[Asterisk-Users] RE: [on-asterisk] containers, virtualization, and high availability
I've been trying out asterisk on xen myself. One thing to note for anyone that is experimenting that by default the Xen kernel runs at 100hz. To use ztdummy you need a 1000hz source so you need to recompile the dom0 and domU kernels with 1000hz. If remember correctly you also need to set CONFIG_CRC_CCITT=y in the kernel as well. I have not tested the service migration but it does sound interesting. I'm interesting in trying to get DRDB to work between machines. Thanks John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, March 02, 2006 4:58 PM To: TAUG Subject: [on-asterisk] containers, virtualization, and high availability I've been meaning to try out migrating a running Xen machine running asterisk from one computer to another to see if there's any call interruption. My rudimentary ping/ssh tests have been successful, showing no interruption of service. Can you migrating a running container from one computer to another with Solaris 10? I know it's supported in VMWare ESX($$$), and Xen. The feature I'm really looking foward to in Xen is something called lock-step execution -- a technique which allows you to have two or more machines running the same instructions and have one immediately step in for another if the other crashes or has an underlying hardware fault. Along with that comes replayability, so that you can go back in time and see what the machine was doing just before the failure. Sweet! Cheers, Simon P. Ditner On Thu, 2 Mar 2006, Paul Nash wrote: running x number of virtual Asterisk servers on one physical Linux server to a SAN, I assume that you're thinking of Linux running on VMWare running on Linux. VMWare have an enterprise product (not sure if it's hit the streets yet) that is similar to IBM's mainframe VM supervisor. Very lightweight, partitions the machine, loads onto bare metal. Almost no overhead. If you want to do the same thing for free, look at Xen, which has a similar approach. The guest operating system has to understand Xen (which makes for great performance), but both Linux and NetBSD have Xen ports. You may have to hack up some digium/sangoma drives for Xen to present virtual cards to the VMs, but that shouldn't take very long. Solaris is a fine enterprise OS, but is *very* resource-hungry. paul - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: [on-asterisk] Brainstorming dual-core and Asterisk
I believe you can assign processors in vmware, and xen as well. So you could probably do something funky like that to try to reduce load. The only thing that probably becomes difficult is trying to manage physical hardware between virtual machines. John -Original Message- From: Jim Van Meggelen [mailto:[EMAIL PROTECTED] Sent: Thursday, March 02, 2006 7:47 PM To: 'TAUG' Subject: [on-asterisk] Brainstorming dual-core and Asterisk Let me run something that's been floating about in my noggin by everyone: Given that Asterisk does not make use of dual core CPUs or dual processors, I was contemplating whether running Asterisk in two (or more) VMWare sessions on a system might actually allow for more total performance. For example, set up one VM to handle incoming lines, echo cancellation and all sets, and then set up the other VM to handle VoIP, including transcoding. A bit kludgy, to be sure, but would VMWare allow for both cores/CPUs to be more fully utilized? Very possibly not practical, but it's been floating about my head for a bit and I figured I'd send it out into the ether to see what thoughts might come back. So . . . thoughts? Jim. -- Jim Van Meggelen [EMAIL PROTECTED] http://www.oreillynet.com/pub/au/2177 A child is the ultimate startup, and I have three. This makes me rich. Guy Kawasaki -- -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.1.1/272 - Release Date: 01/03/2006 - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Xen
Hey All, Ive been working on trying to get asterisk to play nice under Xen and Ive run into a bit of a road block. Im not using any hardware stuff only ztdummy. First I had issues getting ztdummy to work but that was solved by recompiling the xenU kernel to have CONFIG_CRC_CCITT=y which it was missing. Now ztdummy is seen fine in the virtual machine though when I run zttest I get horrible results, so I havent even bothered to test any config yet. This results are the same for the host machine and the virtual machine. --- Results after 21 passes --- Best: 0.00 -- Worst: -800.012207 -- Average: -799.998233 Anyone know if this is normal, is there a patch/other kernel options or something that is needed so that ztdummy works correctly? Thanks John ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Xen
Get to answer my own post. I found an article that talks about the need for 1000HZ timing in the kernel for ztdummy to work properly. Xen's kernel builds default to 100HZ just like 2.4 kernels. I changed the values to 1000 in /xen-3.0.0/linux-2.6.12-xenU/include/asm-xen/asm/param.h and xen/include/asm-x86/config.h And recompiled and now I get decent results --- Results after 51 passes --- Best: 99.987793 -- Worst: 99.829102 -- Average: 99.972953 Any other xen gotchas I should know about? Thanks John From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Cianfarani Sent: Thursday, February 09, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and Xen Hey All, I've been working on trying to get asterisk to play nice under Xen and I've run into a bit of a road block. I'm not using any hardware stuff only ztdummy. First I had issues getting ztdummy to work but that was solved by recompiling the xenU kernel to have CONFIG_CRC_CCITT=y which it was missing. Now ztdummy is seen fine in the virtual machine though when I run zttest I get horrible results, so I haven't even bothered to test any config yet. This results are the same for the host machine and the virtual machine. --- Results after 21 passes --- Best: 0.00 -- Worst: -800.012207 -- Average: -799.998233 Anyone know if this is normal, is there a patch/other kernel options or something that is needed so that ztdummy works correctly? Thanks John ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Xen
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Thursday, February 09, 2006 6:33 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk and Xen On Thu, Feb 09, 2006 at 06:23:26PM -0500, John Cianfarani wrote: Get to answer my own post. I found an article that talks about the need for 1000HZ timing in the kernel for ztdummy to work properly. Xen's kernel builds default to 100HZ just like 2.4 kernels. I changed the values to 1000 in /xen-3.0.0/linux-2.6.12-xenU/include/asm-xen/asm/param.h and xen/include/asm-x86/config.h And recompiled and now I get decent results Does the RTC code of ztdummy work in Xen? How would I check that? From this http://project-xen.web.cern.ch/project-xen/xen/howto_slcXen.html under Time Sync it seems like the RTC is handled by the host machine by default. Thanks John ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls fading in and out
What model phones are you using? Ive noticed this before on spa841s. John From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nora Lavelle Sent: Friday, February 03, 2006 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Calls fading in and out Hi again everyone ! Was wondering if anyone had any pointers on how to debug voice quality issues in asterisk. Ive got a user who either cant be heard on her phone calls (outgoing and incoming) and today someone that called her said that her voice was coming in and out. Any pointers or suggestions are appreciated! Thanks so much again ! This list has been so helpful to me. Nora Lavelle ___ --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] Asterisk Dial Failover
Ryan/Jonathan, Couple quick questions regarding your setup? Do you operate this in a strictly master/slave setup? Do you have anything(mon/ha's internal status/monitor options) that actually monitors the asterisk process (to determine if it is hung). Or is it only with total box failure to you fail over? Do you use something to sync config/vm/cdr? Rsync/unison? Thanks John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan All, I have an Asterisk system that sends PSTN calls to an OpenSER system to be routed. I have a command like this in my extensions.conf: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) There's actually two OpenSER systems for redundancy. I'm trying to find a way to have Asterisk attempt to route the call to one OpenSER system, and if it's down, fallback to another. Any first thoughts on how to achieve this? I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups are broken. If I issue a series of Dial commands, such as this: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) ... what seems to happen is that when proxy1 is down, Asterisk waits the full 20 seconds before returning control. Also, This 20s includes the time is takes for the other end to answer, so if I put a small value of say 5s in there, the dial command will probably give up before someone answers at the other end. Neither is workable. Asterisk SHOULD be able to distinguish between a TRYING and no response. In the event it gets no TRYING response to a dial command within a specified timeout it should return control and flag an error. If on the other hand it does get a TRYING response (and maybe a RINGING too) it should continue to wait until the 20s has expired. I can't use dynamic DNS (ie putting two A records for a hostname in DNS) because Asterisk reads the extensions.conf on startup and also seems to cache what the host maps to on startup. Subsequent calls to the host always return the same IP address. But... in general... how have people implemented this? Help appreciated! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Synthesized Voice for Asterisk
Cepstral has some pretty decent quality voices at like $29 they don't break the bank. https://www.cepstral.com It also can integrate directly into asterisk I believe. Hope that helps John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dakota Sent: Friday, December 09, 2005 7:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Synthesized Voice for Asterisk Are there any cool free software I can use to create automated voice message greetings for my PBX? I want to customized some of my messages, however prefer to use a standard voice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Missing smp kernel package in Asterisk 1.2installation...
Have you done this? ln -s /lib/modules/`uname -r`/build /usr/src/linux-2.6 ln -s /lib/modules/`uname -r`/build /usr/src/linux to make sure the sources are linked correctly in the /usr/src directory? John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Burd Sent: Thursday, November 17, 2005 7:08 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Missing smp kernel package in Asterisk 1.2installation... Hello there, I've just downloaded Asterisk 1.2 into my RedHat Enterprise Linux machine and got the following problem when I tried to compile zaptel: You do not appear to have the sources for the 2.6.9-22.ELsmp kernel installed. However, according to rpm -qa, I do have the following packages installed in my system: kernel-smp-2.6.9-22.EL kernel-smp-devel-2.6.9-5.EL Am I doing anything wrong? If so what shall I do to fix this problem? In fact, I've never experienced this issue in the previous version of Asterisk. BTW, I'm only planning to use my server to handle voip calls ... do I still need to compile zaptel? Thank you so much for your help, Leo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] free dids on goiax.com
Why not just ask for a small one time payment $1 or something from a credit card, or paypal, or something along those lines so you would have someway to trace back to an abuser. John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Simpson Sent: Tuesday, October 18, 2005 3:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] free dids on goiax.com GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can unrestrict outdial again, but this is the problem with free.. there is always someone that will abuse it. If anybody has any ideas on how to keep the abuse down let me know. The best ideas I have now is to only allow a certain amount of calling per month, add velocity checking, and somehow put some accountability into the sign up process to keep the prank callers and multiple account abusers away. yours, Matthew Simpson GoIAX -- www.goiax.com TxLink -- www.txlink.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Failover question
What if asterisk was to start have more options for failover from an application perspective? Eg. Some form of heartbeat between the two servers. Within the heartbeat it could pass registration information and call information between servers. (Not sure if this is somehow possible already) So if you were to use that with something like HA/clustering the backup server would always know what calls / registrations were active. Thanks John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Thursday, June 30, 2005 11:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Failover question I think this is a weak point in asterisk. It doesn't even have a means of email notification if IAX or SIP registration fails. This would need to be added to the list of priorities. But I'm not sure who to address to. Most phone are controlled by their own software interface and have the ability to re-register at certain intervals (ex. every hour) but that is not much of a help and or acceptable if you are left without phone for one hour. So this is not an asterisk related problem but the software interface that controls the phones. The simplest solution would be to add email notification in such software and/or fail-over IP if one fails. But that is up to the hardware manufacture to come up with this solution. All asterisk could provide is just an email warning that certain phone failed to register. -- #Joseph On Thu, 2005-06-30 at 15:19 +0300, Mohamed A. Gombolaty wrote: Dear All, I am using Linux-High Availability between two Asterisk servers, everything is fine but I do have one problem with this, When a server fails and the other assumes the ip address and start asterisk on server 2, the ip phone must re-register themselves again, otherwise the phones won't ring. Does anyone have Ideas of how to overcome this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk failover solution
Then you need to cluster your DB servers so they aren't a point of failure. Thanks John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vamsi Pottangi Sent: Thursday, June 30, 2005 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk failover solution Use Realtime and host the database on a separate machine. This should solve most of your problems. ~Vamsi On 6/30/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote: Dear All, I am using Linux-High Availability between two Asterisk servers, everything is fine but I do have one problem with this, When a server fails and the other assumes the ip address and start asterisk on server 2, the ip phone must re-register themselves again, otherwise the phones are dead. Does anyone have Ideas of how to overcome this. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Failover Design
Hello All, Ive been investigating and playing with asterisk to see how it would work out as a small-medium business pbx to handle mostly interoffice/branch communication and a possibly communication out to pstn in later stages of implementation. (All communication would be VoIP internally with possibly 4 pstn lines at the HQ with either a TDM04B or spa3000s and 1-2 pstn lines across 7-8 branches with spa3000s). Still evaluating my options about which devices will be best. One thing Ive been trying to figure out with little luck is regarding true failover/redundancy design and would like some suggestions from the list. Ive looked through http://www.voip-info.org/tiki-index.php?page=Asterisk+High+Availability+Solutions and I can understand the high availability solutions. Though there seems to be several possible solutions for failover but none really seems to be the recommended way to implement. Most seem just another possible way. Since my proposed setup is fairly small 20-40 voip phones max maybe 12-20 pots lines my main goal is: - 2 asterisk boxes, Hopefully possible to be in different sites - If box1 fails calls will stay up (except those that are pots connected off that box) - Rest of the VoIP phones will re-register with box 2. (I noticed the cisco 79xx have backup proxy so this could be handled by the phones). Or is there some way to for asterisk box 1 to pass registrations to box 2 as well? What about the spa3ks how would they handle re-registration to a different ip? - Configs / Voicemails mirrored across both servers (probably easy done with rsync) I took a brief look at proxy with SER but I believe there are a few things I possibly dont understand. - How does SER determine if an asterisk box is down? Or it SER only for load balancing? - Looks like I would need another box or 2 specifically to do SER and then well that seems to become a point of failure? What are some ways people implement real failover with asterisk? Or if there are some other resource that I should look at? Please feel to point out if Im way off the mark in anything or my expectations. Thanks for your time John Cianfarani ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HT-488 vs. SPA-3000?
I've been looking into this more for a small deployment. Is it at all possible to put some other line adapter to amplify/increase signal before it goes into the spa3k? Something like these? (Found these after a quick google search) http://www.harriscomm.com/catalog/default.php?cPath=1141_47_167 http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=1503item=5784527504; rd=1ssPageName=WDVW Would love to know if anyone has tried them. Thanks John Cianfarani --- I had exactly the same experience with the SPA-3000. Too bad too since it's nice device...if it were 6 db hotter. I also installed a TDM-400, which was better in a lot of ways but not perfect. When I rebuild my server I ended up simply call forwarding my POTS lines to a DID provided by an ITSP. This has been the best as far as quality is concerned. If my DSL line goes down I simply defeat the call forwarding on the main line and answer an analog phone for a while, or call forward to me cell. Michael On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote: Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... I've not played with the ht488, but I believe others have posted this device does not provide access to the pstn-fxo port. The spa3k does provide that access (if you want it). Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. The spa3k works fine with asterisk as many have posted. However, once in awhile it does act a little strange in two different ways: 1. the spa3k will sometimes interpret some voices as tones which cause a little disturbance to any conversation going on. It is sort of like the old telephony talk off that existed years ago. Doesn't happen all that often and seems to be more sensitive to female voices based on my one-year of experience. 2. sometimes it seems to operate in half-duplex mode, where if you try to talk at the same time as the other end is talking, the other end won't hear you. Neither one of those have been all that objectionable to me, but they happen and others have posted roughly the same issues. I've not heard of anyone that has found a way to minimize those two issues. The down side of the spa3k right now is that Cisco bought the company and there likely won't be much advancement of the code until after the ownership (and development efforts) are sorted out by both companies. (The same kind of product delays has been seen with their Linksys purchase, as well as when other companies are bought/sold.) Its fairly common knowledge that ex-Cisco folks started Sipura for the sole purpose of selling the company for a hugh profit. Their success in accomplishing that objective could only be measured in terms of producing Sipura products that had at least some acceptance of those products by end users. With those previous objectives accomplished, how will Cisco handle the Sipura products in the future? (It's any- one's guess at this point since Cisco also has at least some track record of mismanaging purchased companies for whatever reason.) From an internal Cisco strategic perspective, they now own the assets that can make a major dent in the mass-market end-user voip product arena, and hopefully they'll take that in a positive direction. Given the price of the spa3k, I don't have any issue with purchasing more of them right now. Excellent choice for the one-to-three pstn-fxo market space. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850
Does anyone know what the reason why Dell servers cause so many problems for the digium hardware? Better question any Dell models that don't have any these problems with the digium hardware? Thanks John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoa Sent: Tuesday, June 21, 2005 3:40 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ee1000 Ethernet in Dell 1850 Hmm, i dont think thats the reason they dont recommend the dell server. The problems with the ee1000 kernel module are easily resolved, compile the module into the kernel. Zoa, Andres wrote: Digium's site now lists the Dell 1850 as a potential problem server, as it uses the intel ee1000 Ethernet chipset (as do a majority of servers in the market!). To my knowledge, ALL dell servers with Gigabit interfaces now use the same chipset. Does this mean the Digium cards can't be used in Dell servers unless you disable the onboard ethernet? I don't want to disable the onboard interface, as I use the IPMI management facility for lights-out management. I have a 2850 that doesn't have any audio problems (the reason that I contacted Digium in the first place), so I'm wondering if Digium are simply guessing at problems. Does anyone know anything specific about the supposed incompatibilities with the ee1000 kernel module? I am not sure where you got that chipset reference but all our PowerEdge 1850s come with: Ethernet controller: Intel Corp. 82541GI/PI Gigabit Ethernet Controller ...and they work fine with the TE410. There seems to be an ever-growing list of situations where you can't use the Digium cards. This is a concern to me. ___ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX with shaw cable not going through
Rogers does the same thing all you need to do is a DHCP release (or the equivalent in your FW). I had similar issues (not asterisk related) since I have a pix fw and it has no option to do a dhcp release. John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Saturday, June 18, 2005 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX with shaw cable not going through On Sat, 2005-06-18 at 14:16 -0600, Joseph wrote: I just changed the from DSL to Shaw Cable (static IP) configure the firewall but now asterisk I can not register with FWD nor VoipJet calls going out. I am using IAX with FWD Did I missed to change a setting? I don't think there is any though. I am on shaw extreme connection; I just talked shaw tech. and they are not blocking any port - I was told. So why IAX will not register with FWD and calls to VoipJet are not getting connected. I've boot my asterisk backup server to ADSL and everything is working FWD, VoipJet. Short story: The only thing I've done differently is I've spoofed MAC address on the firewall on an external port - eth0 to get the same IP address from Shaw, but I don't see how that could make a difference. Long Story: Shaw has those new Cable Modem - Motorola SURFboard SB5100 that once configured to an IP address with one firewall it will retain that MAC address of that first firewall for about 4-hours. When I first experimented that Cable Modem I've connected my backup firewall and the Modem retained that MAC address. So in order to connect the second firewall and get the same IP address, I need to spoof the MAC address of the first firewall or wait 4-hours. So I went with the second solution but I don't see how that could make a difference, the only way to tell is to wait 4-hours to remove the spoof MAC address from the firewall. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Issue with IAXy in Canada?
Although I have not really tried much IAX stuff yet I am on Rogers in Ontario, Canada. So if you need someone to do a bit of troubleshooting with you I'd be glad to help. The only ports I know that Rogers blocks are 139 and the 1433. They don't block 25 (as I run a mail server and everything gets through) Though they do plan to block 25 soon. Rogers does does some application throttling but this is mostly for bit torrent/kazza etc type traffic. John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Obaid Siddiqui Sent: Monday, June 06, 2005 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Issue with IAXy in Canada? I tested IAXy with my asterisk server in US, using both DSL. It was working fine. I gave it to my friend who was traveling to Canada. He is saying that it is not working with Rogers Cable. It is getting busy tone after 20-30 seconds. Is it possibly port blocking? or any other problem. Do somebody has any port blocking issues with IAXy's in Canada. *please* reply if you any clue. Obaid. - Original Message - From: Dean Collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 06, 2005 8:36 PM Subject: RE: [Asterisk-Users] OT: Please comment on Dvorak's troll Brian, interesting comment. Can you provide more information? Do I understand from reading that this was settled outside of court therefore no precedent was made? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian Litzinger Sent: Monday, 6 June 2005 7:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Please comment on Dvorak's troll On Mon, Jun 06, 2005 at 03:03:49PM -0600, Colin Anderson wrote: The Slashdot guys are choked 'cause he was right about Intel and the Macs. While I agree he sensationalizes I was looking for opinions on whether there might be something to this ISP/ILEC attempt to control VoIP traffic. It's of concern to me, since I have rolled out a substantial portion of our company's PSTN traffic over the public Internet, and I am in Canada, where everything is legislated and legislation is largely determined by lobbyists. My default argument against any regulation is that I would not comply simply because my company's VoIP traffic is tantamount to traffic on our internal PBX and we can do whatever we want with it. However, I don't want to have to be forced into doing something goofy like running IAX over port 80 because some upstream provider is looking for a revenue grab. I'm just wondering if anyone in the community has considered what if and what would be a meaningful response, either technologically, legally, or socially. Encryption comes to mind. Also, Dundi's RFC perhaps addresses some of these issues by obsfucating centralized directories and might be modified to encompass port number in order to force bad ISP's play wack-a-port. I can muse about a real world experience. I worked for company that distributed data via the Vertical Blanking Interval (VBI) of standard television signals. The company had local and nationwide converage through local and superstations including over-the-air and cable. One day we starting getting calls from subscribers in New York that they were no longer getting data. A cable operator they had come to understand our signal and blocked it with equipment at his head end. I found it interesting he choose to block the signal and then wait for us to come calling. We did talk with him and he had intentionally blocked our signal and was waiting to negotiate for his share of our proceeds. It was an interesting area of contention where previous contracts to carry did not make clear what was to happen in this situation. The New York cable company basically claimed their contractual obligation was only to the active video period. In other words, their 'right-to-carry' (which they paid for) only covered the active video period, rather than the entire video signal. This area of uncertainty was clarified in later contracts. -- Brian Litzinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
RE: [Asterisk-Users] [OT] stupid firmware question...
Out of curiosity since you brought this up, what are the chances of having a simple device that can do a sip/sccp/h323/etc conversion/proxy to IAX2 much like the way you can get ATA adapter for analog phones. At first glance I would think something like this could help out with having analog or sip phones working behind a NAT gateway. I could see a use for small remote offices with maybe 1-5 phones where putting extra * box might not be feasible. John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, March 02, 2005 4:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [OT] stupid firmware question... On Wed, 2005-03-02 at 14:24 -0600, Chris Wade wrote: I know this is a really stupid question, but I just have to ask... Where would I start if I wanted to try and develop my own firmware for a particular phone. Namely, I want to try and 're-write' the SIP firmware for Cisco 7940's. Any ideas? -Chris PS: [* put on flame suit *] why won't any of the phone manufacturer's just open-source the firmware for their phones? [* ducks head back inside gopher hole just before a giant fireball hits *] Simple, it is likely that they put the code for the codecs in the firmware. They have to ensure you pay your patent license fees. Granted it is easy enough to get around once you have a firmware image for the ciscos, but it doesn't let them off the hook on how they must guard the code. If Cisco didn't have access to G729, most people would laugh at their offerings and quickly jump to another product. If you where going to rewrite the firmware, you should try and make an IAX2 firmware for it. In fact, you should try and contact Cisco and see if you could partner with them to write the code for an IAX2 firmware and get help from them. They would be able to incorporate the codecs they have licenses for and charge money for the product. Might be interesting. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems Starting Asterisk - FOP AM Portal
Hello All, Im new to the list and the whole voip server side. Im trying to setup Asterisk to just do internal dialing, no access out to the pstn is required/wanted at the moment. Im running Fedora Core 3 with Cisco 7960s phones (running SIP 6.3). Ive set it up following these guides: http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3 http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.2.pdf Versions: asterisk-1.0.6 AMP-1.10.006 mpg123-0.59r zaptel-1.0.6 spandsp-0.0.2pre10 I configured zaptel with the ztdummy option since I dont have any fxs/fxo modules. (as per http://sourceforge.net/forum/forum.php?thread_id=1188938forum_id=414452 ) Asterisk seems to start fine but the FOP op_server.pl doesnt seem to want to start. Ive tried running it by hand as the asterisk user but it doesnt spew any errors, and I cant find any log files that would help me troubleshoot this issue. Ive searched different archives and google but cant find much related to this problem. Any help or suggestions would be appreciated. ERROR MSG STARTING FOP SERVER -bash: line 1: 23134 Killed /var/www/html/panel/safe_opserver - The FOP's server (op_server.pl) could not start! Please correct this problem - Thanks John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems Starting Asterisk - FOP AM Portal
Thanks I set the option for selinux to disabled in the /etc/sysconfig/selinux config and that seems to have fixed the issue. Thanks for your help John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker Sent: Tuesday, March 01, 2005 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problems Starting Asterisk - FOP AM Portal John Cianfarani wrote: Asterisk seems to start fine but the FOP op_server.pl doesn't seem to want to start. I've tried running it by hand as the asterisk user but it doesn't spew any errors, and I can't find any log files that would help me troubleshoot this issue. I've searched different archives and google but can't find much related to this problem. Please read the following: http://fedora.redhat.com/docs/selinux-faq-fc3/ and if you still have issues post to the amportal mailing list and/or Help forum. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
