Re: [asterisk-users] g729 on 1.4.10.1
PSTN - g729 requires transcoding at that point. You can however do: G.729 phone - asterisk - G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman Sent: September-18-07 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk VOIP Jobs version 2 Launched!
Great job on the new site... i found some really great people to do some asterisk installs that i needed to have done for clients through your site hope your new site does well! i'll be using your site for anything i have in the future for sure. -- Matt #1 From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Jobs [EMAIL PROTECTED] Sent: Saturday, March 15, 2008 5:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk VOIP Jobs version 2 Launched! Greetings VOIP Job Seekers, We wanted to let you know that we've completed the revamp of Asterisk-Jobs.com. There's not much there now after scrapping version 1.0 of the site, but we expect many postings to come soon. Keep an eye on the site for the latest in Asterisk and related VOIP employment. http://www.asterisk-jobs.com Thanks, Asterisk Jobs Staff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with Kewlstart hangup detection
Hello all, I;m having a (what seems to me) strange problem with some analog lines and hangup detection. The site I;m working on has 10 analog lines, my understanding is these are broken up in 2 invidiaul hunt groups (no idea why, or if this is even true). I;ve always been told that they are yyy-xxx-7200 and yyy-xxx-2115 - no idea which other lines are part of which hunt group. Anyways... I have these lines plugged into a Digium TDM800P and a TDM400P I have all the lines set to kewlstart signalling in zaptel.conf and zapata.conf, the lines are all provided by MTS Allstream (in Canada). It appears to me that the disconnect supervision on all of these lines *except* those 2 (7200 and 2115) send the disconnect notice very delayed. Calling in from an outside line to 7200 or 2115, i hangup on the remote site, and Zaptel/Ast detects the hangup almost instantly (and having debug turned on on the wctdm wctdm24xxp drivers shows the NO BATTERY / BATTERY messages almost instantly like it should). However... on the remaining 8 lines, it takes approximently 10 seconds before i see the NO BATTERY / BATTERY message. It would seem to me that for whatever reason this message is delayed on the telco side. I;m not very familiar with hunt groups on analog lines... and I can;t really see how this is possible... but do I need to do some kind of special configuration on my end to make the non-pilot numbers of the hunt group get their disconnect notice quicker? Is the telco at fault here? Is there something I can ask them specifically that will make sense to one of their lower end CSRs? Its really not a *huge* problem... except our receptionist is getting quite annoyed with several callers that hung up sometime during our IVR (her phone rings as the timeout off the IVR). Also people are getting VMs that aren;t really being left and stuff. I first thought that perhaps for some reason we only had kewlstart on the 2 lines... but like I said... I do infact see the NO BATTERY /BATTERY on all lines... just on all of them but the 2 its very delayed. Thanks, -- Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk
There is a .NET 1.1 library out there... I've played with it a little bit, but not enough that I could comment on how feature rich or stable it is... http://www.voip-info.org/wiki/view/Asterisk+.NET It'll more than likely not be compatible with AMI 1.1 however, which I believe is included in ast 1.6 -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, April 03, 2008 5:28 PM To: asterisk-users Subject: [asterisk-users] C# SIP API to Comiunicate with Asterisk Do anyone has an idea about an open source SIP API written in C# that can communicate with Asterisk, to call out? Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk
I've takena quick peak at it before... but I don;t know anybody that has actually used it... I do intend on giving it a try myself though... it comes with a very very basic sample. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Rodrigo Gonzalez [EMAIL PROTECTED] Sent: Thursday, April 03, 2008 6:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk Matt Watson escribió: There is a .NET 1.1 library out there... I've played with it a little bit, but not enough that I could comment on how feature rich or stable it is... http://www.voip-info.org/wiki/view/Asterisk+.NET It'll more than likely not be compatible with AMI 1.1 however, which I believe is included in ast 1.6 -- Matt Do you, or someone else, know where to get some example about using it? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on best operator phone (with attendant console)
We are using 57i + 560M combination as well... though we are not using the 57i ct... but the idea of giving them a cordless is a good idea. The only downside to the Aastra 57i + 560M is that it can only subscribe to 50 extensions for BLF... i haven;t run into this cap yet myself, but I have heard others talk about it... I think it was a cap introduced in one of the newer versions of firmware... not sure though, and not sure why. I'm running the latest 2.2 firmware on it... the addition of one-touch transfers in the last firmware was very nice so operator can transfer very fast, instead of having to do xfer-BLF key-xfer (for attended transfer), now they can just hit the BLF key for a blind transfer. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Sigma Networks [EMAIL PROTECTED] Sent: Saturday, April 05, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Advice on best operator phone (with attendant console) We have been marketing ipPBX systems based on asterisk for 3+ years. For the last year we've been placing Aastra 57iCT with 560M sidecars. Our attendants like the idea of a cordless handset so the attendant can go to the copy room, etc. The LCD based sidecar means you can keep it up to date without marking up paper strips. We deploy Thirdlane PBX Manager which allows us to setup the BLF (busy lamp field) via a web interface. Aastra 57iCT: http://neobits.com/aastra_-_a1758-0131-10-05_-_57i_ct_p11471.html Aastra 560m: http://neobits.com/aastra_-_a1760--10-55_-_560m_p11472.html Thirdlane PBX Manager: http://www.thirdlane.com/products/pbxmanager Feel free to contact me off list if I can be of any assistance. Regards, Jim ph: 408-701-9929 Faraz R. Khan wrote: One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets stuck 4. Voice 'gaps' (blackouts) for 4-5 seconds 5. The phone also completely locks up regularly 6. ping response goes from 8ms to 3000ms (after which the phone locks up) Wondering which operator phone would work best. I have the following choices: 1. Linksys SPA 932/962 with attendant console 2. Polycom 601/650 with attendant console I cant confirm online whether the BLF functionality will work with Asterisk 1.2.26. Is somebody using either of these phones in a high volume environment successfully? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium HPEC license counting
Not that I;m complaining But I just got my 2 HPEC license keys from digium... for TDM800P and TDM400P asterisk asterisk # zaphpec_enable Digium High-Performance Echo Canceller Enabler Copyright (C) 2006, Digium, Inc. Version 1.0.2 Use the '-l' option to see license information for software included in this program. Found key 'HPEC-KEY1' for 8 channels. Found key 'HPEC-KEY2' for 4 channels. Found valid HPEC licenses for 13 channels. Since when does 8+4 = 13 ??? maybe I should ask thinkgeek.com to make another t-shirt like this one: http://www.thinkgeek.com/tshirts/itdepartment/60f5/ ? When they first issued my TDM800P key they incorrectly set it up as a single channel license instead of 8 channel... but after going back and forth with them a couple times they got it fixed... when I had a 4+1 license it correctly showed 5 channels... is it possible that somehow my old license for KEY1 is giving me an extra license and not showing it? They didn't actually issue me a new key... just fixed it on their end and had me re-register it. After I re-registered I unloaded the zaptel, wctdm, and wctdm24xxp modules and re-loaded them all... so I;m not really sure how that original single channel license might still be lingering... but that's all I can think of. -- Matt Disclaimer Statement: This e-mail is confidential and is intended for the above-named recipient(s) only. If you are not the intended recipient and/or have received this e-mail in error, please notify us by telephone and delete this e-mail from your system without retaining a copy in any form. Any unauthorized use or disclosure of this e-mail is prohibited. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attrafax
I have a single channel license of Attrafax right now... It seems to work well from the testing I have done with it so far, which admittedly isn't as much as I was hoping to have done at this time. I;m using Linksys SPA2102 ATA's with it... basically what I;m doing is... FAX Machine - Linksys SPA2102 - SIP/T.38 - Asterisk - TDM card (currently TDM800P + TDM400P, but moving to TE220B soon) - PSTN I had some trouble with Attrafax at first, but updating the firmware on my SPA2102 fixed the problem. I've also tried interfacing a couple of Ricoh multifunction copiers with it (we have Ricoh MP 2500 MP 5000 which can both talk SIP if you have the fax option)... I haven't had any luck at all getting T.38 negotation to happen between attrafax and the Ricoh's though... I kinda decided it wasn;t worth spending the time fiddling with it when I could just attach a SPA2102 to it for 80$ Attractel will gives you a 2-week demo license of Attrafax if you request it from them, if you want, I can send you the email address of the contact I have there, just shoot me an email off list if you want his contact info. -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Wednesday, April 09, 2008 10:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Attrafax Has anyone had any luck with Attrafax? I'm looking to use it as the T.38 gateway (PRI in, T.38 out). -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switch recommendation?
I'm using Dell 3548P switches currently which I have powering 25 phone (mostly Aastra 9133i's, a couple 480i's a 57i + 560M) So... basically I have a phone on about half my ports... my power utilization on the switch is: console# show power inline Unit Power Nominal Power Consumed Power Usage Threshold Traps --- - -- --- - 1 On 370 Watts 78 Watts (21%) 95 Disable The unit will supply up to 370W of power, or 470W if you buy the additional power supply for it. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan C. Bailey Sent: Monday, April 21, 2008 9:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Switch recommendation? We've been using D-Link DES-3028P and DES-3052P switches. They can supply full power to EACH port unlike the Linksys switches we've tried. They're also rock solid from our experience. -Jon - Original Message - From: Hilary Miller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 21, 2008 8:21:12 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] Switch recommendation? On Mon, Apr 21, 2008 at 5:54 PM, Sean Dennis [EMAIL PROTECTED] wrote: The Cisco 3524 switch doesn't support 802.3af which is what your Linksys phones are going to want. Thank you for sharing Sean! When I saw them I felt a disturbance in the force, and now I know why! -- Just Hil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AST-2008-006 - 3-way handshake in IAX2 incomplete
I can;t imagine what headaches you'd have going from 1.4.11 to 1.4.19.1... that is a minor version upgrade... no real change in functionality thats basically 8 versions of bug fixes... if you just apply the IAX2 patch, you'll be fixing 1 out of probably a hundreds of bugs. Going from 1.4.x to 1.6.x however... you'd run into some headaches probably... but if you are staying in the 1.4 series you shouldn;t have any problems... worst case is if its broke you just make install your 1.4.11 overtop of 1.4.19.1 to revert back. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Brian J. Murrell [EMAIL PROTECTED] Sent: Tuesday, April 22, 2008 8:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AST-2008-006 - 3-way handshake in IAX2 incomplete On Tue, 2008-04-22 at 17:58 -0500, Security Officer wrote: Asterisk Project Security Advisory - AST-2008-006 So given that I'm new to asterisk's svn and bug tracking tool, is it sufficient then to apply the two patches (iax_dcallno_check-1.2.rev3.txt and iax_dcallno_check.rev9.txt) listed in http://bugs.digium.com/view.php?id=10078 to a 1.4.11ish release to correct this vulnerability? I really don't feel like buying into any/all of the headaches that went into 1.4.11-1.4.20. You know, if it ain't broke don't fix it, and my corollary, if it is broke, only fix what's broke, don't try to make it better. :-) Thanx, b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium B410P or Sangoma A502D?
I haven;t used any BRI cards but... call me crazy but wouldn;t they still be using Zaptel (even your sangoma... the script might just be configuring it for you)... and btw, software echo cancel happens in the zaptel kernel driver... it has nothing to do with the hardware (hence why its a software echo cancel) You also would of had the option of buying HPEC licenses for software echo cancel from digium for a rather cheap price. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Andres [EMAIL PROTECTED] Sent: Thursday, April 24, 2008 5:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium B410P or Sangoma A502D? We have tested both and they work fine. The Sangoma is much easier to install as it does not depend on any other driver, you just run 'setup-sangoma' and follow the instructions. You don't have to fiddle with the linux kernel or zaptel or chan_misdn. It just works. Plus its more modular. You can chose 2/4/6 ports to buy and if you need more just add remoras up to 24 ports. The Digium card is fixed to 4 ports, period. Having said that, make sure you stick with the version that has hardware echo cancel and not even try the other one. We made the mistake of buying the first time without echo cancel expecting to test the 'software echo cancel'. But there is no such thing as 'software echo cancel' on this card. I do not even understand why Sangoma would make a version without the hardware echo cancel. You get some degree of echo on practically every call. Andres. Patrick wrote: Hi, I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the specs of various cards I favor the Digium B410P and Sangoma A502D because of hardware echo cancellation. Does anyone have any experience with either card, good or bad? Which one would you choose and why? Thanks for your insight. Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] choopy audio when both side talk at the same time
You might want to begin with tuning your rxgain and txgain settings... there are a few methods for doing this on the internet, unfortunatly nobody can give you exactly values to use for tx/rxgain as they will be not only specific to your install, but specific to every single analog line you have... you can probably get away with setting it once for all of your lines, but i'd recomend setting it for every one. http://lists.digium.com/pipermail/asterisk-users/2004-November/064312.html That is the best guide i;ve seen to doing it, but there may be better ones. After you get rxgain and txgain tuned for every one of your lines, you'll probably notice a dramatic decrease in echo right away, but you can also tune your echocancel= and echotraining= after that. You can set these values on a per-channel basis by doing it like: rxgain=6.3 txgain=-1.0 channel = 9 rxgain=7.595 txgain=-2.0 channel = 10 rxgain=6.3 txgain=-1.136 channel = 11 etc. First step IMHO is getting your rx/txgain set properly... don't underestimate how important those values are... I learned that the hard way. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Ruben Zamora [EMAIL PROTECTED] Sent: Friday, April 25, 2008 7:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] choopy audio when both side talk at the same time Hi I have a server with the last version of asterisk branches, zaptel branches, 2 Digium Card with TDM800P 16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10 Grandstream GXP2000. zapata.conf echocancel=64 rxgain=0 txgain=0 when i place a call o receive a call, I finish a sentence i hear a ,AND when the both side talks at the same time i have choppy audio. Any help i appreciate. Thanks Ruben ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hyperthreading and multicore
This is my understanding of hyper threading, which I believe to be accurate. Basically, as some have mentioned previously, the OS 'sees' your single physical core processor as 2 logical processors, in generally, logical processors are treated exactly as if they were real processors, and in the case of many OS's. they probably don't understand the difference - Linux however does have specific SMT support for hyperthreaded cores. Basically not all CPU instructions take the same amount of clock cycles to complete, some may take 3, some may take 7, etc. Many of these clock cycles actually goto waste because the CPU is waiting for something, for example, an instruction that involves a fetch from memory, if this takes 7 clock cycles to complete, 4 of those cycles might go wasted while the CPU essentially just sits there and waits for the data to be fetched form RAM, L1 or L2, or L3 cache. Hyperthreading essentially puts these wasted CPU cycles to use by allowing the CPU to execute a separate thread while it would otherwise be idle waiting. To me Hyperthreading is an excellent technology... I;m all about efficiency and trying to maximize resource usage whenever possible... and that exactly what hyper threading does. That all being said... Hyper threading should not be thought of as effectively doubling your CPU power... as previous posters have said, Hyper threading will result in single threaded applications actually running slower.. this is because you still have other background processes running which may run on the other logical processor which could steal CPU cycles away from your main application... since you essentially have 2 threads executing on the same physical core, there are going to be times when one thread has to wait extra clock cycles while the other thread is executing. Remember its only those normally wasted clock cycles that you are going to gain a performance boost out of by making use of them... only 1 thread can actually be executing at any given time, so the CPU has to schedule these and try balance the threads equally so they each get an equal share of the physical core. I can't say how Asterisk behaves or makes use of additional cores or if hyper threading is advantageous to Asterisk or not... I don't know enough about the low level parts of Asterisk enough to make an informed opinion about that. I just thought I'd throw in my 2 cents about what hyper threading is and what it does. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis Sent: Tuesday, April 29, 2008 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hyperthreading and multicore Matt Florell wrote: Also, I have heard HT processors explained this way, on an HT processor it's like running 2 virtual processors at 70% of the specs of the processor with HT turned off. It's not really like that in all situations, but overall it has held pretty much true for me in most non-Asterisk situations. Asterisk didn't benefit much from having HT enabled on a P4 with HT capability. That wouldn't surprise me - after all, HyperThreading works on the principle of allowing two threads to use different dedicated processor resources (such as floating point math processors and so on) at the same time... however if two threads are trying to use the same processor resource, one thread will be suspended until that resource becomes available. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.10.1 Released
Does anybody know if this version fixes the soft lockup during ztcfg using a TE200B? http://bugs.digium.com/print_bug_page.php?bug_id=12468 -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Development Team [EMAIL PROTECTED] Sent: Thursday, May 01, 2008 1:07 PM Subject: [asterisk-users] Zaptel 1.4.10.1 Released The Asterisk.org development team has announced the release of Zaptel version 1.4.10.1. This release is a bug fix release for a regression in which the Zaptel udev rules were not installed correctly, as well as a few minor fixes in the xpp drivers. This release is available as a tarball as well as a patch against the previous release. It is available for download from downloads.digium.com. Thank you for your support! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.10.1 Released
err, that should of read TE220B From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Matt Watson [EMAIL PROTECTED] Sent: Thursday, May 01, 2008 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zaptel 1.4.10.1 Released Does anybody know if this version fixes the soft lockup during ztcfg using a TE200B? http://bugs.digium.com/print_bug_page.php?bug_id=12468 -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Development Team [EMAIL PROTECTED] Sent: Thursday, May 01, 2008 1:07 PM Subject: [asterisk-users] Zaptel 1.4.10.1 Released The Asterisk.org development team has announced the release of Zaptel version 1.4.10.1. This release is a bug fix release for a regression in which the Zaptel udev rules were not installed correctly, as well as a few minor fixes in the xpp drivers. This release is available as a tarball as well as a patch against the previous release. It is available for download from downloads.digium.com. Thank you for your support! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rebooting newer cisco phones
I don't know about the Cisco phones... I;m using Aastra phones which I can send a SIP NOTIFY to have them check for updated config... when they detect a new config they reboot themselves and download the new config. But your switch might also have an option to disable PoE on a per-port basis... I know our Dell 3548P's do. -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail Sent: Friday, May 02, 2008 1:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] rebooting newer cisco phones Does anyone have a solution for remotely getting the newer cisco phones (7941, 7961, 7970, etc ) to reread their configs (or even rebooting). I am running SIP firmware connected to asterisk. Check-sync doesn't seem to work anymore, I can't login to the phones as root because I am given a challenge: random digitspassword: prompt. occasionally I need to make changes to phones at remote locations, only solution I have now is rebooting the POE switch. Kinda an overkill. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI asterisk high balance
There is really no reason why you cannot. Personally... I'd avoid using Java for AGI's that you think are going to receive heavy use... simply because the JVM adds a lot of overhead, and possibly a very real performance impact from having the load the JVM everytime. Of course there is overhead as well if you do PHP instead, as the PHP interpreter has to load everytime... but that's probably pretty light-weight in comparison to the JVM. Of course you could compile your Java code to native binaries to work around that problem, but I have no experience doing that. Please keep in mind that I have not actually created or used any Java AGI's... really just my thoughts without any experience. -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chetherston miles Sent: Saturday, May 03, 2008 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AGI asterisk high balance Hello, Is there a problem to use AGI JAVA to write an AGI to billing calls and customer accounts? Anyone have experience with it could give me some tips? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] China vaults past USA in Internet users - now 220 million users in China
What Godaddy.com has told you is more or less correct. Its not their fault that Chinese visitors cannot hit your pages... the internet is China is highly censored, and quite often they firewall even very large big name sites like BBC news. Typically they block sites that have any type discuss any type of political matters that might be going on in China, or blog sites where chinese citizens might speak out. I'm not saying your site is one of these, but if they are infact doing it by IP address, its perfectly possible that your site just happens to be hosted on the same IP (or even IP block) as a site they decided to firewall - or perhaps a site used to occupy the same address space as you and they just haven't noticed its no longer there and un-firewalled it. And yes, godaddy.com cannot guarentee that if you change IP addresses that the new one will work... just like they can't guarentee that myself or any visitors to my home will be able to access your website from my internet connection... i could firewall IPs from my home just like the chinese government can firewall sites from all of their citizens. There is a chance that changing IPs will make it work, but theres also a chance the new IP will be firewalled too... Just google for Great Firewall of China -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] [EMAIL PROTECTED] Sent: Sunday, May 04, 2008 8:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] China vaults past USA in Internet users - now 220 million users in China The statistics you write of maybe true but there is a small downside to this. Presently, our website can not be seen in China and we are hosted by Godaddy.comhttp://Godaddy.com. The explanation we receive more than three times is China is blocking a number of IP addresses and there is nothing they can do! This is the kicker we work with a Federal Organization here in USA who is also hosting at Godaddy.comhttp://Godaddy.com and their website can be seen. The only difference is this ogranization is using a Class A static IP address. Godaddy.comhttp://Godaddy.com can not guarantee if changing to a static IP address this situation will change. One more thing, Godaddy.comhttp://Godaddy.com can be seen in China because their website is hosted on a separate corporate server. This is not a gripe but a realty and we are a digium select reseller. There is a consultant in China we work with and our website is translated in Mandarin. If anyone has a proposed solution for this we would greatly appreciate the dialogue. Regards, Original Message Subject: Re: [asterisk-users] China vaults past USA in Internet users - now 220 million users in China From: Dean Collins [EMAIL PROTECTED] Date: Sat, May 03, 2008 11:27 am To: [EMAIL PROTECTED]mailto:[EMAIL PROTECTED].com China has surpassed the USA as the No. 1 nation in Internet users. The number of Chinese on the Internet hit more than 220 million as of February. http://mobileanalytics.com/forum/index.php?topic=28.0http://mobileanalytics.com/forum/index.php?topic=28.0 I wonder how Americans are going to handle this little turn of events. What is really interesting in the rest of the article it discusses how the percentage of penetration for china is 17% of it's 1.3 Billion population versus 71% penetration of the USA's population of 304 million people. So with China expected to increase another 13 million users this month alone (March 2008) to 233 million users how long before there are more people using the internet in China than the entire population of the USA (I'm guessing about 7 months so about the end of this year). Does anyone in the Asterisk community have a good website for getting accurate voip minutes or some other field of reference for how successful voip penetration is in the respective countries? Would be interesting to see what countries are leading Voip implementation penetration regardless of whether it is Asterisk or Avaya etc etc. I know everyone freaked when Trixbox was collecting stats but I think it would be great for someone to write a small ‘anonymous collection module’ that an Asterisk sys-admin could download and install on their asterisk server which uploaded the stats to a community website. Even if it just collected number of new installations globally this would be a huge help to people selling asterisk to their customers who continually ask “I’ve heard about this Asterisk open source stuff but how many are there installed globally anyway?” Regards, Dean Collins mailto:[EMAIL PROTECTED][EMAIL PROTECTED]mailto:[EMAIL PROTECTED] Cognation Limited +1-212-203-4357 +61-2-9016-4652 (Sydney indial) P.S. In case you are wondering Australia has a piddling little 15m users but this is against a pop of only 20.5m so the Internet penetration is actually higher than the
Re: [asterisk-users] Asterisk in Production ?
I'm using 1.4.18 in production on 2 boxes... one of which being a custom built desktop basically, the other being a Dell 1950 III We are in a migration phase to the Dell box, right now the 1st box is doing nothing more than being a PSTN gateway to some FXO lines... basically waiting for numbers to be ported off the analog lines and onto the new T1 which is connected to the Dell box. We have the 2 boxes connected by IAX2 trunk. I had 1.4.19 and 1.4.19.1 running on the Dell box, but it started giving me a lot of trouble with the IAX2 trunk, the trunk would (seemingly) go into UNREACHABLE status and never come back without restarting asterisk (reload, or iax2 reload wouldn’t cut it). Also, occasionally people trying to make outbound calls (and this probably happened on inbound as well), would get a all circuits are busy message because of the IAX2 channel driver reporting congestion on the trunk even though it was up (and not congested) Unfortunately as this is a production box I didn’t really have time to try and debug it so I simply downgraded to .18 since it has proven itself well on the 1st box. So far since I;ve downgraded to .18 I haven’t had any problems. Both installs I have running ontop of Gentoo (wouldn’t recommend it if you are new to Linux or don’t like tweak-ability). That all being said, I'll probably give .20 a try when its released, as I see there have been some IAX2 bug fixes in it... but also by the time .20 is released I probably will have retired the box being used as a PSTN gateway and won’t need the IAX2 trunk anymore. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vinícius Fontes Sent: Tuesday, May 06, 2008 8:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk in Production ? There were some really unstable Asterisk releases in the 1.4 branch. I personally use 1.4.13 or 1.4.15 in production. Every single time I tried 1.4.16 or higher I had problems. Att Vinícius Fontes Desenvolvimento Canall Tecnologia em Comunicações Ltda. - Steve Totaro [EMAIL PROTECTED] escreveu: On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). regards I am personally a proponent of Asterisk 1.2.X as I see more and more fatal bugs in the 1.4.X code come up on the lists as well as IAX2 bugs. I constantly hear Asterisk 1.4.whatever is much better, but the bugs coming out are not just unexpected behavior that one could live with, they are segfaults, system crashes, modules not getting installed (Zaptel). I use SIP since I have seen quite a few issues with IAX2 that were solved by simply switching to SIP. The above two yield solid systems under heavy load for me. OS is not so important I do not believe. I have some running FC8 and more running CentOS, both rock solid. I think the general consensus on OS is use what you are most familiar with. While these may not be popular opinions, I still ask, what does SwitchVox use? What do some of the guys around here that setup large systems use? Is ABE even using 1.4 yet? All I see in the ABE release notes is 1.2 although I have heard that ABE should be running 1.4 Very Soon many many moons ago http://www.digium.com/en/docs/ABE/README . So either Digium doesn't trust 1.4 enough to use it for ABE or the README is out of date. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Performance issues
Google is awesome http://www.voip-info.org/wiki-Asterisk+AGI -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chetherston miles Sent: Tuesday, May 06, 2008 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Performance issues Hello, We are thinking in use asterisk-java to an billing solution, wich is the better choice, and if someone could give us a understandable description about the difference between DeadAGI and FastAGI, i found a very interesting project called asterisk2billing and they use DeadAGI, anyway wich one scale better? And there is a tool for test performance? Thanks, Roberto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI D-Channel reconfiguration = crash asterisk?
Hello, I just had to have MTS Allstream fix a new T1 install that we have that we aren't running in production yet, but it is attached to a production machine. Apparently they setup the T1 with only a 1 B-channel (how useful!) even though we had ordered it fully loaded with 23. Anyways... they just reconfigured the T1 to activate all the T1 channels and this is what I got on my * console: == Primary D-Channel on span 1 down nelson*CLI Disconnected from Asterisk server ^^ asterisk crashed. Unfortunately I didn't have * setup on this box to dump a core file, so the only additional debug info I can provide is from my asterisk log file: [May 6 11:42:23] VERBOSE[16656] logger.c: == Primary D-Channel on span 1 down [May 6 11:42:23] WARNING[16656] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [May 6 11:42:23] WARNING[16656] chan_zap.c: The PRI Call have not been destroyed Those are they only 3 relevant lines in the log file. -- Matt Disclaimer Statement: This e-mail is confidential and is intended for the above-named recipient(s) only. If you are not the intended recipient and/or have received this e-mail in error, please notify us by telephone and delete this e-mail from your system without retaining a copy in any form. Any unauthorized use or disclosure of this e-mail is prohibited. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI D-Channel reconfiguration = crash asterisk?
My bad, I also should of mentioned... That was on Asterisk 1.4.18 and Zaptel 1.4.10 Using a TE220B -- Matt From: Matt Watson Sent: Tuesday, May 06, 2008 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: PRI D-Channel reconfiguration = crash asterisk? Hello, I just had to have MTS Allstream fix a new T1 install that we have that we aren't running in production yet, but it is attached to a production machine. Apparently they setup the T1 with only a 1 B-channel (how useful!) even though we had ordered it fully loaded with 23. Anyways... they just reconfigured the T1 to activate all the T1 channels and this is what I got on my * console: == Primary D-Channel on span 1 down nelson*CLI Disconnected from Asterisk server ^^ asterisk crashed. Unfortunately I didn't have * setup on this box to dump a core file, so the only additional debug info I can provide is from my asterisk log file: [May 6 11:42:23] VERBOSE[16656] logger.c: == Primary D-Channel on span 1 down [May 6 11:42:23] WARNING[16656] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [May 6 11:42:23] WARNING[16656] chan_zap.c: The PRI Call have not been destroyed Those are they only 3 relevant lines in the log file. -- Matt Disclaimer Statement: This e-mail is confidential and is intended for the above-named recipient(s) only. If you are not the intended recipient and/or have received this e-mail in error, please notify us by telephone and delete this e-mail from your system without retaining a copy in any form. Any unauthorized use or disclosure of this e-mail is prohibited. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] This e-mail is confidential ... (was: Re: PRI D-Channel reconfiguration = crash asterisk?)
That's fine... honestly I hate the message myself, however corporate policy is corporate policy so there isn't much of a point in discussing it. That being said, the message does clearly say that the message is for the named recipients, in this particular case, the named recipient is a public mailing list. By my action of sending a message to a public mailing list, one can say there is implied consent that it gets distributed to whomever the mailing list chooses on my behalf. Thanks, -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Tuesday, May 06, 2008 12:27 PM To: Asterisk Users Subject: [asterisk-users] This e-mail is confidential ... (was: Re: PRI D-Channel reconfiguration = crash asterisk?) Matt Watson schrieb: Disclaimer Statement: This e-mail is confidential and is intended for the above-named recipient(s) only. If you are not the intended recipient and/or have received this e-mail in error, please notify us by telephone and delete this e-mail from your system without retaining a copy in any form. Any unauthorized use or disclosure of this e-mail is prohibited. Your confidential e-mail is going to end up on Google ... Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Storing voicemail on samba share
It would probably be wiser to run an IMAP server and do imap storage instead of writing to a cifs-mounted directory... or use ODBC storage... assuming they are running a database server somewhere. I don't have any experience with having * write voicemail files to CIFS/SMBFS, but I also think its not something I would try... I've personally always found that most network file systems don't tend to handle disconnects (server reboots, network outages, etc.) very well. Mind you, it might of come along way since the last time I tried. -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: Tuesday, May 06, 2008 3:34 PM To: 'Asterisk Users List' Subject: [asterisk-users] Storing voicemail on samba share A client has asked that our asterisk installation leverage their large investment in their existing data center infrastructure. We're thinking about putting the voicemail messages onto a Samba share (on their file servers). Any pros/cons to this? Does network/samba latency create choppiness? Thanks, MD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] better enumlookup handler
There is a enumlookup.agi that is included with FreePBX and thus trixbox, PBX in a flash, etc. etc. If you have trouble finding it let me know and I can send you it. I can;t really vouch for its quality, but I do use it and it does work... but i;m not sure how well it handles multiple results. I know it will successfully connect to systems that give multiple results, i;m just not sure if it does infact failover if the first one doesn;t work. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Brian J. Murrell [EMAIL PROTECTED] Sent: Tuesday, May 06, 2008 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] better enumlookup handler Does anyone have a better ENUM lookup handler than the built-in ENUMLOOKUP() function? The built-in function does not properly handle multiple return values such as: 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP !^\\+1866(.*)$!sip:[EMAIL PROTECTED] . 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP !^\\+1866(.*)$!sip:[EMAIL PROTECTED] . And thus does not handle roll-over should one be unavailable for whatever reason. There is this voip-info.org wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+and+multiple+ENUM +entries but the downloads that it's pointing to seem to be dead. Sure I could take to writing an AGI script and probably be done it in a few hours, but why re-invent the wheel? Thanx, b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting the TOS using IPtables screws up the DSCP field
Why are you trying to change the ToS from 46 (0x2e) Expedited for the RTP/RTCP packets to 16 (0x10)? I mean... these values really only need to be meaningful to yourself, your switches, your routers etc however ToS 46 (0x2e) is the standard value for RTP / RTCP as it is basically the highest priority value you can assign to something... ToS 46 is basically suppose to indicate that it trumps all other traffic and should be send before anything else (Which is a good thing for the RTP traffic) The SIP Signalling traffic is a little less important and its standard ToS value is 26 (0x1a). You also don;t need to use IPTables to set these values... Asterisk will do it for you as long as you have installed libcaps (I believe its required for it). And I don;t know what phones you are using... but your phones are probably also setting these values for you I know the Aastra phones have QoS/ToS settings under Options - Network - Type of Service -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Vikas [EMAIL PROTECTED] Sent: Wednesday, May 07, 2008 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Setting the TOS using IPtables screws up the DSCP field Concise summary: When I set the TOS to Minimize-Delay the DSCP field in the packet changes from Expedited Forwarding to Unknown Here are the details: Scenario 1: IpTables is not used to set the TOS This is what the packet looks like using wireshark: Internet Protocol, Src: 59.93.192.xx (59.93.192.xx), Dst: 64.62.134.yyy (64.62.134.yyy) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xb8 (DSCP 0x2e: Expedited Forwarding; ECN: 0x00) 1011 10.. = Differentiated Services Codepoint: Expedited Forwarding (0x2e) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Scenario 2: IpTables is used to set the TOS Output of $/etc/rc.d/init.d/iptables status Table: mangle Chain PREROUTING (policy ACCEPT) num target prot opt source destination Chain INPUT (policy ACCEPT) num target prot opt source destination Chain FORWARD (policy ACCEPT) num target prot opt source destination 1TOSudp -- 0.0.0.0/00.0.0.0/0 udp dpts:5060:5069 TOS set 0x10 2TOStcp -- 0.0.0.0/00.0.0.0/0 tcp dpts:5060:5069 TOS set 0x10 3TOSudp -- 0.0.0.0/00.0.0.0/0 udp dpts:1:2 TOS set 0x10 This is what the packet looks like using wireshark: Internet Protocol, Src: 59.93.192.xx (59.93.192.xx), Dst: 64.62.134.xxx (64.62.134.xxx) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xb0 (DSCP 0x2c: Unknown DSCP; ECN: 0x00) 1011 00.. = Differentiated Services Codepoint: Unknown (0x2c) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 I have no idea what am I doing wrong. Here is some reference reading I did: http://www.tucny.com/dscptos Any pointers in the right direction will be very much appreciated. Thanks for your time, Sysadmin http://www.debtconsolidationcare.com Internets First get out of debt community ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi network - redundancy / fault tolerance ?
I don;t have any answers for you... But I would love to hear about the results after you get this working and what road blocks you hit and how you overcame them. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Ex Vito [EMAIL PROTECTED] Sent: Wednesday, May 07, 2008 10:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] dundi network - redundancy / fault tolerance ? Hi list, I'm planning a private DUNDi network for a cross-country distributed PBX. Initially it will be composed of about 10 systems, growing to about 20. Current requirements point to a topology of two interconnected DUNDi hubs, each peering with half the PBXs... This would lead to two interconnected / inter-peered stars. Example: - Consider PBXs A to H - C and E will be hubs and peer with each other - A, B and D peer with C - F, G and H peer with E This leads to a maximum three hop lookup and will make good use of current network topology / bandwidths. Of course, should any of the hubs be unavailable and the lookup capability is severely compromised. Now, how to move on to acheive some kind of fault tolerance ? According to the docs we've studied, DUNDi does not like loops (which we assume one can limit with low enough TTLs). Our doubts are: - Should one use the order peer parameter to specify alternate lookup paths / peers ? Is that its purpose ? If not, what is it used for ? - Alternatively, should one create loops in the DUNDi topology and limit them via TTL ? - If both options are possible, which would be the trade-offs between them ? (Not clear at all to us!) - Assuming any of the above is possible as a means to acheive redundancy, which of the following topologies would your prefer ? (hmmm, maybe I need to refresh my graph theory...) ;-) #1 - Peer each PBX with both hubs #2 - Duplicate both hubs and peer each PBX with its hub and its hub dup For better understanding, take a look at: #1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png #2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png Thanks in advance for review and feedback. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi call impossible in one direction
Are you using IAX2 as your transport between the 2 servers or SIP? If you are using IAX2, are you using Asterisk 1.4.18.1 or 1.4.19.1 on either machine? If so, you may be encountering the IAX2 bug that some have been discussing on the list recently you can read it here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211000.html -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Russell Bryant [EMAIL PROTECTED] Sent: Wednesday, May 07, 2008 6:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi call impossible in one direction Andrea Spadaccini wrote: I've set up DUNDi between two asterisk boxes, and sometimes happens that calls from machine A can't reach peers in machine B, but calls from B to A work correctly. The strange thing is that the CLI command 'dundi show peers' shows correctly the registered peer in both servers, and in this situation if I make a call from B to A, suddenly peers in server A are able to call peers in machine B. Try using the DUNDi query CLI command to see what results your server is getting when you try to make calls. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Linux distribution to use in Asterisk server
I'm a Gentoo guy myself too... but the best advice I can give is just re-hashing what others have already set... pick whatever you are the most comfortable with... and if support contracts are important to you, then that will be a factor as well. I've used most of bigger distros out there over the last 10 years, but right now Gentoo is where I am at. It also might depend on if you intend on using Asterisk from the package system fo your distro or if you intend on compiling it yourself. On my * box I compile Asterisk, Zaptel, LibPRI by hand, everything else I've installed from portgage (Gentoo's package system). -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Marco [EMAIL PROTECTED] Sent: Saturday, May 10, 2008 4:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best Linux distribution to use in Asterisk server Personally, I love the debian way, but I must admit that when it gets to Asterisk, I prefer to use a RedHat-based distro like CentOS, first of all for the proven reliability, then for the widely used rpm packaging system and last because there are many distro CentOS-based that provide a stable system with FreePBX and all the stuff :-P Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?
FreePBX has this functionality... they call it Confirm Calls I;m not sure if you can set it on actual extensions, but I know you can set it on ring groups. I don't imagine the dialplan for doing it is very complicated if you wanted to do it by hand. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Robert DeVries [EMAIL PROTECTED] Sent: Sunday, May 11, 2008 12:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect? GrandCentral has a feature where when you call the GrandCentral number it can ring multiple phones. However, it's not the first phone to answer that gets connected, but the first phone to answer AND play a touch-tone after hearing a recording. The advantage of this is that if one of the called phones has voicemail, it won't get connected to the calling party because the VM won't send a touch tone in response to the recording, unlike a live person. I have always resisted implementing a multiple ring scenario with Asterisk that included a cellphone because of the voicemail answering problem, but this seems to be a solution. Anyone know how to implement it with Asterisk? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?
I just took a quick look at the dialplan that freepbx uses for doing call confirmation... the dialplan part of it is actually quite simple... its just a matter of setting the USE_CONFIRMATION varialbe =TRUE. However, the actual magic looks like it happenes through its dialparties.agi... which is a little more complicated than i'd like to try and dissect on a sunday afternoon! but that might be a good place to look at how its done to learn by example. I know in the freepbx implementation what it does is whenever a handset thats part of the ringgroup answers, they get a recorded message You have an incoming call, press 1 to accept maybe it says something else too... can;t recall at the moment. The first member of the Ring group to hit 1 gets the call... if more than 1 person picks up the handset right away, the first to hit 1 gets it, and the rest hear a sorry, too late, somebody else got it-type message (no idea what it actually says). From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Steve Totaro [EMAIL PROTECTED] Sent: Sunday, May 11, 2008 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect? On Sun, May 11, 2008 at 12:24 PM, Robert DeVries [EMAIL PROTECTED] wrote: GrandCentral has a feature where when you call the GrandCentral number it can ring multiple phones. However, it's not the first phone to answer that gets connected, but the first phone to answer AND play a touch-tone after hearing a recording. The advantage of this is that if one of the called phones has voicemail, it won't get connected to the calling party because the VM won't send a touch tone in response to the recording, unlike a live person. I have always resisted implementing a multiple ring scenario with Asterisk that included a cellphone because of the voicemail answering problem, but this seems to be a solution. Anyone know how to implement it with Asterisk? GREAT IDEA! (even if it wasn't yours ;-) I have had so many issues with this and desk phones, cell phones being out of range, turned off, or answering machines set to answer after two rings. If this gets implemented, it would be a great feature and save me tons of complaints and explanations. Maybe a posting on the dev list is appropriate. I would certainly contribute to a bounty. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium AEX410
Poking around the zaptel SVN earlier today i see support was added for an AEX410 card recently... I'm going to go out on a limb and assume this is the PCI-Express version of the TDM410? Any hints on a general availability date? -- Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3U server chassis Digium TE405P?
I'm not sure if a full-height card would fit (vertically) in a 3U chassis... but I would probably also assume that if it would not, that the chassis/mobo would have a PCI/PCI-Express riser card that would mount the cards horizontally. Might want to check that out with the manufacturer of the chassis. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Monday, May 12, 2008 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 3U server chassis Digium TE405P? Gentlemen, First let me say it's great to be back on the Asterisk mailing lists. Those of you who have been around for a while will remember me as Rushowr. I look forward to answering questions and whatnot in the future, but for the moment I have a minor question that I cannot find a definitive answer for online. I am in possession of a Digium TE405P card which I _know_ will fit in a 4U chassis, but we are building a new server and cannot get a 4U from the supplier that my current client wants to use. However, we can get a 3U chassis. My question is, will this card fit? Does anyone out there have a 405 out there that they have installed in a 3U? Thanks in advance for any help that can be offered, Sherwood McGowan VoIP / Telecom Solutions Consultant ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call only for registered sip users...
Do you mean What do I need to configure on my * installation so that only registered sip users can make calls? ? If so, you are going to need to give a lot more details regarding your current configuration for you to get any answers. -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Tuesday, May 13, 2008 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call only for registered sip users... What I need to configure in my * to permit make calls only registered sip users?? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF Compatible Phones
I'm using Aastra 57i + 560M sidecars for receptionists... the only downside is that they support a max of 50 BLF subscriptions... you can setup up to 180 blf keys with 3 560Ms but it will still only subscribe to a max of 50... from what I understand it's a firmware limitation. For 4-6 phones you could probably get away with doing it directly on the 57i with no 560M's (or 536M's) too many more phones and you'd need the sidecars just for the extra buttons I think. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, May 13, 2008 3:50 PM To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.com Subject: [asterisk-users] BLF Compatible Phones I am new to asterisk and am looking to setup a small office with 4-6 IP phones and 4 analog lines from the local telco (primary line with HUNT to the other lines). I am considering purchase of a Digium AEX800. One of the features that will be important (particularly for the receptionist desk is to show status of the other lines in use). I don't want the receptionist to pick up a line if it being used. Is there a list of phones that are BLF (Busy Lamp Field) compatible? I'm assuming (after reading tons of misc articles) that this is what I need in order for the receptionist not to pick up lines in use. If this is not the case please set me straight. I am considering the cisco 7960's, linksys SPA942, and possibly some polycom phones. I was leaning toward the 7960 but I've read that it is not BLF compatible. Are there any workarounds for this? I am new to the game and would be grateful for any recommendations on which phones would be the easiest to setup, etc. I currently have a working asterisk install at home with a single cisco 7960 registered which isn't hooked up to any trunks as of yet. Thanks, Dayton Gray ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk for Larg
You'd probably want to run something else to handle your registrations like OpenSER with that many phones. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bhrugu Mehta Sent: Thursday, May 15, 2008 8:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk for Larg hi, I have not tested that but I have seen 100 agents configure with asterisk. thnks Bhrugu mehta On 5/15/08, gmail [EMAIL PROTECTED] wrote: Is Asterisk practically stable and reliable for a larg Enterprise has say a 1 phones , is there any case study like this? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *72 Telco Call Forwarding
Is there any reason you don't want to use Wait()? However, I would use WaitForRing() myself - its also a great solution on dirty analog lines where you receive phantom calls. That being said, I don't know how to do it without using some form of Wait.. as far as I know zapata.conf doesn't provide a method of telling Asterisk to wait for a specific period of time or rings -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Thursday, May 15, 2008 12:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] *72 Telco Call Forwarding Is there a way to force asterisk to ignore the first ring of a call without using Wait() ? When I active *72 call forward on my analog lines from the telco, they always send a single ring and then do the forwarding. Asterisk picks up essentially a dead line and rings the phones which gets really annoying. Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk concurrent calls count
You can NOT use bogomips as any kind of measurement for system performance. First of all, Bogomips is a linux-specific thing and not available on other OS that Asterisk runs on. The second, and far more important point. Bogo is taken from the word Bogus. Bogomips are not a measurement of system performance, it is simply a number used for calibrating parts of the kernel for your CPU. The problem with coming up with these numbers of concurrent calls is that Asterisk is not a complete package. Meaning, it's the software portion only, most other systems when you get them are going to be the software the hardware in one package, the 2 go hand in hand and are specifically designed for each other. Asterisk does not fall into that category unless you invest in one of the many asterisk appliances out there. Digium has no control over what hardware you are going to run Asterisk on, so they can't provide you with these numbers. Heres a few questions at the top of my head that I think would influence the answer: are you recording calls? are you transcoding calls? are you using T1s or SIP/IAX trunks? Did you buy the 7.2krpm, 10krpm, or 15krpm hard-drives? Do your harddrives have 8mb, 16mb, or 32mb cache? Did you buy the better SAS controller? Did you buy 667mhz or 800mhz ram? Are you using EXT3, ReisferFS, XFS, JFS, ZFS, UFS? Are you using AGIs? Are you using MeetMEs? How many? Whats the average length of the conferences? Are devices using re-invites to take Asterisk out of the call loop? The list goes on and on... and every single one of those answers is going to influence that number for How many calls can my system handle? -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Backeberg Sent: Friday, May 16, 2008 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk concurrent calls count I wonder if there's a proportion where somebody could take some standard kernel output, say bogomips, and guesstimate some proportionality from that. As in: bogomips says this, expect ballpark 120 SIP over codec calls. It certainly seems like there could be some kind of asterisk benchmarking utility kindof like Sandra for Windows. I know there are a gazillion variables in asterisk, and that's why asterisk is so powerful. But some benchmarking utility would at least allow some (even if phony baloney) relative comparisons between similar hardware. Has anybody ever tried to roll their own VoIP or Zaptel load simulator? How did they do it? On Fri, May 16, 2008 at 7:59 AM, Al Baker [EMAIL PROTECTED] wrote: this is one very weak area for *. There is NO ANSWER. Now in fairness to *, the answer DOES depend on a # of critical variables. How much CODEC to CODEC transcription is going on. How many MEET Me conferences are going on. On the other hand, DIGIUM COULD, since they have a lab take 4-5 'standard' workloads on two of the most common hardware boxes, say Dell HP, and run x # of transcriptions and show the #'s. Then x # of meet-me conferences. Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks Rockwell and NORTEL can tell you this for every piece of hardware they sell. It is a an area DIGIUM need to man-up in. Alexey Shimeshov wrote: Hello, Alexander. AO Hi Asterisk Users, AO I'm interested in how many concurrent calls Asterisk can process without AO troubles. I mean 1 Asterisk server (software) like either proxy or media AO server (any numbers will be appropriate). AO 1. Is there any limitations by the software? What is this number? AO 2. What is the maximum count of concurrent calls you've ever seen/tested? Look at this example http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Card: Power Connector, from SATA to NORMAL
On May 17, 2008 06:59:43 am Gordon Henderson wrote: On Sat, 17 May 2008, bilal ghayyad wrote: Well, why Digium is still using this kind of power connector while all new machines does not come with these types? The new machines that I buy come with legacy power connectors. The flash IDE drives I buy need legacy power connectors, and since convertors are trivially avalable why is it an issue? molex power connectors are not legacy or old style except when used in reference to SATA devices. Seeing as how Digium interface cards are not SATA devices, why would you expect them to be using a SATA connector? Its not Digiums fault that the PSU you bought doesn't include molex connectors. SATA uses a different power connector for a few reasons, but the biggest is that SATA supports hot-plugging (assuming your controller, drive, and OS support it), in order for hot-plug to work the drive needs a 3.3V voltage as well as 5V and 12V, molex only gives 5V and 12V. The actual physical connector that molex uses also does not lend itself very well to hot-plugging. SATA power connectors, while there is no real reason that non-SATA devices couldn't use them, they simply were designed for SATA specifically. Personally, I prefer molex connectors for most things simply because they are far more secure than SATA connectors (at least the ones i've used). -- Matt http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF Compatible Phones
On May 19, 2008 12:51:09 pm Grygoriy Dobrovolskyy wrote: Hmm, i dont like aastra really much, their transfer management is not human friendly ;) What do you mean by that? I've run my Aastra's with BLF using both Aastra's 'blf' mode and 'blfxfer' mode... the former is basically attended transfer, and the later is basically blind transfer... both of which our staff haven;t had any problems with. using 'blf' its basically 1. press xfer 2. press blf key 3. ask if they want to take the call 4. press xfer again using 'blfxfer' its: 1. press blf key In both modes the blf keys also act as speed-dial keys. I don't really see how either of those is not 'human friendly'? Or maybe I'm just completely off on what you mean. -- Matt http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk first time user
On May 19, 2008 03:21:34 pm Aaron Stranberg wrote: Folks, We are a small office with remote users less than 20 total phone extensions, and I am looking for some guidance on choosing between asterisknow and a centos/ubuntu or any other os with an asterisk + asteriskgui build out? Looking to get up and going quick with some method of GUI administration that won't require a ton of ongoing linux admin level support. I hit a couple of stumbles going the asterisk + asterisk GUI route (404 errors on ivr page etc..) and am tempted to take the easy path of asterisknow iso and go. Thanks for any pointers, and advance apologies if this had been beat to death. -Aaron IMHO, there is really no way to say this one is best. Each solution might be better at X while the other is better at Y... its very dependent on your situation Though, I gather you'd rather not deal with the actual OS-level, so you are probably best to stick with one of the complete packages like AsteriskNOW, Trixbox (they have a free and paid version), PBX in a Flash, and i;m sure there are many others... I haven't used any of them however so I can't really speak about the pros and cons of them. -- Matt http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fedora 9 + Asterisk
On May 19, 2008 06:49:23 pm Kevin Smith wrote: I almost hate to admit this...but I'm still running Asterisk 1.2 on Fedora 4 :D IMO theres nothing wrong with running an old version of * or an old version of the OS... as long as the box doesn't have a public IP bound to it. If its an internal-only box and its running bug-free, why screw with it? If you can get it to over the internet however... security becomes a big concern. -- Matt http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Machine Options
The first part of this is kind of off topic as it doesn't answer OP's original question, but instead is a reply to one of the replies. Cisco is certainly not the only option for doing T38 gatewaying with Asterisk. I believe Asterisk 1.6 with app_fax supports T.38 origination and termination, that is not gatewaying, however if origination and termination are already there, gatewaying should be fairly trivial to implement. I haven't actually tested 1.6 using T.38, however I have read: http://www.asterisk.org/node/48457 11873, Added core API changes to handle T.38 origination and termination (The version of app_fax in Asterisk-addons now supports this.) Additionally, there are some 3rd party modules available for Asterisk 1.4 that will add T.38 termination, origination, and gatewaying. The ones I am thinking of specifically are the ones made by Attractel in there Attrafax package (previously known as Faxterisk): http://www.attrafax.com/attrafax.php I have used Attrafax before and it works great for us. We use it in combination with Linksys SPA2102 ATAs. We had problems with it at first but upgrading the firmware on the Linksys ATAs made the problem go away. In our case we have a PRI however and are not using SIP connections over the internet. Another option as you have already stated is using a SIP provider that supports T.38 such as gafachi. However in this particular case I understand the OP has already provisioned DIDs from a SIP provider, assuming one of these DIDs is your fax number you may find yourself with a bit of a problem if your provider does not support T.38. You may have some luck with faxing w/o T.38 using G.711a/u over the internet, but it will be patchy at best, you will probably find you will have many failed faxes doing this. Using G.711a/u internally over a LAN is one thing (still wouldn't recommend it, but you would get a high success rate), but doing it over the internet is a completely different story. If you have no real PSTN connections and are SIP only, your provider *must* support T.38 to achieve an acceptable success rate. If your DIDs are already on print materials and your provider doesn't support T.38, the only options I would see for you are: 1) Have your Fax DID ported to another SIP provider that does support T.38, you can leave your voice ones with your current provider 2) Get a new DID from a another SIP provider and re-print all of your materials (probably incredibly expensive) -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Monday, May 19, 2008 11:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Machine Options Cisco gateway with T.38 support. That's the only real way to do faxing through asterisk. I think a VG200 with newer firmware will support SIP + T.38 but don't buy on my suggestion because I've never used that device outside call manager configuration. Or see if your VoIP provider supports T.38 fax but you must use SIP in that case. It will work very well once you get it working hint: check udptl.conf On Mon, May 19, 2008 at 11:27 PM, Joseph L. Casale [EMAIL PROTECTED] wrote: Is my only solution to add a fax machine to our VOIP only setup by using an IAXy? I should specify the office people want a traditional fax machine in the sense that fax's be sent and received from a physical unit, they don't want an email to fax setup. They have a dedicated sip did provisioned just for the fax. What are others using? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Machine Options
Interesting... I really don;t know the T.38 protocol other than what it does. How it goes about doing it I haven;t really gotten into. I would of thought that gatewaying would of (essentially) be a bridge between a termination and origination action. However that is just completely sort of what i think without any real evidence behind my thoughts. One day i'll have to read up on it. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Steve Underwood [EMAIL PROTECTED] Sent: Tuesday, May 20, 2008 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Machine Options Matt Watson wrote: I believe Asterisk 1.6 with app_fax supports T.38 origination and termination, that is not gatewaying, however if origination and termination are already there, gatewaying should be fairly trivial to implement. I haven't actually tested 1.6 using T.38, however I have read: http://www.asterisk.org/node/48457 T.38 gateway is a totally different problem than T.38 origination/termination. They share very little code, and almost none of their design. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server recommendation help
You might want to see if you can change the IRQ assignments in your servers bios (might have to turn off the PNP OS Installed option if you have one) -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Cavanna, Richard [EMAIL PROTECTED] Sent: Tuesday, May 20, 2008 11:04 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Server recommendation help On Tue, May 20, 2008 at 07:55:32AM -0500, Cavanna, Richard wrote: I am having a issues with adding a analog card to my dell 2800. I already have a t1 card installed and running fine but when I install the analog card asterisk will not start (ztcfg fails). I have determined it is because of a IRQ problem and have decided to get a new server. Can anyone suggest a server grade setup that supports this? I would rather not buy a machine that will be unstable. I am not opposed to building one if need be. You omit some obvious details, such as the actual error message you get, your /etc/zaptel.conf and the output of 'cat /proc/zaptel/*' . I figure that the driver for the analog card loads before the driver for the T1 card, hence invalidating your configuration. Cards I have installed: DigiumTE205P - 5v TDM410 Thanks for the offer to help trouble shoot but I never even got that far. On boot it actually makes my Ethernet ports fail. I am thinking of disabling the internal ports and adding in a card. Might help but as this box is in production I do not have the ability to do that much testing on it. I am looking at some planned downtime next weekend to go further with this. Also this box is ~3 years old so I figure If I can find a box that is not too expensive I can migrate this to a FTP box and get a new * server. As added bonus I can use the reload to clean up my dial plan. If anyone has a TE205P-5v and TDM410 running is same box I would like to know the setup. Richard -- This message was scanned by ESVA and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Does your extensions.conf have any more configuration than what you've shown?
If not, then you are lacking dialplan for anything but internal calls.
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd
Sent: Wednesday, May 21, 2008 9:01 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn
calls)
Hello all,
its been a while im trying to setup my asterisk/sipura 3102 to recieve/make
calls from softphones on pcs in my home..
i've set up 5 SIP extensions in sip.conf and made the dialing plan in
extensions.conf..
i could make calls from 1 sip phone to another in my home.. but i cant call out
using pstn line interface nor recieve calls..
please find below my topology as well as config info:
(192.168.0.0)
LAN__
|| |
softphone asterisk sipura-PSTN LINE
Configuration:
ASTERISK:
sip.conf
[101]
type=peer
port=5062
host=dynamic
secret=1234
context=spa
[103]
type=peer
port=5061
host=dynamic
secret=1234
context=spa
[100]
type=peer
port=5061
host=dynamic
secret=1234
context=spa
[111]
type=peer
port=5060
host=dynamic
secret=1234
context=spa
== ===
EXTENSIONS.CONF
[spa]
Exten = _1XX,1,Dial(SIP/${EXTEN})
== ===
and this is the settings i have right now for sipura 3102 in my PSTN LINE:
http://img84.imageshack.us/my.php?image=40541922um2.jpghttp://www.voipuser.org/ship_to.php?url=http://img84.imageshack.us/my.php?image=40541922um2.jpg
http://img98.imageshack.us/my.php?image=55448347ss9.jpghttp://www.voipuser.org/ship_to.php?url=http://img98.imageshack.us/my.php?image=55448347ss9.jpg
http://img262.imageshack.us/my.php?imag ...
472qz3.jpghttp://img262.imageshack.us/my.php?imag%20...%20472qz3.jpg
ps: i read so many tutorials and none seems to help..
lately whenever i try to call out using my sipphone.. it gives me 503 service
unavailable and this is wht shows on the CLI of asterisk when i set sip debug
on..
ubuntu-pbx-desktop*CLI
== Connect attempt from '127.0.0.1' unable to authenticate
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/1003-b5f05600, SIP/1009) in
new stack
-- Called 1009*CLI
-- Got SIP response 410 Gone back from 192.168.0.111
-- SIP/1009-081741d0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/1003-b5f05600' status is 'CONGESTION'
Invite your mail contacts to join your friends list with Windows Live Spaces.
It's easy! Try
it!http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us
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Re: [asterisk-users] Asterisk Wackyness
On May 22, 2008 02:06:06 pm Jared Smith wrote: On Thu, 2008-05-22 at 10:48 -0700, Douglas Garstang wrote: We didn't want to be generating actual network traffic for this, so I tried originating a call to [EMAIL PROTECTED] Why not try [EMAIL PROTECTED] and see if that solves the problem for you? I'm going to make a wild guess here that Asterisk is trying to do a DNS lookup on whatever you have there for xxx. Is your xxx numeric? I'll bet if you change it to [EMAIL PROTECTED] you won't see the problem. specifying localhost is probably a good idea... if you just specify a random string say asdf more than likely whats going to happen is asterisk will try and do a DNS query via gethostbyname() (guessing thats what * uses). Whats then going to happen is the resolver is first going to try and do a DNS query on asdf thats going to fail, what will happen next is it will try and search for asdf.your search domain specified in /etc/resolv.conf. Depending on what your asdf string is... that domain might actually resolv to something and gethostbyname() is going to return the results. That all being said... I'm surprised you are not more concerned with fixing the real problem instead of your workaround... By any chance is your SQL server not on the same subnet as your * box? If not, do you have something like a cisco router/PIX/ASA between the subnets? if thats the case, your router might be detecting the idle connection and killing it. I know our PIX will do this. I suppose even if its on the same subnet, you could have something running on either your * box or mysql box that will blow away idle connections... but that would probably be a little more obvious and you'd know about it. -- Matt http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handling multiple fax machines and the fax extension, and general call routing
On May 22, 2008 04:42:27 pm Steve Totaro wrote: PS. Figured I would start with DHADI now. psst. its DAHDI ;) -- Matt http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] forwarding pots lines
On May 23, 2008 11:25:55 am Dennis P. Clark wrote: Will fax and dial-up internet work through the gateway? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Friday, May 23, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List -Non-Commercial Discussion Subject: Re: [asterisk-users] forwarding pots lines There are a couple of companies out there that make 24 port fxo and fxs boxes. If you have some unused fibers you cout do this very reliably with two channel banks... One with fxs ports and the other with fxo ports and t1 media converters. The grand stream solution mentioned in an earlier post does 8 ports, you could get one 4 port model and one 8 port model of fxs and the same of fxo and accomplish your goal rather inexpensively as well. In generaly this is a bad idea (especially dialup internet). If both the gateways you use support T.38 origination/termination then faxing will not be a problem at all. However, in your case I assume you are only transporting the calls over LAN, and there is no WAN/Internet involved... which means you will probably achive a high success rate for both dialup and fax... I wouldn;t be surprised if you can;t max out the baud on your dialup internet connections though... i'd expect a slight reduction in speed (and errors, though error correction built into your modem would hopefully take care of this, at the cost of a a little speed due to re-transmissions) -- Matt http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *#%! Polycom...
On May 23, 2008 05:27:49 pm Ken D'Ambrosio wrote: I used to do lots of Asterisk, but got an offer I couldn't refuse, and went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want to set up a test system. One thing I'd really like to get my hands on is recent firmware, etc., for SoundPoint IP 430's. Freedomphones.net, my old source, seems to have been kaput about as long as I've been a sysadmin; are there any other sources out there? (And, yeah, if anyone wants to e-mail them to me directly, I won't say no.) My source was google, and I came across this almost right away: http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html presumably their RPM includes firmware for all of the polycom's I don't use polycom's and never actually downloaded the RPM... but it seems to me thats what you are looking for. You should also be able to contact whomever you bought your Polycom's from to obtain the most recent versions. -- Matt http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Have you tuned rxgain txgain in Zapata.conf? shameless-self-plug http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/ /plug Also, have you used fxotune to tune each FXO interface? I believe echo cancellation happens at the Zaptel / DAHDI level, so using Asterisk 1.6 probably isn't going to give you any benefit. -- Matt Watson http://www.mattgwatson.ca -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: Wednesday, June 04, 2008 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6 vs 1.4? Is there some location that outlines the major differences between Asterisk version 1.4 and version 1.6? I've read through change logs and several technical discussions, but technical details don't really give me the big picture. Basically, is 1.6 more stable than 1.4? Is it more efficient? Does it work better with echo cancelers like Oslec? I'm currently using Asterisk as a PBX for our branch offices and will soon be converting our main office. Our goal is to be able to have 2 analog lines at each office, calls come in to each PBX and are routed by VOIP to a receptionist at one of the offices who then routes calls appropriately. We're currently using Asterisk 1.4.19, Zaptel 1.4.10, Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones. All of our branch offices have 1MBPS DSL connections and are linked to each other by VPN's running on our Cisco 1720 routers. Our only problem so far is with intermittent echo on calls. Most of the calls have a little echo right at first, but it goes away almost immediately as the echo canceler trains. Every now and then, however, we get a call with terrible echo. I've put in several e-mails to rhino support asking if the hardware echo canceler needs something I haven't done but didn't get a response. I know echo is just something we have deal with when using analog lines, but I didn't think it would be this big of a problem. All of our offices are in rural areas where digital lines are unavailable so that is not an option. Given this setup, is there any reason for me to switch to Asterisk 1.6 or should I stick with 1.4? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune vs rxgain/txgain
In short, fxotune adjusts line impedance, where as adjusting gains I believe is essentially adjusting the amplification / deamplification of the signal. http://www.voip-info.org/wiki/view/Asterisk+fxotune -- Matt Watson http://www.mattgwatson.ca On June 6, 2008 12:43:51 am Noah Miller wrote: Hi All - I hope somebody can clarify for me what exactly fxotune does, and how it is related to gain settings. I've been reading what appears to be conflicting information from various sources. I've got a box with an AEX800 with 6 lines (from Qwest) running asterisk and zaptel versions 1.4.20.1 and 1.4.11 respectively. We've been experiencing some echo/quality issues on certain calls which seem to happen on all 6 of the lines. I manually calibrated the rxgain/txgain using ztmonitor and a milliwatt test line to the somewhat improbable levels of +10.0/-2.0 (about the same for all 6 lines). These settings yield acceptable call volumes, but echo and noise are problems. If I run fxotune, it gives me the following numbers: 1=10,0,0,0,0,0,0,0,0 2=12,0,0,0,0,0,0,0,0 3=12,0,0,0,0,0,0,0,0 4=10,0,0,0,0,0,0,0,0 5=10,0,0,0,0,0,0,0,0 6=10,0,0,0,0,0,0,0,0 Two questions here: 1) What do these numbers mean? Are they in any way related to either rxgain or txgain? 2) Am I supposed to set rxgain and txgain back to 0 if I use fxotune -s? If I do use these fxotune settings and set rxgain and txgain to zero, the volume on incoming zap calls is almost too low to be heard, but echo issues seem to be solved. Do I have to choose between 1) acceptable call volume with echo or 2) super-quiet call volume without echo? Should I petition Qwest to install a repeater? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax on FXS
On June 7, 2008 11:37:20 am bilal ghayyad wrote: Hi List; What configuration needed to let my FXS send and receive FAX? Your probably going to need to give some more details about your setup before anybody can help you... theres really nothing special you need to configure for an FXS port to attach a fax machine to it... keep in mind that faxing over VoIP is extremely tricky at best, but if your entire call path is TDM then you shouldn;t have much of a problem. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax on FXS
On June 9, 2008 12:57:11 pm John Morey wrote: I've been thinking about something around these lines that I'd like feedback on. What I'd like to d,o if it works, is have a fax machine in St. Louis connected up to my asterisk box in Atlanta via Internet/SIP so that anytime the fax machine in St Louis sends a fax it actually goes out through the asterisk box in Atlanta. Something if I understand it correctly like : Fax-SIP(long distance)-Asterisk-FXO-Customer Fax. Would something like this work? This will not work: Fax - SIP ATA - [internet] - Asterisk - FXO - Fax Because you don;t have end-to-end T.38 support, Asterisk supports T.38 pass-thru but not origination/termination (yet). However, what *should* work is: Fax - SIP ATA - [internet] - Asterisk - SIP ATA - Fax the SIP ATAs obviously have to have T.38 support - for example a Linksys SPA2102 should do it for you. I've never tried faxing between ATA's so I don;t know if they can actually negotiate T.38 support between each other, but I don't really see a reason why they couldn't. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax on FXS
On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote: You should not expect FaxOverVoiceOverIPOverInternet to work well. If you stick to ulaw codec for the entire call, it might work well enough for your use, but it might not. Just as an FYI - you have too many Over's in your description FaxOverVoiceOverIP would make sense, but seeing as how IP is short for Internet Protocol, saying Internet Protocol Over Internet doesn;t make much sense... I would interpret that description as somebody trying to send a fax through a VoIP system using a voice-codec like G.711 However, there is the T.38 protocol which is designed to solve this exact problem, Asterisk support for it is just rather limited currently (pass-thru only). T.38 often gets referred to as FoIP (Fax over Ineternet Protocol) -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interoffice phone setup
On June 9, 2008 07:49:13 pm Joseph L. Casale wrote: What type of PBX hardware do you have on-site? Also what make/models of phones? Michael/Darryl, I do have a local asterisk box, which is why I am baffled. I am new to Asterisk and there is lots to learn, but my config is pretty basic, my sip.conf simply has the phones and single sip provider context in it. It doesn't make sense that the voip provider going offline takes the whole setup out with it. I am suspecting something else went south at the same time. I have snom m3's and one Astra 480i. Thanks! jlc I've seen this behaviour from Asterisk as well... while I can't say I have tracked it down and verified this... I've seen other talks about how Asterisk gets rather unhappy when it can't preform DNS queries. I suspect that may be your problem. Might want to check the archives for other issues that people have talked about DNS as a possible cause and see if there are any similarities. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax on FXS
Ah, you got me there! Could start throwing in a lot of Over's going down that road :) -- Matt http://www.mattgwatson.ca -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, June 10, 2008 4:10 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fax on FXS On Mon, Jun 09, 2008 at 06:53:34PM -0400, Matt Watson wrote: On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote: You should not expect FaxOverVoiceOverIPOverInternet to work well. If you stick to ulaw codec for the entire call, it might work well enough for your use, but it might not. Just as an FYI - you have too many Over's in your description FaxOverVoiceOverIP would make sense, but seeing as how IP is short for Internet Protocol, saying Internet Protocol Over Internet doesn;t make much sense... Unless you use an openvpn / ipsec tunnel :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems configuring a PRI...
1. Ditch the channels= in zaptel.conf that doesnt belong there (you've done the channel config with the bchan= and dchan= 2. your span= should *probably* be 1,1 instead of 1,0 in zaptel.conf the 2nd 1 indicates to use that span as a primary timing source 3. not that it should matter, but you don;t need the duplicate group=, signalling=, switchtype= in zapata.conf 4. you can ditch rxwink= that setting is for non-PRI T1s try that and see if that helps... I suspect the span not being used as primary timing source is whats causing your greif. good luck! -- Matt Watson http://www.mattgwatson.ca On June 10, 2008 05:22:40 pm Christopher Hoff wrote: I'm trying to get a Qwest PRI configured and working with my lab Asterisk server. They said that the switchtype is 5ess and the signaling is pri_cpe. My entries into zaptel.conf are: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us channels=1-23 And my entries in zapata.conf are: language=en context=telco-incoming switchtype=5ess signalling=pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group = 1 switchtype = 5ess signalling = pri_cpe group = 1 channel = 1-23 I'm not able to make/receive calls, and the error I'm receiving is: [Jun 10 11:32:37] WARNING[31768]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 down Qwest says that the PRI is fine. I have a green light on the PRI card. Help! ___ Chris Hoff Telecommunications Administrator SEI LLC Voice +1 701 298 8865 Ext 2189 Mobile +1 701 361 5976 Fax +1 701 298 8860 Email [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice on Best High traffi c fxo gateway/cards
On June 15, 2008 10:53:35 am Brian J. Murrell wrote: On Sun, 2008-06-15 at 17:43 +0300, James Mutuku wrote: Please advice on channel bank Dude. There's the cool new website you should check out. It's www.google.com. Seriously. This list is not full of people waiting to do the simplest research at your request. Spend a few minutes and do some self-help before coming here asking the simplest, most general questions. You are more likely to get answers to interesting questions rather than mundane-google-would-have-told-you-all-you-need-to-know-in-5-minutes questions. b/ Heh, there seems to have been a few of these kinds of requests on the list lately eh... I think I saw one a couple days ago where somebody was asking about a dozen very basic questions that they were tasked with to find out of Asterisk was suitable for their organization... and it was seemingly like they were literally trying to offload their job to this list -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *OT* DLI Ethernet Power Controlle r $289 (I paid $200 for a two port websw itch)
On June 15, 2008 12:11:13 pm Mark Hamilton wrote: This sounds good. Except I'm a little confused. Is this a reboot bar which uses Ethernet to do the reboots? Like a reboot bar, except in a PoE lifestyle? Its just a PDU (power distribution unit) that has a web-interface (accessed via Ethernet)... it looks like it has the added bonus of having some RS232C ports that you can either attach a modem (to dial-into the device) or to connect to serial-console based equipment, like certain routers and switches so that you can access their serial console remotely. Essentially its the equivalent of the APC AP7902 - http://www.apc.com/resource/include/techspec_index.cfm?base_sku=AP7902 it has a couple feature differences, but for the most part they do the same thing. However, the cost is significantly less than than the APC model. I don't have any experience with either however. All in all it looks like a decent product... i'd be interested in hearing from anybody that might of been using them for a long period of time (1-2yrs+). I'm pretty picky about power distribution, i've seen bad power cause too many problems in my computing history. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *OT* DLI Ethernet Power Controller $289 (I paid$200 for a two port webswitch)
On June 15, 2008 12:10:03 pm Dean Collins wrote: Do you know if this unit has any power metering capability? I'd really like to start measuring which of my servers are using the most power etc and not sure from this description if this is possible. Just FYI - the APC model I mentioned in my last post will do load metering... it'll cost you about twice as much as the one Steve posted however. The sale price that is, couple hundred more than the regular price. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *OT* DLI Ethernet Power Controller $289 (I paid $200 for a two port webswitch)
On June 15, 2008 12:45:07 pm Steve Totaro wrote: If a PDU is just a power strip then this has many more bells and whistles. My usage is being able to control power to those outlets to reboot or turn on/off equipment if it is hung or whatever other reason. Thanks, Steve T Yep, a PDU is really just a broad term for anything that takes power input and distributes it to many devices - so yes a power strip/bar would be classified as a PDU...technically. However I'd probably expect to get laughed at if I called a power bar a PDU! There are pretty big differences between the $5 power bar you can buy at walmart and stuff that people typically put into server cabinets. technically this device specifically would be a Switched PDU the switched part meaning it has the ability to turn on/off individual ports. The interface to turning those ports on/off however is irrelavent to its device classification it could be a web interface like this device, a telnet/ssh interface, it might not even have any remote capability and just have physically switches for each port. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *OT* DLI Ethernet Power Controller $289 (I paid $200 for a two port webswitch)
On June 15, 2008 01:05:40 pm Andres wrote: All in all it looks like a decent product... i'd be interested in hearing from anybody that might of been using them for a long period of time (1-2yrs+). I'm pretty picky about power distribution, i've seen bad power cause too many problems in my computing history. We have used the APC 8 port version for abour 5 years now. It is rock solid. The connected load metering is a very nice feature as well. The cost is about $380. Andres http://www.neuroredes.com Thanks Andres, I was actually was hoping to hear from somebody that has used the device that Steve orignally posted though! I've used several other APC products and I don't think I'd need to think twice about buying anything else from them. I do kind of think APC is a little like buying stuff from Sony... you pay a bit of a brand tax just to have the APC logo printed on it. However I think that the APC logo on something means alot more than the Sony logo :) -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Send cell phone #VM waiting, just like cell carrier
On June 22, 2008 11:32:56 am OCG Technical Support wrote: Well, I realize that there must be some proprietary protocol between the carrier and the phone, since they have a dedicate spot on the cell screen for # VM waiting... As for an SMS message, is there a module/app which allows easy SMS messaging? (I looked a couple of years ago but only found commercial modules) Don't quote me on this, but I think that the standard voicemail notifications that cell phones recieve are just a regular SMS message but in some kind of special format that the phone recognizes and does not display as a regular text message. I had an unlocked Moto razr at one point that didn't come from my cell provider, at one point while I had it the VM notifications didn;t really work... everytime i got a VM I'd get a SMS message with special characters in the message body. I sort of chalked it up to not using the branded firmware that my provider would have put on my phone if the phone had come from them. At any rate, if I'm right, and you can find some information on that protocol, it might turn out to be alot easier than you think. and btw - Asterisk ships with app_sms but depending on your country it may not really be what you are looking for. I could be very very wrong here, but I don't believe Asterisk has a great implementation for text messaging in North America. This is probably because every cell carrier seems to do it just a little bit differently. Depending on your needs however, most cell carriers offer email-to-SMS gateways so you can just send an email to yourcellnumber@yourcellcarrier.com kind of thing... if you need it to work from a machine with no Internet access however, then you might need to do something like use a dialup modem to dial into your cell carriers TAP number (provided they offer one and you can either explain to the tier 1 support guy what you are looking for, or you can find the number on the internet). However again, I don;t believe Asterisk has any interface for TAP, but in theory you could use something like the sendpage daemon to handle the TAP stuff and then have Asterisk hand off the text messages to sendpage using a simple bash script and the System() application... if Asterisk had an implementation for SNPP then you could send your messages to sendpage that way - also *some* cell providers even offer SNPP gateways over the internet, but again good luck explaining to the tech what you want. I've actually been toying with the idea of creating a res_snpp or something for this purpose since I already use sendpage for our server monitoring/alerting system to page me on my cell via dialup TAP (so my monitoring server can still page me in the event of an internet outage). Anyhow, sorry that message got a bit lengthy pretty fast! -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Controlling cell phone VM / Fax waiting notification icon for asterisk VM
On June 23, 2008 08:08:53 am OCG Technical Support wrote: I little more digging and I confirmed that cell phone VM and FAX waiting icons are in fact controlled by a proprietary SMS message format. Here's what I found: Message Waiting Indication Group: Store Message This Group allows an indication to be provided to the user about the status of types of message waiting on systems connected to the GSM PLMN. The mobile may present this indication as an icon on the screen, or other MMI indication. The mobile may take note of the Origination Address for messages in this group and group 1100. For each indication supported, the mobile may provide storage for the Origination Address which is to control the mobile indicator. Text included in the user data is coded in the Default Alphabet. Where a message is received with bits 7..4 set to 1101, the mobile shall store the text of the SMS message in addition to setting the indication. Bits 3 indicates Indication Sense: Bit 3 0 Set Indication Inactive 1 Set Indication Active Bit 2 is reserved, and set to 0 Bit 1 Bit 0 Indication Type: 0 0 Voicemail Message Waiting 0 1 Fax Message Waiting 1 0 Electronic Mail Message Waiting 1 1 Other Message Waiting* * Mobile manufacturers may implement the Other Message Waiting indication as an additional indication without specifying the meaning. The meaning of this indication is intended to be standardized in the future, so Operators should not make use of this indication until the standard for this indication is finalized. Now the tough part...does anyone want to create an app to send notification to a cell phone to set/clear these bits? could you provide a link to where you got the info from? I'd be interested in seeing if i can get this to do anything useful. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Codecs and app
On June 24, 2008 01:57:45 am troxlinux wrote: Hi list, recently install asterisk 1.4.21 in a centos 5, and after having installer the zaptel 1.4.10.1 and libpri 1.4.4 I don't see in the directory module any codec, and neither app. almost install all the asterisk options this worries to me ! alone I see these packages inside the directory app_addon_sql_mysql.so cdr_addon_mysql.so res_config_mysql.so app_saycountpl.so chan_ooh323.so format_mp3.so some help that they can provide me? Looks like you have installed asterisk-addons and not asterisk itself... if you compiled from source maybe you just forgot to 'make install' for asterisk? -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centos-5.2 and zaptel-1.4.11 do not get along well
On June 30, 2008 06:25:17 pm Mark G. Thomas wrote: Hi, After doing a yum update on my previously Centos-5.1 system, now zaptel-1.4.11 fails to build with this below. CC [M] /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o In file included from /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/xpd.h:26, from /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.c:27: /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/xdefs.h:117: error: conflicting types for ‘bool’ include/linux/types.h:36: error: previous declaration of ‘bool’ was here I;m not sure but I think somebody about a month ago had a very similiar compilation problem... infact I think it was the same error. I;m not sure what the proper way is to fix your CentOS box is, however an option for you might be to just not compile the xpp module. xpp I believe is the Xorcom AstriBank... if you don;t actually have an AstriBank then there is no sense in even compiling/installing the drivers for it. I;m guessing you haven't run a make menuselect to select only the drivers you need? -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disto choice for Asterisk with AVM Fritz!PCI cards
On June 30, 2008 08:44:44 pm Simon wrote: Hi There, I am looking to build an Asterisk server with dual AVM Fritz!PCI cards linked to 2 BRI in New Zealand. Just wondering if anyone has done this, and if you have any ideas about the best disto choice for this task? Let me be the first to say: Pick whatever distro you are comfortable with Distro is more of a personal choice than anything... ultimatly they all have the same software available to them (for the most part), they all just do it a little bit differently. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cell phone to PSTN adapter or IAX
On July 5, 2008 01:50:52 pm Joseph wrote: Are they any such things as Cell phone PSTN adapters? Openmoko is coming out so I hope it will be possible to register it to Asterisk and/or there will be a small iax registration program to communicate with asterisk. The beauty of the OpenMoko is that it is entirely open and completely hackable... they encourage people to modify it and write software for it... I imagine making a SIP/IAX client for it would probably be rather easy. That being said... since the OpenMoko runs Linux as its OS... its probably possible to run Asterisk on the OpenMoko... you could probably even use Asterisk as that SIP/IAX client using chan_alsa to access the built in speaker/microphone... (I;m assuming they are accessible through ALSA). I don;t know the OpenMoko inside and out... i;ve done some light reading on it over the last year or so... so i;m really just making some logical assumptions here. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Polycom SpectraLink 8002 Wifi SIP Handset
On July 5, 2008 09:00:57 pm Julian Yap wrote: Nice review. Can this phone be provisioned without using TFTP? For example, over the internet? Are there other provisioning methods? Theres no reason that you can;t do TFTP over the internet... Not so certain i'd recomend it since TFTP is authentication-less, but its still possible... and if security is in mind you could limit your TFTP server to specific source IPs -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cell phone hangup not getting recognised by system
is it only cell phone calls that don't work? or is it any external call coming in over your lines? What type of inbound lines do you have? I;m guessing analog lines... if thats the case what type of signalling are you using? if its only cell calls and not all external calls then I have no idea what it'd be... but if its all calls then its probably a signalling problem... you might be using loop start when you should be using kewlstart... or it might be that you need to get disconnect supervision added to your lines (essentially converting loop start - kewl start) -- Matt http://www.mattgwatson.ca On Wed, Jul 9, 2008 at 3:08 AM, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi Kate, have you tried the busydetect parameter in zapata.conf? Take a look here for other useful parameters: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf Giorgio. Lists wrote: Hi all, When I do a test call into the box (which is running latest version of Trixbox) it all goes fine. If i decide to hangup the cellphone (during the ivr playing options) the system does not recognize the hangup and the system continues through and ends up at the timeout option. What settings do I need to change to fix this. Is it the rxgain? If so is there something i can use to figure out what the gains should be set at? Thanks Kate ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (announce) asterisk T.38 gateway
I'd probably be a little pissed if I were Steve Underwood if somebody pocketed over 10k $USD for taking credit for a product that my free library did the bulk of the work for. I don;t think i'd feel that the entire bounty should be mine - after all there would of been nothing stopping me from doing it myself... but credit should be given where credit is due. Even if its just something as a sign of appreciation. Given that spandsp is GPL'd Steve obviously never intended to make a ton of money off of it... but I;m sure he'd love to receive something for his work, or use some of that money to further develop spandsp. That being said... i;m also quite pleased to see T.38 support being worked on for Asterisk... its a pretty important area to further develop IMHO. -- Matt Watson http://www.mattgwatson.ca On Thu, Jul 10, 2008 at 11:54 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Thu, Jul 10, 2008 at 11:43 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Thu, Jul 10, 2008 at 10:24 AM, Steve Underwood [EMAIL PROTECTED] wrote: Vinícius Fontes wrote: When people release software under the GPL license, like Steve Underwood did with libunicall, spandsp and so on, they were supposed to know that other people has the right to use their code. The problem is that almost any licence term which tries to limit the obnoxious behaviour of other people has too many unpleasant side effects. GPL 2.0 is the best compromise I've found, so that is what I used for everything unless recently. To make my stuff licence compatible with FreeSwitch I recently relicenced most of my work as LGPL 2.1. This is having undesirable consequences, though. Its really a tough issue, and GPL 2.0 showed immense foresight in just accepting the non-existence of perfect solutions. GPL 3 seems to have forgotten the lesson somewhat. Most of the time I just want to give up producing anything at all. Steve So are you angry that he may gain monetarily from your your work, or is it hurt pride that he is basically taking credit for it? The answer to that should guide you in how you release your work in the future. Thanks, Steve Totaro I also want to add that if someone asked me to name the top five names that came to mind when thinking of Asterisk, Jim Dixon, Mark Spencer, Steve Underwood, Nicolas Gudino, and I will leave off the fifth as to not leave anybody out ;) Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls on zaptel not answered.
On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote: After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop working. THis isn;t going to fix your problem... but just FYI, you don't need to install libpri if you are just using a TDM400P (since its not a PRI / BRI [1.6 libpri does BRI as well] card). Might save you a little bit of time in the future, and its one less thing to consider as a problem. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium PRI and Echo cancellation
On July 16, 2008 08:01:38 am Loic Didelot wrote: Hello, I would like to double check what Echo Cancellation my Digium Card uses. I thought I bought the little more expensive card that integrates EchoCancellation. How can I check? If you load the modules with debug=1 (maybe this appears without it too... not sure): Apr 23 12:52:55 nelson VPM400: Not Present Apr 23 12:52:55 nelson VPM450: echo cancellation for 64 channels Apr 23 12:52:55 nelson VPM450: hardware DTMF disabled. Apr 23 12:52:55 nelson VPM450: Present and operational servicing 2 span(s) Apr 23 12:52:55 nelson Completed startup! Thats on my TE220B. the VPN450 is the hw echo can daughter board. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium PRI and Echo cancellation
I believe HPEC actually is the same algorithm (G.168) that the HW echo cancel modules use.. the difference being that HPEC uses up CPU cycles and its performance will be impacted on a system with higher CPU load, whereas the HW modules have a dedicated DSP for it. http://blogs.digium.com/2007/09/06/fun-with-hpec/ -- Matt On Thu, Jul 17, 2008 at 10:00 AM, Noah Miller [EMAIL PROTECTED] wrote: Hi Loic - According to that its using MG2. I think it will say MG2 regardless of whether or not there is a hardware module present. Shouldnt it be using something like HPEC? I don't think the hardware echo cancellers use the HPEC algorithm. As Eric and Matt have mentioned, dmseg will tell you if a hardware echo cancel module is being loaded. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
On July 19, 2008 11:22:08 am Mark Wiater wrote: Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the Mitel. Or so it seems. The receiving Mitel phone thinks the call is in session though. Asterisk is reporting errors like: [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' Might want to post a sip debug of one of the sessions from the Mitel phone. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Visual Dial Plan
I've seen it before infact there is a website setup where people can post stuff made with it... kind of super nerdy! http://www.ratemydialplan.com -- Matt http://www.mattgwatson.ca On 7/27/08, Peter Lindquist [EMAIL PROTECTED] wrote: Dean Collins wrote: I just stumbled across this on youtube. Does any on the list us it? This is the first I've heard over it. http://www.youtube.com/watch?v=H1j5OrgL1og Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (New York) +61-2-9016-5642 (Sydney) http://www.Cognation.net http://www.cognation.net/profile Yes, I use it and it is a great tool I think. If anything I do miss an ability to print out the graphical representation of the dial plans in the current version - this is being worked on though. Best regards, Peter Lindqvist Voxion Ltd. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Aastra 480ci and qualify=yes
I;m using Aastra 480i's 9133i's, 9112i's, and 57i's and none of them have experienced problems with qualify=yes. I;m currently on Asterisk 1.4.17, but I've also tested them with 1.4.14 up to 1.4.19. -- Matt http://www.mattgwatson.ca On Fri, Aug 15, 2008 at 10:59 AM, Drew Gibson [EMAIL PROTECTED] wrote: James Lamanna wrote: Hi, We have a few Aastra 480ci phones and we've noticed that in order to get the phone to receive a call, qualify must be = no. Apparently the Aastras do not respond to the qualify message (or respond in a way Asterisk doesn't understand) and Asterisk thinks the phone is unreachable. However, this now prevents MWI from working properly on the phones. Does anyone know how to get MWI working without qualify? Or how to get qualify working again with the Aastras? We have a number of 480i and one 480ct all setup with qualify=yes (Asterisk 1.2.24) Our inbound call centre seems to be pretty busy and my own MWI lights up far too often. Never had a problem with either. Which version of firmware? Which version of Asterisk? What's in your sip.conf? What error messages show on the console? Anything relevant in the logs? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi-monitor in France
1. Ask your telco, they probably have them, but you may have some difficulty in finding somebody at your telco that has a clue about what you are talking about. You can find some lists doing some google searches for the numbers and hope to get lucky... but as far as I know, there is no official repository for these test numbers. 2. I wouldn;t use an overseas number personally... those calls are certainly getting encoded / decoded and reencoded several times, and more than likely getting compressed, all of which is going to have an impact... it *might* be better than nothing... but i would expect very poor results. 3. You are right, you can';t really just make one yourself from scratch, you need a source that has already been tuned properly to use as a reference for creating your own. -- Matt Watson On Thu, Dec 11, 2008 at 11:01 AM, Olivier [EMAIL PROTECTED] wrote: Hi, I would like to tune rx/tx gains using dahdi-monitor for a system which will be connected to french PSTN. I'm not aware of any public phone number in France I could call to get a normalized 1004Hz signal. My questions are : 1. Does such numbers exist ? Is there a directory somewhere listing some of them ? Do you think regulations could make providing such numbers mandatory for (some) Telcos ? 2. Does it make to use a number aboard instead if I can't find any local ones ? I don't think so, but I prefer to check. 3. I can't imagine a process allowing me to create my own (chicken and egg problem). Is it correct ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI install dont need download of echo cancel
after you have configured zaptel manually the first time, copy the menuselect.makeopts file that is generated in the root directory of the zaptel source to a file /etc/zaptel.makeopts. presumably this is available for people that have moved on to DAHDI as well, and I would guess it should be /etc/dahdi.makeopts - but I have not verified that. -- Matt Watson On Thu, Dec 18, 2008 at 11:49 AM, Jerry Geis ge...@pagestation.com wrote: Jerry Geis schrieb: / Is there a way to install DAHDI and NOT download the echo canceler files? // I dont have firewall access and its failing. // I dont need the files as there is no card installed. // // How do I get past this? / If I remember correctly you can un-check them in `make menuselect`. Philipp Kempgen Thanks I did not look there. Is there a way to do it in an automated fassion. (doing more in the future). Dont want to have to manually change menuselect. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Root Password not taking
On Thu, Jan 22, 2009 at 11:32 AM, Max Brooks m...@legatio.com wrote: Jim Dickenson wrote: What I have done in the past to set the password for root is to boot in rescue mode and edit /etc/shadow setting the password to some know value from another system. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ I personally prefer to chroot into the / partition and run passwd. Yep, thats pretty much the best way, and more or less one of the only methods that is going to work regardless of Linux distribution, or other UNIX variant. Many distros now, like most of the UNIXs actually still require your root password when booting single user mode - as they should. Gaining root access to a system even with physical access to the machine *should* be more difficult than simply picking a different grub boot option. I realize that is not the case across all distros, but IMO it should be. For distros that do require a root password when booting single user mode, your only real options have already been mentioned here... 1) boot from a CD, mount your partitions then: a) manually edit /etc/shadow (Linux only) and change the field containing the encrypted password with another encrypted password that you know what the uncrypted version is b) chroot into your mounted partitions and then run passwd as normal (this should be work almost all UNIXs) (b) is the more generic and preferred method IMO - it should work just about everywhere... unless you have total disk encryption or encrypted filesystems and are unable to mount the partitions... in which case... best of luck to you. -- Matt Watson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] random Linksys question
Yes, it is available on the SPA2102 - you just login to the web interface, goto the advanced section, then lan setup... its the very first option. -- Matt On Thu, Jan 22, 2009 at 3:29 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: On Thu, Jan 22, 2009 at 3:11 PM, Jeff LaCoursiere j...@jeff.net wrote: Can you configure the LAN port on the back of a 2102 to be bridged rather than routed to the WAN port? To my knowledge this is available on all Linksys ATA type devices that offer both ports. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Root Password not taking
On Thu, Jan 22, 2009 at 1:48 PM, Wilton Helm wh...@compuserve.com wrote: There have been a number of answers provided. The one that was given to me when I encountered this same problem was to boot a live CD, mount the root file system and delete the password file which would force your normal distro boot to request a new root password next time. I'd highly recommend to NOT simply delete your password files (/etc/passwd /etc/shadow)... you are deleting a lot more than just your compromised your root password by deleting these files. You are also deleting every other account that is on your system (ie. you should be running Asterisk under its own account and not as root!). Of course it depends on what other things are running on your server, but you cannot reasonably expect your distro to keep track of every change you have ever done to these and re-create it. It might give you a skeleton version like you are suggesting, but you are certainly going to lose data. Step 3 was replace a few key executables like ps so I couldn't do administrative tasks I can't really speak about your case specifically, but generally replacing certain executables like ps, netstat, login, w, who, etc. etc. are not done to prevent administrative tasks - infact, a person that actually knows what they are doing with a rootkit doesn;t actually want to prevent you from doing administrative tasks. Generally they don't want you to know at all that they have broken into your system. The reason why these binaries get replaced is to try and cover their tracks. for example, they replace 'ps' so that it will not display certain processes running on the system (making it harder for you to even notice), the same with netstat, it will not display network connections to certain IPs, 'login' gets replaced with a version that has a hard-coded login password that will grant root access to your system without knowing (or caring) what the real root password is, and might also not do a few things that are part of the normal login process... like writing to UTMP / WTMP so there is no record of the login. The problem is that you don't know exactly what files have been changed and if they have left a trap door or something. You could fix the root password, and even discover and restore a few key files, only to find it hacked 5 minutes later because you didn't know everything that had been altered. For that reason, few people will put a system back on line after the root password has been compromised. Re-installation is the only safe way. If some of your directories like /home and /user have separate mount points, they don't have to get wiped out in the process. This is getting a little off topic... but there actually are ways of determining exactly which files have been changed. Stuff like 'tripwire' has been around for like a decade and is designed to do exactly this... everytime you run it, it will tell you every single file that has changed. It does this by creating hash of every file when you run it and then compares those hashs to previously stored values. If the hashes do not match, then something has changed and gets reported. Of course hashes can be defeated, it is however not entirely easy, and way beyond what your typical script kiddie hacker is going to be capable of doing. And of course, that is just what trip wire does... there are plenty of other intrustion detection systems that I;m sure do things a little differently. Some other things like FreeBSD used to (maybe still does?) come with some basic security stuff that was installed by default and emailed you daily reports about things that changed. Like any new binaries it found on the system that were setuid root (ie. cp /bin/bash /var/spool/.myhiddenrootshell ; chown root /var/spool/.myhiddenrootshell ; chmod u+s /var/spool/.myhiddenrootshell). Something like FreeBSD's scripts were intelligent enough to find basic things like that. That being said, I do agree with you, re-installation in 99% of cases is a superiour option when compared to trying to restore a compromised system. Typically its much quicker too. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Root Password not taking
On Thu, Jan 22, 2009 at 1:52 PM, David @ULC ucoms2...@gmail.com wrote: I tried : 1. Shut down the machine. (Ctrl+Alt+Del) 2. When it reboot and reach the CentOS boot up screen, then press any key to go into a select menu. Then press e and navigate to the second line grub.conf line (kernel) and press e to edit the line to option 1 : SPACE 1 ( 1″) at the end of the line. OR option 2: SPACE S ( S) a the end of the line OR option 3: add single to ro root=LABEL=/ single Your error is at option #3, the root= line should be pointing to the partition of your root (/) filesystem. You are supposed to replace LABEL with it - typically this is going to be something like /dev/sda1 (if you are SCSI or SATA - /dev/hda1 if its IDE)... though I believe CentOS uses initrd, and it might be a little different when using initrd, i;m not very familiar with it so hopefully somebody else will chime if it should be something different. but it also may not be sda1... it could be sda2, sda3, sdb1, sdb2, sdz4, hdg3... in otherwords, too many options to list, this is specific to your system - but sda1 would be typical. Then hit ENTER and press 'b' to reboot. After it reboot, and stop at '# command line, type passwd to create the new root password. Reboot the machine as usual and access your root with new password. On Thu, Jan 22, 2009 at 9:22 PM, David @ULC ucoms2...@gmail.com wrote: In one of my center , its not taking root password. Anyways to recover it ? In other terms , I lost the control of server. Any solution or re-installation is the only way left ? I am using CentOS. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Root Password not taking
On Thu, Jan 22, 2009 at 5:30 PM, Wilton Helm wh...@compuserve.com wrote: Tripwire would be fine, if it had a baseline, but I don't think its any good after the fact. Correct - tripwire does need to be setup beforehand, and its not the most simple thing to setup *properly*. After the fact... you are basically out of luck unless you are using a binary-based distribution and want to re-download all the packages and compare the hashs... but thats simply too time consuming and reinstalling is faster and more reliable. -- Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Root Password not taking
On Thu, Jan 22, 2009 at 6:05 PM, Wilton Helm wh...@compuserve.com wrote: making sure to patch any holes through which the hacker might have come In my case, I had been getting regular attacks through SSH for months, probably 100 a day (bots). Apparently after nine months of this, someone stumbled on to my password which regrettably was composed of two dictionary words with no special characters, making it susceptible to dictionary search. When I re-installed, I put SSH on a non-standard port and haven't had a single attempt to attack the system since. While its certainly not a good idea to rely on... security through obscurity can work very well :) -- Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 T38 to G711 transcoding is this possible?
On Sat, Jan 17, 2009 at 11:51 AM, Steve Gladden aster...@michiganbroadband.com wrote: The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and reliably? Even if you find something that claims to be capable of doing this, it will not work reliably. T.38 is a protocol designed specifically to overcome the challenges of faxing over IP networks. Without it, depending on how good of a network and route you have to your ITSP, your results will either be: A) won;t work at all due to unpredictable network connections B) Will work sometimes occasionally most of the time - all of which are not suitable answers in an business environment. It will almost certainly never be works flawlessly In short, you cannot fax reliably over IP without T.38... you might want to get an additional ITSP account with another provider and use it strictly for faxes. I;m sure there are lots that support T.38, but the only one I know that a friend uses is Gafachi. Your other option is to get a POTS line into your * box, in which case (I believe) Asterisk 1.6 has the ability to act as a T38 Gateway for this purpose. It does not exist without 3rd party modules for 1.4 I believe though... unless somebody has backported all the new T.38 stuff in 1.6. -- Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TAPI and Asterisk
There is a Digium solution to this as well... Digium recently acquired snap-a-number and has put own their own version of it already called Asterisk Desktop Assistant It does exactly what you are looking for plus some more... Information is still a little scarce since it is pretty new, but there is some limited information as well as some manuals that should give you enough information to get it going (its actually very easy). http://forums.digium.com/index.php?c=8sid=0a71abedc29238acee530a0461c914b0 Oh yeah, its also free (as in beer). Though I am not clear if its going to stay that way or not... But I haven;t seen anything that would indicate otherwise. -- Matt Watson On 30/01/09 11:11 AM, Jeff LaCoursiere j...@jeff.net wrote: Following up my own thread, I am kicking myself for quickly posting without doing a bit of research. Apparently (no surprise) this integration of Outlook and Asterisk is very old news, and there are many projects out there. Anyone dealt with Thirdlane? http://www.thirdlane.com/products/thirdlane-dialer This seems to be just the ticket... Cheers, j On Fri, 30 Jan 2009, Jeff LaCoursiere wrote: Funny how a topic will come up that you have never dealt with before, and suddenly it comes up from multiple directions at the same time. I was recently involved in a meeting where TAPI (which I understand only vaguely) was proposed as way to link a custom application to Asterisk for outbound and inbound call processing, much like SugarCRM and probably others are doing. Today I was asked by an existing client if I knew a way to synch their mobile device contacts with the system in some way so that they would have quick access to speed dial or otherwise call up a personal directory on their (Polycom) phones that could be synched in this manner. It struck me that the Polycom directory interface is a bit kludgy for such things, having no search capability and no sorting capability once loaded that I am aware of. Given the meeting last week it seems that a more elegant solution would be to link Outlook itself with Asterisk, so right clicking a contact makes it possible to launch an outbound call. That would take care of integrating a WHOLE LOT of devices, as (sadly) the MS contact database would be the go-between that all of these devices synch with in one way or another already. Is TAPI the right protocol to investigate for this purpose? Would something like Fonality's HUD software bridge this gap? Has this wheel already been invented? Hoping for some thoughts! Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel telephone cards and asterisk in another pc
Your boss is going to change their mind when they see how awful and unreliable this setup is going to be lol. On Fri, Feb 20, 2009 at 1:42 PM, Ignacio sanfermi...@gmail.com wrote: yep, it is mainly due to cabling issues. My boss doesn't want to recabling the office. On Fri, Feb 20, 2009 at 7:19 PM, Jeff LaCoursiere j...@jeff.net wrote: Why do you want the cards in the client machines instead of the main asterisk server? Cabling issues? This sounds like it will be a horrible spaghetti mess... j On Fri, 20 Feb 2009, Ignacio wrote: Thank you very much for your fast answer Eric. I was trying to avoid to have to install as many asterisk as pcs I have. But I think there is no way to do it. I only have seen something like network block device, but not sure if it is going to work and quite difficult to configure properly. Anyway I think the fast and easier way will be installing and asterisk in every client. Thanks again. On Fri, Feb 20, 2009 at 6:41 PM, Eric Wieling, Asteria Solutions Group ewiel...@asteriasgi.com wrote: Ignacio wrote: I have some zaptel cards, and I would like to install them in some user's computers. Is there any way to connect those cards with asterisk server (which is in another computer)? All manuals I have read explain how to connect asterisk and zaptel cards in the same computers, but not on different ones. You can install Asterisk on the PCs with the cards in them and then use SIP or IAX2 to transport those calls to your main Asterisk server. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] building asterisk-1.6.0.6 failed!
I find it a little strange that for some reason your box is using includes located in /usr/local... while there could be reason for this, that seems like a sign that something might be a little broken on your box. Also, if you don;t mind me asking... why would you want to install * directly in /usr? I could undersatnd if you are building a distribution package or something, but personaly, i would install to /usr/local or even some special place just for * just to help keep the box more organized. -- Matt On Mon, Feb 23, 2009 at 7:17 PM, Tamer Higazi th9...@googlemail.com wrote: Hi! I have problems building asterisk 1.6.0.6. ./configure --prefix=/usr make gets me: enerating embedded module rules ... [CC] extconf.c - extconf.o In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file or directory In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t' /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here make[1]: *** [extconf.o] Error 1 make: *** [utils] Error 2 Now, I think this is only a dependency problem. could anyone of you tell me, which and where I am able to get the missing sources to successfully compile asterisk?! Tamer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDD FULLL
On Mon, Feb 23, 2009 at 9:38 PM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 24 Feb 2009, David @ULC wrote: When I am trying to delete voice logs, [r...@vicidialnow monitor]# rm * -r -f -bash: /bin/rm: Argument list too long In the past 30 days, you've asked questions about configuring Apache to process PHP files, Vicidial, Ntework Cards, Auto Detecting hardware, BAT files on CentOS, Root Password not taking, How to find a file, Looking for a Free VOIP Billing and Soft Switch, What is a VPN, How do delete files, oh, and a couple of Asterisk questions. This is an Asterisk users list. We're here to help each other with Asterisk questions and problems, not to be your personal, for free, life coach. If you are being paid to work on an Asterisk system, you are in over your head. You are defrauding your boss and most likely will give him and everyone in the company a bad impression of Asterisk. Continuing to answer your questions will only continue to enable you. Please take a step back, buy some books, take some courses, practice on your own systems on your own time. Just have to say it: I totally agree with you and was thinking the same thing as soon as I read that they tried rebooting the machine thinking that would free up space on the HD... didn;t even realize this was the same person that had posted some of those other threads -- Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra 9133i programmable buttons (* 4.1.23)
The 1.x firmware for Aastra's (for the 9112i / 9133i / 480i) do support some of the XML functionality that you see in the newer 2.x firmware (for the more recent models). I;m not sure if controlling LED status of the keys is supported by 1.x - but you should be able to find that out by taking a look at Aastra's XML API document here: http://www.aastra.com/cps/rde/xbcr/SID-3D8CCB6A-2E5763F4/04/Telecom_PA-001004-00-03_XML_Development_Guide_Release_1.4.2.zip -- Matt On Mon, Mar 16, 2009 at 1:34 PM, Steve Davies davies...@gmail.com wrote: 2009/3/16 David Ruggles da...@safedatausa.com: Is it possible to control the light on a programmable button without the blf option? I'm using a programmable button to turn call recording on and off and I would like the light to indicate the status. Thanks, 9133i phones are pretty much obsolete, and are not getting firmware updates, so I do not know whether Aastra ever put any of their XML application control code into that model. If they did, then it should be possible to respond with button status using XML updates from the server, otherwise you'd need to upgrade to one of their currently supported phones, which are almost certainly capable of this sort of thing. PS. I have never personally used the XML facility of Aastra phones, but I hear quite good things about it. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail and remote directory with SSHFS
Not that I;m exactly a big fan of NFS but... why would you choose to implement a filesystem that was designed to emulate Windows shares for your UNIX-type environment? You have to kind of expect odd problems like this when you choose to use things for other than their intended purpose. Samba I would say is probably alot more focused on providing storage shares for Windows desktop clients, not for UNIX-type clients. Sure there is some support to do what you want, but just keep in mind that similiar to using sshfs like you were trying before, Samba, was really not designed to be used by UNIX clients. You've already found the most obvious reason... case sensative filenames - which Windows does not support, and UNIX programs expect filesystems on your UNIX machine *will* support it. That seems kind of like me deciding to use ntfs on a local partition on linux box instead of ext3/4, jfs, reiserfs, etc. -- Matt On Thu, May 21, 2009 at 5:06 AM, Elliot Murdock murdo...@gmail.com wrote: Hello! Thanks...I set up a Samba mount, which works ok, except that Asterisk confuses a wave file as a wav49 file. I think it may have something do with the way Samba supports case sensitivity. Since Windows is not very aggressive when it comes to being case sensitive, I am thinking that Samba is saving files with the last three characters, wav, as uppercase, WAV. What is the procedure to ensure all the files are saved as is in Samba? Thanks, Elliot On Thu, May 14, 2009 at 5:12 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Thursday 14 May 2009 08:14:17 Elliot Murdock wrote: The problem is a file locking problem that Asterisk needs to make changes to the directory. I was initially shying away from NFS and Samba, because I prefer to avoid any sort of security issues with only remotely mounting one or two directories. NFS and Samba are designed for larger applications, which makes those types of technology worthwhile. No, they're both designed as filesystems, which makes typical things like locking possible. SSH is designed as a communications medium, and someone has hacked filesystem support on top of it (poorly, apparently). SSHFS was never designed to be used in server production environments and should not be used there. I am wondering if there is any way to disable Asterisk's request to lock the directory. I know this may cause some loss in data, but for the volume voicemail receives, it should be rare enough that would make this approach an option. There is not. Use a real filesystem that supports file locking (or really, file linking, which is how the locking is implemented) procedures. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] From 1.4 to 1.6.0
I'd be interested in this as well... I;m coming up to an upgrade cycle and trying to decide if I should upgrade to the latest 1.4 or 1.6.1 When others that have commented on this say they have had problems with PSTN connections, are you referring to T1 or POTS? I;m in a T1 scenerio, so if problems are specific to POTS then thats obviously not a deal breaker for my setup. Thanks, -- Matt On Thu, May 21, 2009 at 10:02 AM, Danny Nicholas da...@debsinc.com wrote: My issues are all DAHDI/POTS related. Unfortunately, our present communication depends on the POTS lines, so I’m back to 1.4.25-rc1 as stated earlier. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Thursday, May 21, 2009 8:53 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] From 1.4 to 1.6.0 2009/5/18 Danny Nicholas da...@debsinc.com I'd love to see this as well. After a few days of trying 1.6.1 (from 1.4.21) I dropped back to 1.4.25-rc1 and that is going pretty well. Which issues did you get ? I'm about to deply a 1.6.1 system it does seem to work ok in a pure SIP environment. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: Monday, May 18, 2009 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] From 1.4 to 1.6.0 Hi everyone, I was just wondering, does anyone managing production asterisk servers migrated successfully from 1.4.2X to 1.6.0.X? I would like to see if there are some successful cases. Is your 1.6.0.X behaving well, with acceptable stability? Please share your experiences. Thanks, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parsing Asterisk's .conf files from Perl, Java or PHP file
There already is a special character to tell asterisk not to parse a line... its: ; that is why the default configuration is completely filled with lines that start with ; its considered a comment character to asterisk and will make it ignore the line... you'd just want to add some extra characters to your program probably... something that would denote this is a config line for script A so if you make your lines something like ;A: then you could have another character for script B that would have lines in the same file like: ;B: or w/e... its up to you to determine what characters you'd want to use beyond the ; to denote config lines for your other programs. -- Matt On Fri, May 22, 2009 at 3:23 AM, Olivier oza-4...@myamail.com wrote: Hi, To a large extend, Asterisk's /etc/asterisk/*.conf configuration files conform to a format such as: [section1] key1=value1 key2=value2 [section2] key1=value1 key2=value2 ... To increase coherence when running custom-made application in Perl, Java, PHP, ...) and Asterisk on the same platform, I'm wondering if could extend a bit Asterisk's config files instead of duplicating data in an independant config file. For instance, an app that uses Manager interface needs to be configured with : - the Asterisk manager interface IP address, - a username and secret. The later 2 parameters are included /etc/asterisk/manager.conf but the first one is not. So instead of writing an independant myapp.conf holding all 3 parameters, should I only add the first parameter to existing manager.conf file ? Doing this, I would have to make sure that when Asterisk is parsing its config file, it doesn't stop when it reads unkown supplementary parameters (those added for custom app). It seems to be the case now with 1.6.1 : a NOTICE warning is sent but it doesn't really hurt. [May 22 09:15:32] NOTICE[15917]: manager.c:3903 __init_manager: Invalid keyword foo = bar in manager.conf [general] Maybe, adding a keyword that would tell Asterisk to skip reading this config file line would be a plus (avoiding warnings and collision with new keywords). What do you think ? Then, my next question, is there widely available librairies to parse Asterisk's config files-like files ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 fax solution, opinions?
I;m not sure how your solution would work... but I thought I'd throw out some ideas that we are having to implementing faxing here on a new install. We are going to be bringing in a PRI and routing all the DIDs from our existing copper lines to the PRI (including fax DIDs)... the solution we are working towards is certainly not ideal, but we are hopeful its going to work... Incoming fax calls will come into Asterisk, asterisk will route them to IAXmodem which will feed Hylafax (all running on the same box as Asterisk to reduce latency... and we are talking about fairly small volumes). Hylafax will then do fax 2 email Outbound faxing however is a bit trickier... We are going to use Linksys ATAs (tested with a SPA2102) which will have the POTS fax machines plugged into them, the SPA2102 connects to asterisk with SIP, asterisk will then route the calls to t38modem (recent dev versions of it support SIP and not only H.323), t38modem is basically just like IAXmodem except its SIP and supports t38 termination. T38modem will again feed hylafax, which will then route back to asterisk through IAXmodem and then up the PRI. Its certainly not a pretty solution... but we haven't come up with anything else yet the only step I can see possibly simplifying is on outbound faxes Hylafax can possibly be bypassed and have t38modem talk directly to IAXmodem. -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of arkda Sent: Sunday, December 09, 2007 11:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T.38 fax solution, opinions? Hi, I'm putting together a fax solution for my company that utilizes T.38. I wanted to throw out my plan and get some feedback if anyone is doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by VoIP providers (for voice Vitelity inbound, Gafachi outbound). These satellite offices are using a T.38 fax DID from Gafachi, passed through the Asterisk server to a Linksys 3102 ATA and then to a POTS fax machine. This all works well thus far. Our HQ has a full voice PRI, terminated on the Asterisk server with a TE120P. Additionally, right now they have five fax lines totally separate from the PRI that are used for POTS fax machines. I'm thinking of porting those numbers to the PRI and purchasing a TDM880B (comes with eight FXS modules) and routing the fax DIDs to the 880 in Asterisk. Five of the ports would connect into a Linksys 3102 that would speak T.38 to what would be our new fax environment (Exchange 2007 Unified Messaging). That part isn't implemented yet, but it shouldn't be a problem. Once it's implemented I'll probably reroute the Gafachi T.38 fax DIDs to Exchange through Asterisk (with sipX in there somewhere). The part(s) I'm unsure about is the TDM880B. I haven't used a FXS card with Asterisk, and I certainly haven't used a fax machine on that FXS. Additionally, I'm not 100% sure the 3102 will talk directly to Exchange UM yet, but that's something I can figure out myself soon; I'm just not sure about spending the cash for a TDM880B without knowing someone has thrown faxes through it from a PRI terminated on the same box from a separate card. Anyway, thoughts, criticisms, insults and stinging barbs all welcome. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 fax solution, opinions?
The fax machines will be talking directly to the spa2102 - the problem is that Asterisk only supports being a T.38 pass-thru and not an end-point. And I need the ability to fax over WAN links like ADSL... so I'll have a fax machine plugged into a Linksys SPA2102 which will connect to Asterisk over ADSL, if I route the calls directly to the PRI and use G.711u for the Linksys - Asterisk connection, then any hiccups on the DSL line are going to cause the fax to potentially get screwed. This is what T.38 is designed to fix... but since Asterisk doesn't support being an end-point... you need something else to do it... that could be a SIP provider that supports, t38modem, another ATA, etc. That's why our planned setup is way overly complicated... just to get T.38. -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Augustyn Sent: Monday, December 10, 2007 4:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] T.38 fax solution, opinions? How about fax machines talking directly to spa2102 and then out the pri or am I missing something? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Watson Sent: Monday, December 10, 2007 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T.38 fax solution, opinions? I;m not sure how your solution would work... but I thought I'd throw out some ideas that we are having to implementing faxing here on a new install. We are going to be bringing in a PRI and routing all the DIDs from our existing copper lines to the PRI (including fax DIDs)... the solution we are working towards is certainly not ideal, but we are hopeful its going to work... Incoming fax calls will come into Asterisk, asterisk will route them to IAXmodem which will feed Hylafax (all running on the same box as Asterisk to reduce latency... and we are talking about fairly small volumes). Hylafax will then do fax 2 email Outbound faxing however is a bit trickier... We are going to use Linksys ATAs (tested with a SPA2102) which will have the POTS fax machines plugged into them, the SPA2102 connects to asterisk with SIP, asterisk will then route the calls to t38modem (recent dev versions of it support SIP and not only H.323), t38modem is basically just like IAXmodem except its SIP and supports t38 termination. T38modem will again feed hylafax, which will then route back to asterisk through IAXmodem and then up the PRI. Its certainly not a pretty solution... but we haven't come up with anything else yet the only step I can see possibly simplifying is on outbound faxes Hylafax can possibly be bypassed and have t38modem talk directly to IAXmodem. -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of arkda Sent: Sunday, December 09, 2007 11:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T.38 fax solution, opinions? Hi, I'm putting together a fax solution for my company that utilizes T.38. I wanted to throw out my plan and get some feedback if anyone is doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by VoIP providers (for voice Vitelity inbound, Gafachi outbound). These satellite offices are using a T.38 fax DID from Gafachi, passed through the Asterisk server to a Linksys 3102 ATA and then to a POTS fax machine. This all works well thus far. Our HQ has a full voice PRI, terminated on the Asterisk server with a TE120P. Additionally, right now they have five fax lines totally separate from the PRI that are used for POTS fax machines. I'm thinking of porting those numbers to the PRI and purchasing a TDM880B (comes with eight FXS modules) and routing the fax DIDs to the 880 in Asterisk. Five of the ports would connect into a Linksys 3102 that would speak T.38 to what would be our new fax environment (Exchange 2007 Unified Messaging). That part isn't implemented yet, but it shouldn't be a problem. Once it's implemented I'll probably reroute the Gafachi T.38 fax DIDs to Exchange through Asterisk (with sipX in there somewhere). The part(s) I'm unsure about is the TDM880B. I haven't used a FXS card with Asterisk, and I certainly haven't used a fax machine on that FXS. Additionally, I'm not 100% sure the 3102 will talk directly to Exchange UM yet, but that's something I can figure out myself soon; I'm just not sure about spending the cash
Re: [asterisk-users] MWI and 1.6.1
I'm having this EXACT same problem, I haven;t been able to narrow down the cause of it yet, but it seems to me that users are receiving notifications for voicemails in mailboxes that belong to other people, as sometimes their mail count magically disappears, which I have been suspecting is when somebody else checks their VM. I found the problem also exists in 1.6.2 which is where I first noticed it (upgraded from 1.4.x to 1.6.2.x). I tried downgrading to 1.6.1 and the problem seemed not quite as bad, but I know its still present. I was actually quite surprised to find that nobody had previously mentioned the problem on this list when I came across it so I thought it might of been something specific to my situation. Even if you turn the polling options back on in the voicemail conf file the problem still persists. We are using all Aastra phones - a mix of 9133i, 9112i, 480, 35i, 57i phones - but the problem seem unrelated to the make/model of the phone based on seeing you having the same problem with Polycom's. Not sure that it should matter, but we are using FreePBX 2.6 ontop of asterisk and running it in users and devices mode (as apposed to the default extensions mode). If you do a voicemail show users from the Asterisk console it shows the correct VM counts for the mailboxes, so its not that Asterisk is counting them incorrectly, it just seems to be sending the notifications of VMs to the wrong places. I'm suddenly very glad I;m not alone on this one! I;m more than happy to do any testing of patches if anybody has any suggestions. -- Matt On Tue, Mar 2, 2010 at 1:36 PM, Dave Poirier dpoir...@mesd.k12.or.uswrote: We are having an issue with Asterisk 1.6.1 and the MWI turning on when a user doesn't have voicemail. We see random MWI lights come on and the phone indicates a random number of messages (its been anywhere from 1-14) when a server reload is done. I just checked one user, they have no messages old or new and the phone (Polycom IP330) indicates that they have 2 messages. The user will check for messages, the system will tell them that they have none and the light goes out. I know that starting in 1.6 Asterisk moved from a polling system to an event based system but it's unclear to me what is causing these events to be generated. Anyone else experience this? Any tips, suggestions? Thanks, Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.
Awesome! I was an Attrafax customer and was very disappointed when it vanished and couldn;t get new modules for newer versions Asterisk with our paid license. If anybody is working on t38 gatewaying code for 1.6, it would be worth a look at this, as I can attest that Attrafax worked quite well at t38 gatewaying. -- Matt On Sun, Mar 7, 2010 at 4:52 AM, Zoa zoach...@securax.org wrote: On friday we finally released Attrafax under a GPL2 license. It comes with its own set of modems and built in transparent gatewaying. The solution should be quite stable as long as the line quality is ok. (Some tools for measuring the line quality are included in the release, as well as some fax2mail scripts). There is an example implementation included for Asterisk 1.4, if someone wants to porting it to the new fax backend or more recent asterisk versions and needs some help, let us know. The full press release can be found here: http://www.zoiper.com/downloads/opensource_fax_stack_PR.pdf the project homepage can be found at www.zoiper.com/foip/ Cheers, Zoa -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.
I just downloaded a copy of this, by any chances does Zoiper by any chance have diff files available for a more recent 1.4.x release? (I know 1.6 is probably out of the question) Thanks, -- Matt On Mon, Mar 8, 2010 at 12:11 PM, Matt Watson m...@mattgwatson.ca wrote: Awesome! I was an Attrafax customer and was very disappointed when it vanished and couldn;t get new modules for newer versions Asterisk with our paid license. If anybody is working on t38 gatewaying code for 1.6, it would be worth a look at this, as I can attest that Attrafax worked quite well at t38 gatewaying. -- Matt On Sun, Mar 7, 2010 at 4:52 AM, Zoa zoach...@securax.org wrote: On friday we finally released Attrafax under a GPL2 license. It comes with its own set of modems and built in transparent gatewaying. The solution should be quite stable as long as the line quality is ok. (Some tools for measuring the line quality are included in the release, as well as some fax2mail scripts). There is an example implementation included for Asterisk 1.4, if someone wants to porting it to the new fax backend or more recent asterisk versions and needs some help, let us know. The full press release can be found here: http://www.zoiper.com/downloads/opensource_fax_stack_PR.pdf the project homepage can be found at www.zoiper.com/foip/ Cheers, Zoa -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
