maintains the location table. So, whatever entity runs the RFC 3680 notifier function will need access to Asterisk's location table. This is not the real issue though (the access to this table can be easily granted through the API). The answer to whether RFC 3680 should be built inside Asterisk or outside of it really depends on the kind of scalability one is looking for. The scalability here refers to the number of subscriptions per each AoR (notification fan-out). Basically, you don't want to overload Asterisk with excessive number of subscriptions to the extent that they can negatively impact call/media processing. If you compare this with presence, the PUBLISH method allowed the presentity to inform only one entity (the presence server) of its state changes. The presence server was the one that did the drudgery of maintaining individual subscriptions and notifying the watchers. The presence model assumes that there are multiple watchers subscribing to the state of *a* presentity. RFC 3680 is a bit different than this, however. If you are looking to do free-seating using RFC 3680, I'd imagine that you will need only one subscription for each AoR in Asterisk (so you don't really have a 1:n fan-out issue). In that case, it is probably okay if the RFC 3680 'notifier' function was embedded within Asterisk itself. The current RFC 3265 support in Asterisk code should make the job of supporting RFC 3680 a bit easier. Raj From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Wednesday, August 22, 2007 8:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] rfc3680, reginfo+xml Thanks for replying, Raj. Do you think such feature should, ideally, be implemented in Asterisk should it be implemented in a dedicated software (presence ?) ? It seems to me it should, though I'm not aware of many devices using this feature, beside SIP hardphones. Would it be difficult to extend current code to comply with this RFC, when rfc3265 mechanism is already in place ? ___ --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
[asterisk-users] SIP endpoint registeration problem
Hi List; I have a problem when trying to let an SIP ATA endpoint (got it from broadtel company), I am getting the following message: - Registered SIP 'bilal_sip at 0.0.0.0 port 5060 expires 60 I do not know why it takes it 0.0.0.0 while it has an IP address (192.168.8.3). In the sip.conf, the following configuration to the bilal_sip done: [bilal_sip] type=friend context=internal host=dynamic canreinvite=no dtmfmode=rfc2833 In the general context, I did allow=all for the codec and there is not any disallow. The same SIP ATA endpoint can do a successful registeration on an softswitch by just put the IP address of this softswitch instead of the IP address of my Asterisk. Also, why in IAX2 we keep receive registeration messages every periodically while we do not receive that periodic messages in SIP? Who can help? What kind of traces I can do to know the reason for not happening the correct registeration? At the endpoint, it keep blinking which indicates that registeration is failed. Regards Bilal Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --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] IAX2 trunking scalability
So you are using an asterisk box as an E1 gateway. You want to know if switching from not using IAX trunking to using IAX trunking will have any effect? Yes it will lower your bandwidth usage a little. It will not increase the CPU load. If your system can support x calls it will be able to support the same amount of calls. On about 1/2 E1, it shows that bandwith usage has been about halved - i.e. without trunking each G.729 call takes 50 kbps (inbound + oubound) and with IAX2 trunking it takes about half of that (using trunkfreq=40). Which is good! I'm wondering wether anybody already had a IAX2 trunking ON and managed to push 3 E1s worth of traffic without issues. The best thing you can do for your system is add a TC400B card. It will also legally support G723 codec which I think sounds just fine, but will save you a bit more bandwidth. Using the hardware transcoder will greatly increase the number of calls your system would be able to handle. I'm already receiving the calls as g.729, so there is little gain (slightly less bandwith usage, slighly worse sound) in doing g.729 - g.723 transcoding - while doing IAX2 trunking vs NOT doing it seems to half the bandwith requirement. Cheers, Jean-Michel. ___ --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] which OS would be fine for asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Totaro wrote: I am bringing up several Fedora Core 7 boxen into production now. Besides a knee jerk reaction that Fedora Sucks, can someone give a real argument as to why I should or should not use it for production? (besides the several MB of yum updates daily, which to me is a good thing). Besides naming a flavor and saying It is the best, can someone add a few statements as to why, which will obviously have to compare the other flavors. Hi Steve, I've run most operating systems on various boxes. - From early RedHats through to Fedora Core, Gentoo, Debian, Mandriva, Suse, CentOS, Ubuntu etc etc. Initially I was quite fond of Redhat stuff, but then they went commercial and I didn't want to pay for support. So I moved to Fedora Core. Unfortunately some of the old Fedora Core installations are now unsupported and even the old yum repositories have stopped providing updates. At the end of the day, the problem I see with Fedora is that they do things slightly differently from other OSes in the placement of files etc, which can cause headaches you wouldn't see on others. However, there are so many people using Fedora/CentOS/Redhat Enterprise that a quick search of Google will normally reveal the result. Recently I've been installing Mandriva on boxes (simply because it was on the cover of a magazine when I urgently needed a copy of Linux), and have found that once over the initial learning curve it has proven to be stable. I'm also running Debian and Ubuntu on a few boxes, and find them to be stable and standard. They're all pretty much the same with the exception of Gentoo and FreeBSD, which tend to be for the ricers. I won't argue about the fact that you would definitely get more performance out of FreeBSD or Gentoo, but for me the amount of extra work setting up these systems outweighs the performance benefits later on. A lot of the differences between distros comes from their choice of package management systems. Once you've used urpmi, yum, up2date, apt-get etc a few times it doesn't really make too much difference which one you're using. One thing that bit me with Mandriva though was that they asked me how secure I wanted the box to be at the start. Normally I set up boxes with maximum security and the absolute least amount of software possible, then add what I need. So, I chose the most restrictive security level. Unfortunately (for me) this meant that it would run cron jobs to change the contents of files and ownerships and disallowed most network communications. Once I fixed that it was fine. The other one that has bitten me a couple of times is SELinux on Fedora, which has resulted in some incredibly strange errors that took rather a long time to find. If I have a problem on a Fedora system that doesn't seem to have a logical answer, I'll quite often disable SELinux for a moment to see if that fixes it. Obviously when that is the problem, you can turn SELinux back on and create rules to allow it to function as you expected. So, is there a best distro? Not really, it depends on what you want out of a system, how much work you are willing to put into each machine, and how much time you want to spend doing maintenance. Here's my opinions: Fastest: Gentoo/FreeBSD Easiest: Fedora/Redhat EL/CentOS Most Stable (for me): Debian/Ubuntu/Mandriva - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG0K10DQNt8rg0Kp4RAtzEAJ9+FFgTPjf5CQYxJ0ZE3wNUb81LZwCgv3Dv jdOJ4Trfa0VCY5gYNOunxgU= =uFvm -END PGP SIGNATURE- ___ --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] IAX2 trunking scalability
Jean-Michel Hiver wrote: So you are using an asterisk box as an E1 gateway. You want to know if switching from not using IAX trunking to using IAX trunking will have any effect? Yes it will lower your bandwidth usage a little. It will not increase the CPU load. If your system can support x calls it will be able to support the same amount of calls. On about 1/2 E1, it shows that bandwith usage has been about halved - i.e. without trunking each G.729 call takes 50 kbps (inbound + oubound) and with IAX2 trunking it takes about half of that (using trunkfreq=40). Which is good! I'm wondering wether anybody already had a IAX2 trunking ON and managed to push 3 E1s worth of traffic without issues. I used to do it, but its a while ago. (Before iax2 got some more fixes) The trick was to keep the trunks small (like 40 per trunk, just make multiple), this should no longer be needed. Cpu utilisation with trunking should be lower than without trunking. zoa The best thing you can do for your system is add a TC400B card. It will also legally support G723 codec which I think sounds just fine, but will save you a bit more bandwidth. Using the hardware transcoder will greatly increase the number of calls your system would be able to handle. I'm already receiving the calls as g.729, so there is little gain (slightly less bandwith usage, slighly worse sound) in doing g.729 - g.723 transcoding - while doing IAX2 trunking vs NOT doing it seems to half the bandwith requirement. Cheers, Jean-Michel. ___ --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
Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
Dear Mr Galvin, Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --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
[asterisk-users] Asterisk Speechphone/Mandi
Anyone on the list have a service from Speechphone called Mandi? Have you been able to set up SIP directly with Speechphones Server? If so, would you like to share how you did that? TIA ___ --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
[asterisk-users] PRI cards, Digium vs. Sangoma (was: Re: Saftware RAID1 or Hardware RAID1 with Asterisk)
Andrew Joakimsen wrote: On 8/20/07, Vidura Senadeera [EMAIL PROTECTED] wrote: digium PRI/E1 card Is there any reason you aren't using Sangoma cards? This sounds as if Sangoma was the only serious manufacturer of PRI cards. What are your reasons for not using a Digium card? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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
[asterisk-users] Chan-capi Fedora 7
Has anyone had any success with chan-capi on Fedora 7? If so can you advise what you did? Thanks in advance! ___ --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] Restart status
On Saturday 25 August 2007 11:15, Steve Totaro wrote: Probably not much help, but if you rarely issue commands such as this, hit the up arrow a few times. Steve, The issue is that I am attempting to restart asterisk from external scripts due to certain pbx conditions. I need to have a two step approach: 1. Issue a restart gracefully 2. If x minutes go by and and all channels have not cleared then issue a restart now Each of these items might not know of the other so it would be helpful for item two to know that asterisk is in a restart gracefully mode. Ron ___ --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] which OS would be fine for asterisk
On Thu, Mar 15, 2007 at 06:24:54PM -0400, Steve Totaro wrote: I am bringing up several Fedora Core 7 boxen into production now. Besides a knee jerk reaction that Fedora Sucks, can someone give a real argument as to why I should or should not use it for production? (besides the several MB of yum updates daily, which to me is a good thing). Which Fedora 7 exactly? The one originally released with kernel 2.6.21-1.3194 or the current one with 2.6.22.4-65? Besides naming a flavor and saying It is the best, can someone add a few statements as to why, which will obviously have to compare the other flavors. An obvious problem with Fedora is that while it is being maintained, packages there change rapidly. So in order to get bug fixes, you'll have to get new features (and potentially more bugs as well). And after a period which is not so long, it stops being maintained at all, and you have no source for bugfixes. -- Tzafrir Cohen icq#16849755jabber:[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
Re: [asterisk-users] which OS would be fine for asterisk
Matt Riddell wrote: Steve Totaro wrote: I am bringing up several Fedora Core 7 boxen into production now. Besides a knee jerk reaction that Fedora Sucks, can someone give a real argument as to why I should or should not use it for production? (besides the several MB of yum updates daily, which to me is a good thing). Besides naming a flavor and saying It is the best, can someone add a few statements as to why, which will obviously have to compare the other flavors. At the end of the day, the problem I see with Fedora is that they do things slightly differently from other OSes in the placement of files etc, which can cause headaches you wouldn't see on others. Exactly. I had some difficulties on Fedora as well (can't remember what kind of problem it was - something about zaptel I think) while it just worked for me on Debian or CentOS. (@Steve: So Fedora sucks and Debian is the best ;-) However, there are so many people using Fedora/CentOS/Redhat Enterprise that a quick search of Google will normally reveal the result. While I'm curious if there is a best OS for Asterisk it probably boils down to the simple rule: Use whatever OS you are familiar with and stick to it. If you're used to Debian then CentOS is a bit different too. Unless someone can prove whatever OS is best for Asterisk I'd recommend to use a mainstream distribution. Although I have compiled Asterisk on MacOSX myself this wouldn't be my first choice for a production server - mainly because the whole file system layout is so different and there isn't really an integrated package management. A lot of the differences between distros comes from their choice of package management systems. Once you've used urpmi, yum, up2date, apt-get etc a few times it doesn't really make too much difference which one you're using. Right. But once you need a more complex set of software tools it's a great timesaver to know what the packages are called on a system and what's in there. A word on SuSE: To my impression YaST is an essential part of it. On the one hand I like it but on the other - well, you can shoot yourself in the foot. It tries to be smart and parse all kinds of /etc/* files and doesn't always do a good job. Setting up a DHCP server with some classes and pools for example is almost a piece of cake on Debian. On SuSE it's more like this: Um, I could edit /etc/dhcpd.conf directly but then the next time someone edits the settings with YaST they'd really mess things up - without even knowing. I'm so glad nobody in this thread has argued for using Windows. ;) (It doesn't even come with an ssh client! You really feel like your hands are tied.) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Stable-Stable Asterisk
On Sat, Aug 25, 2007 at 02:02:14AM -0400, Matt Florell wrote: On 8/24/07, Jay R. Ashworth [EMAIL PROTECTED] wrote: Here's a secondary question (and Matt, I *do* plan to get around the damned corner to one of your meetups one of these days :-): Just how easy is it to roll back to the older release when the feces hit the fan? Seems like making that simple would be pretty important? (Context: my boss is about to tip on playing with Asterisk, finally..) It's really just a matter of recompiling Asterisk. Libpri-1.4 and zaptel-1.4 seem to work fine with Asterisk 1.2 as well so in a pinch you may not have to recompile those. Most 1.2 dialplans will work just fine with 1.4, so as long as you delete the modules between builds: rm -f /usr/lib/asterisk/modules/* So it's not svc asterisk stop rm /appl/asterisk ln -s /appl/asterisk1.4.11 /appl/asterisk svc asterisk start ? Note to the release manager... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --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] Restart status
Ron Joffe wrote: The issue is that I am attempting to restart asterisk from external scripts due to certain pbx conditions. I need to have a two step approach: 1. Issue a restart gracefully 2. If x minutes go by and and all channels have not cleared then issue a restart now Each of these items might not know of the other so it would be helpful for item two to know that asterisk is in a restart gracefully mode. Maybe something like this: asterisk -rx 'restart gracefully' Then constantly monitor asterisk -rx 'core show uptime seconds' And if that does not drop to let's say less than 30 seconds: asterisk -rx 'stop now' sleep 1 killall -9 asterisk asterisk Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Stable-Stable Asterisk
On Sat, Aug 25, 2007 at 08:46:10PM -0400, Jay R. Ashworth wrote: On Sat, Aug 25, 2007 at 02:02:14AM -0400, Matt Florell wrote: On 8/24/07, Jay R. Ashworth [EMAIL PROTECTED] wrote: Here's a secondary question (and Matt, I *do* plan to get around the damned corner to one of your meetups one of these days :-): Just how easy is it to roll back to the older release when the feces hit the fan? Seems like making that simple would be pretty important? (Context: my boss is about to tip on playing with Asterisk, finally..) It's really just a matter of recompiling Asterisk. Libpri-1.4 and zaptel-1.4 seem to work fine with Asterisk 1.2 as well so in a pinch you may not have to recompile those. Most 1.2 dialplans will work just fine with 1.4, so as long as you delete the modules between builds: rm -f /usr/lib/asterisk/modules/* So it's not svc asterisk stop rm /appl/asterisk ln -s /appl/asterisk1.4.11 /appl/asterisk svc asterisk start Well, here's a reason for you to start using module embedding :-) -- Tzafrir Cohen icq#16849755jabber:[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
[asterisk-users] Davide Marcellan is out of the office.
I will be out of the office starting Sun 26/08/2007 and will not return until Sun 02/09/2007. Salve sono in ferie, ma leggerò la posta con calma. Per problemi urgenti contattare il centralino @work al 041 5841212 Saluti davide ___ --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] Restart status
On Saturday 25 August 2007 21:03, Philipp Kempgen wrote: Maybe something like this: asterisk -rx 'restart gracefully' Then constantly monitor asterisk -rx 'core show uptime seconds' And if that does not drop to let's say less than 30 seconds: asterisk -rx 'stop now' sleep 1 killall -9 asterisk asterisk This sounds like a feature waiting to be coded. Something like show restart status Anyone have experience with setting up a bounty for a developer to make a change like this ? Ron ___ --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] which OS would be fine for asterisk
What Digium is using is rpath, RHEL /Centos On 8/25/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Matt Riddell wrote: Steve Totaro wrote: I am bringing up several Fedora Core 7 boxen into production now. Besides a knee jerk reaction that Fedora Sucks, can someone give a real argument as to why I should or should not use it for production? (besides the several MB of yum updates daily, which to me is a good thing). Besides naming a flavor and saying It is the best, can someone add a few statements as to why, which will obviously have to compare the other flavors. At the end of the day, the problem I see with Fedora is that they do things slightly differently from other OSes in the placement of files etc, which can cause headaches you wouldn't see on others. Exactly. I had some difficulties on Fedora as well (can't remember what kind of problem it was - something about zaptel I think) while it just worked for me on Debian or CentOS. (@Steve: So Fedora sucks and Debian is the best ;-) However, there are so many people using Fedora/CentOS/Redhat Enterprise that a quick search of Google will normally reveal the result. While I'm curious if there is a best OS for Asterisk it probably boils down to the simple rule: Use whatever OS you are familiar with and stick to it. If you're used to Debian then CentOS is a bit different too. Unless someone can prove whatever OS is best for Asterisk I'd recommend to use a mainstream distribution. Although I have compiled Asterisk on MacOSX myself this wouldn't be my first choice for a production server - mainly because the whole file system layout is so different and there isn't really an integrated package management. A lot of the differences between distros comes from their choice of package management systems. Once you've used urpmi, yum, up2date, apt-get etc a few times it doesn't really make too much difference which one you're using. Right. But once you need a more complex set of software tools it's a great timesaver to know what the packages are called on a system and what's in there. A word on SuSE: To my impression YaST is an essential part of it. On the one hand I like it but on the other - well, you can shoot yourself in the foot. It tries to be smart and parse all kinds of /etc/* files and doesn't always do a good job. Setting up a DHCP server with some classes and pools for example is almost a piece of cake on Debian. On SuSE it's more like this: Um, I could edit /etc/dhcpd.conf directly but then the next time someone edits the settings with YaST they'd really mess things up - without even knowing. I'm so glad nobody in this thread has argued for using Windows. ;) (It doesn't even come with an ssh client! You really feel like your hands are tied.) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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
Re: [asterisk-users] Gizmo revisited
On 8/24/07, Carlos Leal [EMAIL PROTECTED] wrote: Launched the OS X version of Gizmo after about a year of inactivity, downloaded the update and discovered the new improved Giszmo features Asterisk interoperability by allowing a secondary SIP account to be registered simultaneously. It also allows you to make the routing choice for outgoing calls; your own server or via Gizmo. So far, this is the best SIP softphone I've come across for OS X. It comes in other flavors and I thought I'd mention it as it can be free and I haven't seen it mentioned recently. Yes, I agree, this is a darn good free SIP client with a few features such as recording, playing WAV files (good for podcasting, or if you want to irritate people you call with laugh tracks or something). I don't find it better than X-Lite, but about the same. If I had to choose one client, it would probably be Zoiper, because it does SIP and IAX well (or at least I've had good results with it). Zoiper also has a very small footprint on the GUI. /r ___ --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] Polycom firmware download
At 13:29 8/25/2007, Al lists wrote: Thats just sad, I got SIP 2.2 from trixbox now, but still we need to have some sort of place at least for ourselves to download this stuff. Looking for boot loader now. Which version? http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html#download http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip430.html#download On 8/25/07, Andrew Joakimsen mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: On 8/25/07, Al lists mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: Hi, I'm trying to use Polycom 330 and apparently it needs latest firmware (SIP 2.2.0). I dont have access to polycom site to download and was wondering if any of you guys have it. Thank you! Best idea is to ask your reseller. I am not aware of a community site with Polycom firmwares. If you wish I think I know the person who created http://spc.pifiu.comhttp://spc.pifiu.com and if anyone has any Polycom firmwares I could pass them on. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com--http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usershttp://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
