Re: [asterisk-users] Debugging a SIP / AudioCodes Problem
Andrew Joakimsen wrote: Audiocodes blatently violates the GPL... dont use their gear. On 11 Feb 2007 19:11:51 -, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have 2 identical AudioCodes MP-112s. They have the same config except for the SIP usernames/passwords and the device IP. The configs in extension.conf and sip.conf are also identical. On one box, when I pick up the phone, I get a fast busy and the logs/debug show an automatic hangup. On the other device, I can make calls without a problem. I can even call the phone that can't make a call. Any ideas where I could start to figure out where the problem is? ___ --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 Indeed step 1: throw the audiocodes in the trash step 2: buy real hardware -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
Matt wrote: For my next servers I'll be ordering Arima mainboards I think and assemble the things myself again. I'm thinking about going with MBX.The problem with this is we were replacing an IBM server that had the same problem. The Dell Techs all assured me this is how all the new machines work. This coupled with the continual changing SATA drive controller (or so it seems) are about to drive me away from Dell for servers. *sigh*. Worse part is, I just found the original e-mail from Dell in which they assured me that it would NOT share IRQs. Needless to say, I've fired off an e-mail to that guy, his supervisor, my sales guy at Dell, and my boss. Right now I have a very upset customer. ___ --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 try booting with APIC and ACPI disabled? -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
Andres wrote: try booting with APIC and ACPI disabled? Thats right. I have never seen a shared IRQ with Dell servers using APIC. A RHL ES3 by default enables APIC so I have never even had to fiddle around with it. Andres. ___ --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 I've seen it on the PowerEdge 830's, I even spoke with Mark about it at one point. -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
Matt wrote: try booting with APIC and ACPI disabled? ARG. You're going around in circles as bad as the Dell tech. If the IRQs are being shared and setup in BIOS.. no amount of booting with ACPI disabled, or trying to set the IRQ within Linux is going to help.At this point, the BIOS is overriding and throwing the device onto that IRQ. If the device is physically being put onto an IRQ by the BIOS, then no amount of software manipulation is going to change it. If I am wrong, someone please let me know. As I understand it, ACPI will just let you use the extended (15) IRQs.. which is really just IRQ sharing... the problem here is that the BIOS still has them setup as shared IRQs. ___ --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 You've hit the point where I demanded an RMA and refund for my 830. -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] New user question (X100P)
David Ruggles wrote: I'm trying to set up a simple test box to start developing with Asterisk. I've got a Dell GX150 with two X100P cards. I've downloaded, printed out and read through most of TFOT. I've also done a lot of Internet searching. I'm getting this error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) Based on the searching I've done, it seems like the problem must be shared IRQ issues. I've gone in the BIOS and disabled everything I can but I can't stop the sharing completely. One X100P is shared with the built-in video and the other X100P is shared with a serial controller. (I disabled both serial ports) My question is this: (in three parts) 1) Are my research and assumptions accurate? Does this seem to be an IRQ issue? 2) If I have to build another system to prevent the IRQ problem, can anyone recommend hardware (just for a simple test box right now) 3) Is it worth keeping the X100Ps or should I get a TDM400P like used in TFOT, or something else? (again just for testing) If there are other things I should check first please let me know!! TIA! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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 The X100P is going to leave you fairly unsatisfied with your Asterisk experience. It lacks an on-card timing interface, and real X100P's haven't been made in quite awhile (several years). You may or may not get caller ID, you may or may not get two way audio. I know it's a bit more but you're better off with the TDM. Good luck! -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] New user question (X100P)
Andrew D Kirch wrote: David Ruggles wrote: I'm trying to set up a simple test box to start developing with Asterisk. I've got a Dell GX150 with two X100P cards. I've downloaded, printed out and read through most of TFOT. I've also done a lot of Internet searching. I'm getting this error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) Based on the searching I've done, it seems like the problem must be shared IRQ issues. I've gone in the BIOS and disabled everything I can but I can't stop the sharing completely. One X100P is shared with the built-in video and the other X100P is shared with a serial controller. (I disabled both serial ports) My question is this: (in three parts) 1) Are my research and assumptions accurate? Does this seem to be an IRQ issue? 2) If I have to build another system to prevent the IRQ problem, can anyone recommend hardware (just for a simple test box right now) 3) Is it worth keeping the X100Ps or should I get a TDM400P like used in TFOT, or something else? (again just for testing) If there are other things I should check first please let me know!! TIA! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200[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 The X100P is going to leave you fairly unsatisfied with your Asterisk experience. It lacks an on-card timing interface, and real X100P's haven't been made in quite awhile (several years). You may or may not get caller ID, you may or may not get two way audio. I know it's a bit more but you're better off with the TDM. Good luck! I should note I committed two sins in this post. 1. Tzafrir, I didn't see your reply (blame my inability to use my mail reader, and accept my humble apology) 2. the TDM400P comes with installation support from Digium, so you know you're going to be able to get it working correctly. -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] MeetMe w/ SMP was (My Phone Review- Large Scale Corp Deployment.)
Jamin W. Collins wrote: Steve Edwards wrote: I'm running CentOS 4.4, Asterisk 1.2.13 on HP DL380's. My application is mostly meetme conferences being created and closed all day long. Peak load is around 200 SIP calls. I was crashing 7 to 10 times a day until I booted a non-SMP kernel. I haven't had a crash since. Meetme does not play well with SMP. Interesting, I've been running asterisk (v1.2.10) on an SMP system (dual Xeon 2.66Ghz) with several MeetMe conference rooms for quite a while now. The server has a current uptime of 24 days (moved the system to a new UPS). I have not experienced the crashes you reference. Are either of you running hyperthreading? -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] PCI, PCI-X and PCI-e -- Server / Interface Card Selection
Ex Vitorino wrote: Hello List, The main issue is server selection regarding PCI bus connectivity for Asterisk solutions. Most offerings on HP Proliants, something I've been looking into, include PCI-X and/or PCI-e expansion slots. I know PCI-e is totally different from PCI and PCI-X so, for now, that's not an issue. However, regarding PCI and PCI-X, and after googling for a while and checking wikipedia and whatnot, I'm still not clear on my main issue: Can one use a PCI interface card in a PCI-X slot ? If so, under what conditions ? (ex: 3.3v cards only ? PCI-X bus speed is brought down ? what ?) The objective would be to use Digium's echo cancelling PRI and BRI cards and/or beroNet's BRI cards on, for example, a Proliant ML350 G4 or G5 containing PCI-X slots -- would such combination be technically feasible ? If not, where are you guys getting servers for your PCI based solutions ? Can anyone shed some light into my doubts ? Pointers, documentation, experiences ? The second, kind of attatched issue, is associated to the growing PCI-e buses in the current servers. I know Sangoma already has a PRI card for PCI-e. What about Digium ? Other PRI, BRI manufacturers ? Thanks in advance and regards to all, -- Ex Vito ___ --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 The rule of thumb is if it fits you can use it unless it doesn't work, there are few that won't (Creative's soundcards being an example of ones that don't) -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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: Autoreply: [asterisk-users] Snom MWI
Austin Denyer wrote: PLONK! Regards, Austin. and you sent this to a public list? you're a fucking idiot. -A ___ --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: Autoreply: [asterisk-users] Snom MWI
Austin Denyer wrote: [EMAIL PROTECTED] wrote: Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED] Cordiali Saluti Giuseppe Parlato Area Network mailto:[EMAIL PROTECTED] PLONK! Regards, Austin. I'd like to apologize profusely to anyone offended by my previous post. It was not intended to go to Asterisk-Users but to Austin in private. I will graciously accept any and all flames for my unacceptably poor behavior. Andrew ___ --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] NuFone, please send the log file
Ronald Wiplinger wrote: Dear NuFone, Without misunderstanding I ask you again, please send the log file and pay back my money! Not following this request results in the assumption that NuFone is cheating and I will post this info every hour on more Internet places. This should help that other people will not trap into a cheating company. bye Ronald ___ --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 Wow, that was productive, either never do that again or I'm invoicing you for the time it took me to read it write a response telling you what a moron you were for posting it in the first place, and then deleting it and making sure that the poor hard drive it was stored on was shot humanely and put out of its misery. In other words take it off-list. This is not the people-who-bitch-about-nufone (for values of nufone that equate to any provider BroadVoice anyone?), or #nufone-sucks on some IRC channel. Quite honestly (and I've noted before) that NuFone seems to have a business model of catering only to clued customers. I am still curious as to the eventual outcome (their long-term survival), but you have aptly demonstrated above why you yourself aren't a customer. Get a clue, grow one, buy one EBay but quit spouting this crap on a help list, for I tell you that there is no help for you and not because NuFone screwed you. Andrew ___ --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] NuFone, please send the log file
Ronald Wiplinger wrote: Dear NuFone, Without misunderstanding I ask you again, please send the log file and pay back my money! Not following this request results in the assumption that NuFone is cheating and I will post this info every hour on more Internet places. This should help that other people will not trap into a cheating company. bye Ronald ___ --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 I'm going to note two more issues I've just found with this post. 1. this is a specifically NON-Commercial list (your post is commercial) 2. you have threatened to post it to further such lists and forums where it is not desired (your post is being made in bulk) I therefore must determine you have posted UCE/UBE and you are a spammer. ___ --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] NuFone, please send the log file
Michael Workman wrote: So Nufone Screwed ya I feel Sorry... W Take your Lumps... Cut Your Losses and Get on with Life Your not the only one Nufone Screwed They Screwed me Out of $3,000.00 How do you figure this at 2.9c/min? Andrew ___ --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] NuFone, please send the log file
Michael Workman wrote: I am not talk about Call Time.. They Screwed me by Me Hiring them to consult on setting up server and they took the money and never did the work -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew D Kirch Sent: Tuesday, July 11, 2006 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] NuFone, please send the log file Michael Workman wrote: So Nufone Screwed ya I feel Sorry... W Take your Lumps... Cut Your Losses and Get on with Life Your not the only one Nufone Screwed They Screwed me Out of $3,000.00 How do you figure this at 2.9c/min? Andrew Aha! this is rational and makes sense, Thanks! Andrew ___ --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] NuFone, please send the log file
C F wrote: I find this hard to believe, half a truth is a whole lie. First you just say the screwed you out $3k, not saying how, letting everyone assume thru phone service, then you change the story, you lied before, how do we know you are saying the truth now? In any case it doesnt make sense anyhow, why would you pay $3k for just setting up a server, when others here on the list will do it for much less. Also they never did the work? just took the money? thats an accusation that doesn't make sense. On 7/11/06, Michael Workman [EMAIL PROTECTED] wrote: I am not talk about Call Time.. They Screwed me by Me Hiring them to consult on setting up server and they took the money and never did the work -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew D Kirch Sent: Tuesday, July 11, 2006 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] NuFone, please send the log file Michael Workman wrote: So Nufone Screwed ya I feel Sorry... W Take your Lumps... Cut Your Losses and Get on with Life Your not the only one Nufone Screwed They Screwed me Out of $3,000.00 So it doesn't mess up the flow of reading. How come? I prefer to reply inline. What do you do instead? No. Do you like top-posting? That aside, he clearly does not state how he was screwed. I questioned it as I felt that it deserved elaboration (if you manage to screw yourself to the tune of 3 grand for a $2.9c/min phone service, I can only ditto his recommendation for KY). I find his answer both rational and reasonable and in no way evasive. Andrew ___ --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] NuFone, please send the log file
Ronald Wiplinger wrote: Andrew D Kirch wrote: Michael Workman wrote: So Nufone Screwed ya I feel Sorry... W Take your Lumps... Cut Your Losses and Get on with Life Your not the only one Nufone Screwed They Screwed me Out of $3,000.00 How do you figure this at 2.9c/min? Andrew That is easy to calculate: 3,000 US$ times your zip code times the phone number you are calling times 2.9cents/5 seconds divided by the Social Security number of the called party ... Or how does NuFone calculate that? But hey, just look at the log file, hmm, didn't we start here? WHERE ARE THE LOG FILES Thanks for all the encouraging funny answers. I go now to 7-eleven to buy some candies, ... bye Ronald ___ --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 You sir are a candidate for the USENET news.admin.* school of flamewars! Just send me a large box of unmarked non-sequential large bills via US Post and you too can have your very own USENET approved asbestos suit*, and signed copy of How to flame the entire Internet because no one likes you and life isn't fair.. Andrew *not a real asbestos suit, USENET brand asbestos suits do not cause cancer, void in Puerto Rico or where prohibited. PS. Ronald, GET A LIFE. ___ --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] NuFone, please send the log file
Michael Workman wrote: Very Simple. I hired JerJer to Have a SER and Asterisk setup with Acounting... JerJer told me to Talk to Shido6 and he would do it... He told me it Would cost me $3000 and he do it. He demanded the $ first and never did the work. I have said on several occasions that I think that Jeremy's a bit of a jerk, and tends to berate the less than clueful. I think I have demonstrated the same attributes tonight (which I make no apology for), HOWEVER, Greg (shido6) is honest. I'd like some further background information on this, as it is perhaps an issue that can be straightened out. (note well the following: I'm a third party with no stake in this and that could really care less but I'll lend a hand if I can.) Andrew ___ --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] NuFone, please send the log file
Michael Workman wrote: I paid Via Paypal... But they did nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell (NZ) Sent: Tuesday, July 11, 2006 9:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] NuFone, please send the log file -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Michael Workman wrote: Very Simple. I hired JerJer to Have a SER and Asterisk setup with Acounting... JerJer told me to Talk to Shido6 and he would do it... He told me it Would cost me $3000 and he do it. He demanded the $ first and never did the work. How did you pay? - -- Cheers, Matt Riddell I'd personally vouch for Greg, however you may seek remedy via PayPal. Andrew ___ --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] NuFone, please send the log file
Michael Workman wrote: Well that Make me Note that I will never do Biz with you That is if you personally vouch for Greg That is up to you, I'll note that there are many (most) who consider my integrity beyond reproach. And I have personally offered to help resolve this situation. You may choose to work with me or not, to what end I don't care, but my offer stands. Good luck, Andrew ___ --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] NuFone suggests to use Vonage!!!!
Ronald Wiplinger wrote: Part of a conversation with NuFone. It is untrue, that they do not answer, but if than: Quote: 3. change your attitude towards customers!! No, if you don't like it, go use Vonage. End of quote! I had always problems with these people. bye Ronald ___ --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 To some extent I see your point and have been on the receiving end of one of Jeremy's tirades. I've since decided that NuFone is an interesting study in whether your business can survive with only clueful customers. Andrew ___ --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] Dell PowerEdge 830
Hans Witvliet wrote: On Fri, 2006-07-07 at 08:28 -0500, Cavanna, Richard wrote: I am thinking of using this machine to run asterisk. Has anyone had any experience with this machine? Have one here. Get enough mem, default deliverey is with 256 MB. Draw back is however it needs PC2-4200/5300 ECC DDR2 It has only four mem-banks, and needs to use it in pairs Other point is, it has just one 5V pci-slot, and two 3.3V slots When ordering the SATA-version, all powerconnectors (except the one for the mandatory cd/dvd-drive) are sata-power connectors. No spare molex needed foor the tdm400 Hans right but how did you fix the IRQ conflicts? ___ --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] Dell PowerEdge 830
Jason Adams wrote: I actually just deployed this server for a customer with only 4 users. It worked great. The bios control over the IRQ's isn't the best. I would definitely recommend against an integrated NIC. Other than that is works great for them... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cavanna, Richard Sent: Friday, July 07, 2006 9:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dell PowerEdge 830 I am thinking of using this machine to run asterisk. Has anyone had any experience with this machine? Thanks for any info. ___ --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 I'm using a 24xxp and just gave up on this with a dell raid controller, what all are you using as far as hardware. I've already got the RMA to ship it back but if I can make it work I'm all for it. Andrew ___ --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] intel vs amd motherboards
C F wrote: cat /proc/cpuinfo on amd: cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 47 model name : AMD Athlon(tm) 64 Processor 3200+ stepping: 2 cpu MHz : 2000.000 cache size : 512 KB fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni lahf_lm bogomips: 4025.55 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc cat /proc/cpuinfo on intel: cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 6 model name : Intel(R) Pentium(R) D CPU 2.80GHz stepping: 2 cpu MHz : 2800.353 cache size : 2048 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 6 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm pni monitor ds_cpl vmx cid cx16 xtpr lahf_lm bogomips: 5609.03 processor : 1 vendor_id : GenuineIntel cpu family : 15 model : 6 model name : Intel(R) Pentium(R) D CPU 2.80GHz stepping: 2 cpu MHz : 2800.353 cache size : 2048 KB physical id : 0 siblings: 2 core id : 1 cpu cores : 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 6 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm pni monitor ds_cpl vmx cid cx16 xtpr lahf_lm bogomips: 5600.89 On 7/6/06, Fabio [EMAIL PROTECTED] wrote: Hi CF, please could you to include CPUs specs, thanks in advance. Fabio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de C F Enviado el: Jueves, 06 de Julio de 2006 04:32 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] intel vs amd motherboards I have recently build 2 machines, one with an Intel Pentium Dual Core CPU, and one SATA HDD, and the other with a single AMD 64 bit CPU, and a RAID 1 w 2 SATA HDD. Both costed the same even though the AMD had 2 HDDs. Here are the show translations from both: Intel Dual Core machine: pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 517 -17 ulaw - 2 - 1 2 2 1 517 -17 alaw - 2 1 - 2 2 1 517 -17 g726 - 2 2 2 - 2 1 517 -17 adpcm - 2 2 2 2 - 1 517 -17 slin - 1 1 1 1 1 - 416 -16 lpc10 - 3 3 3 3 3 2 -18 -18 g729 - 4 4 4 4 4 3 7 - -19 speex - - - - - - - - - - - ilbc - 3 3 3 3 3 2 618 - - AMD 64 bit machine: pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 313 -12 ulaw - 3 - 1 2 2 1 313 -12 alaw - 3 1 - 2 2 1 313 -12 g726 - 3 2 2 - 2 1 313 -12 adpcm - 3 2 2 2 - 1 313 -12 slin - 2 1 1 1 1 - 212 -11 lpc10 - 3 2 2 2 2 1 -13 -12 g729 - 4 3 3 3 3 2 4 - -13 speex - - - - - - - - - - - ilbc - 4 3 3 3 3 2 414 - - This shows that the AMD 64 bit is worth much more than just the price
Re: [Asterisk-Users] OLD PA system.
Thomas Kenyon wrote: I need to be able to connect an old PA system to an asterisk box, which basically works as a couple of amplifiers taking an analogue phone signal and playing whatever it produces out of some speakers. There is no on-hook state in the whole setup. Obviously If I just connect the input to a port on an ATA, I'll just get a dialtone played through the speakers. Can anyone think of a way I can attach it, so that people can call an extension that will play through the speakers? The nearest thing I can think of, is getting the * box to call it as an extension plugged into an ATA, then while it's calling plug the line in and transfer it to a non-announcing conference (or similar). This is an undesirable approach for many reasons. (mostly the requirement to set it all up again in the event of a server reset). I have a small pile of ATAs I can play with (well, there's already an atcom 468 in there with a spare port, there's a spare PAP2, a spare Cisco 186 or I could borrow an SPA3k from my home setup) If they'd be of any help. I originally thought about using the sound card, till I saw what was already in place. (The client doesn't like spending money). If the client won't spend $30 on an SB Live Value, Fuck 'em you don't want them as a client. You're a PBX consultant not a bean counter. Andrew ___ --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] WCTDM-24xxp woes
I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is any registerable incoming volume from these lines. I've been running them at rxgain = 25 (zapata.conf) to make the audio audible, however this creates poor call quality issues (static and distortion) on most calls, and audio garble in voicemails. Fxotune fails for every line with Could not fill input buffer I've tried changing PCI slots, played with echo settings, and done everything else I can think of to make this card play nice to no avial. Anyone with solutions or ideas, your input will be greatfully appreciated. Thank you in advance. -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key At http://www.2mbit.com/~trelane/trelane.key Key fingerprint = B4C2 8083 648B 37A2 4CCE 61D3 16D6 995D 026F 20CF ___ --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] WCTDM-24xxp woes
On Sun, June 4, 2006 5:10 pm, Gonzalo Servat wrote: On 6/4/06, Andrew D Kirch [EMAIL PROTECTED] wrote: I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is any registerable incoming volume from these lines. I've been running them at rxgain = 25 (zapata.conf) to make the audio audible, however this creates poor call quality issues (static and distortion) on most calls, and audio garble in voicemails. Fxotune fails for every line with Could not fill input buffer I've tried changing PCI slots, played with echo settings, and done everything else I can think of to make this card play nice to no avial. Anyone with solutions or ideas, your input will be greatfully appreciated. Hi, I'm using one of these cards with 3 quad modules (4 x FXO, 8 x FXS) and I didn't have to touch the rxgain/txgain (0.0 for both). This is how it should be set. If you need to adjust these settings, then you have another problem. I know you tried changing PCI slots, but have you looked at /proc/interrupts to see if the card is sharing an IRQ with another device? Regards, Gonzalo. ___ --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 I should have noted in the above pastage that the card is sitting by itself on IRQ20 (good ol' ACPI) 20:8421859 IO-APIC-level wctdm24xxp I wholeheartedly agree that rxgain/txgain should _NEVER_ have to be set that high. -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key At http://www.2mbit.com/~trelane/trelane.key Key fingerprint = B4C2 8083 648B 37A2 4CCE 61D3 16D6 995D 026F 20CF ___ --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] Snom firmwares suck --additional datapoint to consider
Dr. Michael J. Chudobiak wrote: Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. Just curious - what configuration issues did you have in mind? - Mike Replacing it with a Catalyst? Andrew ___ --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] No rings before auto attendant
Dan Elder wrote: Hi all, been searching not finding an answer to this, although I'm guessing it's absurdly simple... I just hooked up a T1 to our * box (1.2.0), which had been using POTS lines via a channel bank.. Now when I call the new T1 circuit, there are no rings, the Autoattendant just picks up right away.. Any clue on how to make it ring twice before getting picked up? I tried immedate=no and some other zapata.conf tweaks, but nothing seems to work. I also tinkered with adding some wait statements before the 'answer' but only heard silence then the attendant..sorry for such a basic question.. I guess I'm just not punching in the right search terms in my queries. Thx in advance ___ --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 exten = context,1,Wait(5) ___ --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] Are my expectations too high?
Derek Lee-Wo wrote: Get an FXO card with hardware echo cancellation. I use the Sangoma A20002D (four FXO ports with echo cancellation). It definitely costs more, but the hardware echo cancellation makes a huge difference in call quality! Software echo cancellation doesn't really work... With this card, would you say your audio quality is identical to that of an analog phone connected directly to the PSTN? I'm trying to understand if I should expect some audio degradation when going through Asterisk. Everything causes degredation. Yes the card is a codec so it will not be perfect. Many modern telephones are also codecs. The question you're trying to ask is is this degradation noticeable? The answer to that is obviously no. There is an application out there that might benefit you in a situation like this. Go look through your zaptel source tree for fxotune and see if it cant possibly correct some of the problem you're having. After you've tried that I'd suggest posting a .wav file of exactly what you're hearing (there are recording functions in asterisk) and maybe someone that's heard the degredation you're hearing before will reply. Andrew ___ --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] Career Opportunities
Douglas Garstang wrote: I've been working with Asterisk for a little while now, and have been looking recently at my next career opportunity. It seems from searching the various job sites that the predominant VOIP technology is not the applications-based open source approach we took, but Cisco, with a really heavy emphasis on the networking (ie network engineer) aspect. If you do a job search for (VOIP or Voice-over-IP or IP telephony) and you mostly get results for network engineers with lots of Cisco experience. Because Asterisk is a feature-rich solution, my emphasis has been on providing and developing features, applications and systems (ie asterisk, Linux), redundancy, customisation, programming, as well as overall architecture, especially in relation to SIP (and working around all those Asterisk HA limiations!). There of course has also been a networking component as well. On a side note, apparently my current employer tried a Cisco solution before I came along, and I hear all the time how absolute crap is was. Is that how people who have used Asterisk feel about Cisco? Is Cisco that bad? Is it lacking in features? I know we investigated a Sylantro solution and I remember that was pretty nasty. Anyway, based on the absolute dominance of Cisco it almost seems like what I have been doing with Asterisk has been a complete waste of time from a career perspective. I'm not sure how I can use Asterisk to my advantage over Cisco here. Having moved to a small city and working for a CLEC makes finding work outside the city even tougher. I'm wondering if I should have stuck with Unix or SAN admin that was I doing before, and if my recent work with Asterisk has jeopardised my current experience status with my previously used skills. Anyway, just my 2c worth. other opinions welcome. 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 Doug, I currently work as an Asterisk solutions provider and can tell you that you're absolutely ahead of the curve. I might suggest two options: 1. start a small asterisk deployment/consultation firm 2. see if there is one already in your area I've yet to run into a contract where I was bidding against a Cisco solution that I didn't win. Some things to consider (in non geek-speak): 1. Asterisk is based on the Linux Software platform. Linux is free (discuss the free as in beer concept and even refer the customer to the Cathedral and the Bazaar (I generally recommend specific essay's that are pertinent, instead of forcing them to read the whole thing. I also find it effecting to give them links to ESR's website outlining the advantages of the Open Source model.) In comparison Cisco Call Manager is currently based on an unsupported or extended support (depending on the version of the server OS) Microsoft Operating system which has had it's source code leaked and is notoriously insecure. Because of this Cisco Call Manager's code base is being migrated to Linux. That means that in a year or two when you want bright shiny new Call Manager features you are looking at a total migration which may result or require that you throw away some or all of the hardware you purchased from Cisco. 2. Cisco's phone have since the beginning supported Skinny/MGCP. These phones and call manager are now being migrated to support SIP. Asterisk has supported SIP nearly since it's inception. 3. Asterisk is an open source open standards platform, there is no vendor lock-in. If your customer doesn't like you he or she may toss your PBX in the street and purchase any PBX which supports the SIP standard without losing his phones. Speaking of those phones they may use any phone they choose. Most non-Cisco phones are $100-200 less than the equivalent Cisco phones. -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] 911 @ Zap Channel Breakin
Mark Coccimiglio wrote: Ok here is one for you. I know we all do the this for 911: exten = _911,1,Dial(Zap/1/911) exten = _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a least cost routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is durring a 911 call test if the Zap channel is Available (probably using ChanIsAvail() ) to test the line. IF the channel is in use I want to barge in with an announcment saying that the line is needed for an emergency and the call we be disconnected. Then immediately drop the call capture the line so noone else can use it, wait about 5 seconds for the telco to clear the far end and place the 911 call. Is this possible? Thnaks Mark C [EMAIL PROTECTED] FWD: 293625 ___ --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 I seem to recall a script that did something similar on voip-info.org. Look through their E911 stuff, perhaps it's still there. Good luck. -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] FW: Solid-PBX
Leo Ann Boon wrote: Anyone else taken a look at this? Looks exactly like Asterisk (the random source files I browsed all show Digium copyright) but with autoconf. https://developer.berlios.de/projects/solid-pbx/ From the summary page: Solid PBX is a multi-platform Soft Switch Class 5 software targeted for home/corporate/operator level markets ___ --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 Looks like they forked Asterisk. GPL's intact, source appears to be there, no news here, move along. -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] Two asterisk process in one hardware.
Josué Conti wrote: Juan, I think daemons of asterisk is not possible to have two in the same hardware, however can be had two different sessions of asterisk in tty different. Best Regards 2006/4/24, Juan Salas [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hello. Has anybody knows how run two asterisk process in one hardware? (each one with its own configuration?) Thanks Juan Salas. ___ --Bandwidth and Colocation provided by Easynews.com http://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 please look into Xen or VMware -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] Re: HardPhone PlanetVIP-150T - Starts music on Hold and i can't get the call again
Marco Mouta wrote: I forgot to write: When i hangup the call, it hangs correctly! On 4/18/06, *Marco Mouta* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work! I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 Everytime i press the music on hold button it seems that it stops music on hold and starts imediately again. Any one can guess what may be wrong? Best regards, Marco Mouta ___ --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 I don't know if this will address your issue but I'll paste the sip.conf block for my planet 150T, pay particular attention to the fact that I've found the phone misbehaving on re-invite. thus canreinvite=no. [104] type=friend context=local username=104 user=104 secret=xxx callerid=xxx104 mailbox=104 disallow=all allow=ulaw host=dynamic nat=yes notifyringing=yes qualify=yes ;dtmf=inband canreinvite=no Andrew ___ --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] Orative
The weather isn't as good in Indiana. Douglas Garstang wrote: Jeez. Why does every startup in the universe have to be in the bay area. :( -Original Message- *From:* Dean Collins [mailto:[EMAIL PROTECTED] *Sent:* Monday, April 17, 2006 1:00 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Orative Has anyone heard anything about these guys? Anyone seen anything like this? http://www.orative.com/solutions.php It’s seems very cool, basically uses GPRS as a digital overlay on your mobile phone for additional functionality such as presence and IM though I’m sure they have some other functionality (voicemail access, call announce etc) coming down the pipeline. Any thoughts, how hard would it be to build something like this from scratch for the asterisk platform? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ --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] Orative
Fine then you can hammer out the hail damage my car suffered on Friday. Andrew Latham wrote: I resent that, the weather here is wonderful today On 4/17/06, Andrew D Kirch [EMAIL PROTECTED] wrote: The weather isn't as good in Indiana. Douglas Garstang wrote: Jeez. Why does every startup in the universe have to be in the bay area. :( -Original Message- *From:* Dean Collins [mailto:[EMAIL PROTECTED] *Sent:* Monday, April 17, 2006 1:00 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Orative Has anyone heard anything about these guys? Anyone seen anything like this? http://www.orative.com/solutions.php It's seems very cool, basically uses GPRS as a digital overlay on your mobile phone for additional functionality such as presence and IM though I'm sure they have some other functionality (voicemail access, call announce etc) coming down the pipeline. Any thoughts, how hard would it be to build something like this from scratch for the asterisk platform? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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 -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] VERY IMPORTANT(TREAT WITH URGENCY)
Andrew D Kirch
Indianapolis, United States
Good day,
I am Mr. Andrew D Kirch, a native of Indianpolis, United States and I am
an Asterisk Hacker with the
Summit Open Source Development Group. First and foremost,I apologized
using this medium to reach you for a transaction/business of this magnitude,
but this is due to Confidentiality and prompt access reposed on this
medium. Be informed that a member of the #asterisk channel on Freenode who
is well familiar with you gave your enviable credentials/particulars to
me. I have decided to seek a confidential co-operation with you in the
execution of the deal described Hereunder for the benefit of all parties
and hope you will keep it as a top secret because of the nature of this
transaction.
Within the Summit Opensource Development group I work as the Security
Administrator and with the cooperation of other top officials, we have
in our possession a Follow Me script which simply does not work More so,
we are handicapped in the circumstances, as we havn't a clue how to fix
it, hence your importance in the whole transaction.
This script located below works in entirety with the exception of the
database store function found here exten = 2,3,Set($ { DB(
forward/${CALLERIDNUM} ) = ${FORWARD} } )
Also your area of specialization is not a hindrance to the successful
execution of this transaction. I have reposed my confidence in you and
hope that you will not disappoint me. Endeavor to contact me immediately
through my e- mail: to confirm whether or not you are interested in this
deal.
Once again,remember that time is of great essence in this transaction.
I wait in anticipation of your fullest co-operation.
Yours faithfully,
Andrew D Kirch
[Forward]
exten = s,1,Playback(forward/extension-forwarding)
;Extension Forwarding
exten = s,2,GotoIf($[${CALLERIDNUM}300]?s,5)
;since 1xx is the pattern match for internal extensions anything less
than 300 has to be internal so we already know that that is the
extension they are wanting to forward
exten = s,3,Read(CALLERIDNUM,foward/please-ent-exten,3)
;if it's not have the user enter their 3 digit enternal extension
;please enter the extension you want to forward
exten = s,4,SayNumber(${CALLERIDNUM})
exten = s,5,Background(forward/extension-fwd-menu)
;to hear your current extension forward options press 1, to forward your
phone press 2, to cancel your forwarding press 3
exten = 1,1,Set(FORWARD=${DB(forward/${CALLERIDNUM})})
exten = 1,2,NoOp(FORWARD is ${FORWARD})
exten = 1,3,GotoIf($[${FORWARD}0]?100,3)
exten = 1,4,Playback(forward/your-ext-not-forward)
;your extension is not currently forwarded
exten = 1,5,Goto(Forward,s,5)
;back to main menu
exten = 100,1,Playback(forward/your-ext-forward)
exten = 100,2,SayDigits(${FORWARD})
;your extension is currently forwarded to extension
exten = 100,3,Goto(Forward,s,5)
;back to main menu
exten = 2,1,Read(FORWARD,forward/please-ent-exten)
exten = 2,2,NoOp(FORWARD is ${FORWARD})
exten = 2,3,Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD} } )
exten = 2,4,NoOp(forward/${CALLERIDNUM} is ${
DB(forward/${CALLERIDNUM} ) } )
exten = 2,5,Playback(forward/your-ext-forward-saved)
;your extension forward has been saved
exten = 2,6,Goto(Forward,s,5)
exten = 3,1,DBdel(forward/${CALLERIDNUM})
exten = 3,2,PlayBack(forward/exten-forward-cancel)
; your extension forward has been deleted.
exten = 3,3,Goto(Forward,s,1)
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Re: [Asterisk-Users] Hardware timing source for MeetMe
Mike Clark wrote: Will the low cost X100P clones available on ebay provide a solid hardware timing source? Our experience shows that while using ztdummy with no zaptel hardware does allow MeetMe to function, we experience unacceptable levels of delay after four ot five users join the conference. With both TDM400 and Sangoma A101 hardware, we have had 20+ users with no problems. We have a pure VoIP system installed, that has nor PRI or analog lines, but does have a need for MeetMe. If a $15 card will do the trick, we would obviously rather do that than spend a couple hundred bucks for the same thing. This card would not be used for voice, just timing. Thanks, Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those do not have timing interfaces on them that I am aware of. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RFC Follow Me Find Me script
This is a follow/find me script that I can't quite get to work, asterisk
wont save forward/${calleridnum} to AstDB... any comments or thoughts on
how to make this work or change it to work differently are appreciated.
The voice prompts to go with all playback/background extensions are
commented appropriately. I hope this code is of use to some of you and
any help with a perfected version is of course appreciated.
[Forward]
exten = s,1,Playback(forward/extension-forwarding)
;Extension Forwarding
exten = s,2,GotoIf($[${CALLERIDNUM}300]?s,5)
;since 1xx is the pattern match for internal extensions anything less
than 300 has to be internal so we already know that that is the
extension they are wanting to forward
exten = s,3,Read(CALLERIDNUM,foward/please-ent-exten,3)
;if it's not have the user enter their 3 digit enternal extension
;please enter the extension you want to forward
exten = s,4,SayNumber(${CALLERIDNUM})
exten = s,5,Background(forward/extension-fwd-menu)
;to hear your current extension forward options press 1, to forward
your phone press 2, to cancel your forwarding press 3
exten = 1,1,Set(FORWARD=${DB(forward/${CALLERIDNUM})})
exten = 1,2,NoOp(FORWARD is ${FORWARD})
exten = 1,3,GotoIf($[${FORWARD}0]?100,3)
exten = 1,4,Playback(forward/your-ext-not-forward)
;your extension is not currently forwarded
exten = 1,5,Goto(Forward,s,5)
;back to main menu
exten = 100,1,Playback(forward/your-ext-forward)
exten = 100,2,SayDigits(${FORWARD})
;your extension is currently forwarded to extension
exten = 100,3,Goto(Forward,s,5)
;back to main menu
exten = 2,1,Read(FORWARD,forward/please-ent-exten)
exten = 2,2,NoOp(FORWARD is ${FORWARD})
exten = 2,3,Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD} } )
exten = 2,4,NoOp(forward/${CALLERIDNUM} is ${
DB(forward/${CALLERIDNUM} ) } )
exten = 2,5,Playback(forward/your-ext-forward-saved)
;your extension forward has been saved
exten = 2,6,Goto(Forward,s,5)
exten = 3,1,DBdel(forward/${CALLERIDNUM})
exten = 3,2,PlayBack(forward/exten-forward-cancel)
; your extension forward has been deleted.
exten = 3,3,Goto(Forward,s,1)
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Re: [Asterisk-Users] RFC Follow Me Find Me script
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
(top posting to follow previous/keep thread sane)
* The applications DBGet and DBPut have been deprecated in favor of
functions. Here is a table of their replacements:
DBGet(foo=family/key)Set(foo=${DB(family/key)})
DBPut(family/key=${foo}) Set(DB(family/key)=${foo})
If I read this correctly the syntax in column two is the current best
practice for AstDB. It, unless I've missed something below is what I
have used in my script.
Johann wrote:
That looks like the dialplan for Asterisk 1.0.x, The AstDB and other
commands have changed in Asterisk 1.2.x(and CVS HEAD). Check the
UPGRADE.txt in the source code directory of Asterisk to get the details
on all the changes...
--johann
Andrew D Kirch wrote:
This is a follow/find me script that I can't quite get to work,
asterisk wont save forward/${calleridnum} to AstDB... any comments or
thoughts on how to make this work or change it to work differently are
appreciated. The voice prompts to go with all playback/background
extensions are commented appropriately. I hope this code is of use to
some of you and any help with a perfected version is of course
appreciated.
[Forward]
exten = s,1,Playback(forward/extension-forwarding)
;Extension Forwarding
exten = s,2,GotoIf($[${CALLERIDNUM}300]?s,5)
;since 1xx is the pattern match for internal extensions anything
less than 300 has to be internal so we already know that that is the
extension they are wanting to forward
exten = s,3,Read(CALLERIDNUM,foward/please-ent-exten,3)
;if it's not have the user enter their 3 digit enternal extension
;please enter the extension you want to forward
exten = s,4,SayNumber(${CALLERIDNUM})
exten = s,5,Background(forward/extension-fwd-menu)
;to hear your current extension forward options press 1, to forward
your phone press 2, to cancel your forwarding press 3
exten = 1,1,Set(FORWARD=${DB(forward/${CALLERIDNUM})})
exten = 1,2,NoOp(FORWARD is ${FORWARD})
exten = 1,3,GotoIf($[${FORWARD}0]?100,3)
exten = 1,4,Playback(forward/your-ext-not-forward)
;your extension is not currently forwarded
exten = 1,5,Goto(Forward,s,5)
;back to main menu
exten = 100,1,Playback(forward/your-ext-forward)
exten = 100,2,SayDigits(${FORWARD})
;your extension is currently forwarded to extension
exten = 100,3,Goto(Forward,s,5)
;back to main menu
exten = 2,1,Read(FORWARD,forward/please-ent-exten)
exten = 2,2,NoOp(FORWARD is ${FORWARD})
exten = 2,3,Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD} } )
exten = 2,4,NoOp(forward/${CALLERIDNUM} is ${
DB(forward/${CALLERIDNUM} ) } )
exten = 2,5,Playback(forward/your-ext-forward-saved)
;your extension forward has been saved
exten = 2,6,Goto(Forward,s,5)
exten = 3,1,DBdel(forward/${CALLERIDNUM})
exten = 3,2,PlayBack(forward/exten-forward-cancel)
; your extension forward has been deleted.
exten = 3,3,Goto(Forward,s,1)
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- --
Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org
Security Admin | Summit Open Source Development Group | www.sosdg.org
Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2
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Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFEEk+YEzF+JcQGyNIRAg9aAKCS3JcXpuWSVNT/Z25FU2Um3o4TVQCgor0u
48W1AzyAkRr3TCgdHwFxIY8=
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Re: [Asterisk-Users] Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mojo with Horan Company, LLC wrote: This may not be the applicable solution, but if you're not using the mysql config capabilities, add noload = res_config_mysql.so to modules.conf Moj Sharath Chandra wrote: Hi all, I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red Hat linux box( Linux version 2.4.20-8smp). I was able to compile both the software but when i start Asterisk, it exits with the following dump. Error Text Start= [res_crypto.so] = (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup' [res_config_mysql.so]Mar 6 05:18:23 WARNING[12779]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: __stack_chk_fail Mar 6 05:18:23 WARNING[12779]: loader.c:554 load_modules: Loading module res_config_mysql.so failed! End=== Can someone suggest a solution. Regards, Sharath Chandra ___ --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 You may want to update/rebuild mysql-addons, if this still occurs file a bug. - -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEDmo+EzF+JcQGyNIRAuauAJ9ITVC+TVbF7UQLo5vaedYA4clvTwCeOo39 UfwgH91CafwZE2ZESjfrfWE= =KzDV -END PGP SIGNATURE- ___ --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 compile error
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 MBIT Technologies wrote: Hi Guys I have a problem compiling Asterisk 1.2.4. I am getting this error make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Has anyone come across this? ___ --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 You should provide more information, paste a few more lines from the breakage, look specifically for lines beginning with the word error. - -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFD+QzQEzF+JcQGyNIRAvhIAJwOIfxnmWRQ6kpPg2/7pRyqyklJhQCgl7LI 5B+xkAiZOTB2s/HqHFWkNmY= =QVhX -END PGP SIGNATURE- ___ --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] editing errors/typos in rev 2 of The Asterisk Handbook (current version on digium's site)
On Sun, 11 Apr 2004 23:28:05 -0500 Andrew D Kirch [EMAIL PROTECTED] wrote: Apologies if any of these have already been fixed in the working version Page 6 third and fourth lines from the bottom Digium is the solely capable should read Digium is solely capable -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key At http://www.2mbit.com/~trelane/trelane.key Key fingerprint = B4C2 8083 648B 37A2 4CCE 61D3 16D6 995D 026F 20CF apologies and ignore this was targeted at asterisk-doc -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key At http://www.2mbit.com/~trelane/trelane.key Key fingerprint = B4C2 8083 648B 37A2 4CCE 61D3 16D6 995D 026F 20CF pgp0.pgp Description: PGP signature
[Asterisk-Users] Invalid module format in 2.6.5 after running make linux26
[EMAIL PROTECTED] zaptel]# modprobe ztdummy WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format FATAL: Error inserting ztdummy (/lib/modules/2.6.5-1.315/misc/ztdummy.ko): Invalid module format FATAL: Error running install command for ztdummy [EMAIL PROTECTED] zaptel]# uname -a Linux asterisk.sosdg.org 2.6.5-1.315 #1 Fri Apr 9 13:44:11 EDT 2004 i686 i686 i386 GNU/[EMAIL PROTECTED] zaptel]# -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key At http://www.2mbit.com/~trelane/trelane.key Key fingerprint = B4C2 8083 648B 37A2 4CCE 61D3 16D6 995D 026F 20CF pgp0.pgp Description: PGP signature
[Asterisk-Users] Re: Invalid module format in 2.6.5 after running make linux26
On Mon, 12 Apr 2004 18:19:51 -0500 Andrew D Kirch [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] zaptel]# modprobe ztdummy WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format FATAL: Error inserting ztdummy (/lib/modules/2.6.5-1.315/misc/ztdummy.ko): Invalid module format FATAL: Error running install command for [EMAIL PROTECTED] zaptel]# uname -a Linux asterisk.sosdg.org 2.6.5-1.315 #1 Fri Apr 9 13:44:11 EDT 2004 i686 i686 i386 GNU/[EMAIL PROTECTED] zaptel]# Also I found this in my syslog while trying to solve the problem: Apr 12 18:48:45 asterisk kernel: zaptel: version magic '2.6.5-1.315custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.5-1.315 686 REGPARM 4KSTACKS gcc-3.3' Apr 12 18:48:45 asterisk kernel: zaptel: version magic '2.6.5-1.315custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.5-1.315 686 REGPARM 4KSTACKS gcc-3.3' Apr 12 18:48:45 asterisk kernel: ztdummy: version magic '2.6.5-1.315custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.5-1.315 686 REGPARM 4KSTACKS gcc-3.3' -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key At http://www.2mbit.com/~trelane/trelane.key Key fingerprint = B4C2 8083 648B 37A2 4CCE 61D3 16D6 995D 026F 20CF pgp0.pgp Description: PGP signature
