Re: [asterisk-users] Custom Pickup and Transfer dial string
If your working with Virtual PBX then why not set your users with there own rules and normal extension numbers in there own context. You can have many context. That way only extensions you allow to see the context there in will have those options. - Original Message - From: Marcello Lupo [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 22, 2008 6:11 AM Subject: [asterisk-users] Custom Pickup and Transfer dial string Hi to all, i already searched the archive without finding a solution to my problem. I have asterisk installation 1.2.18 to support multiple virtiual PBXs. I use SIP peer in the format ID-EXT to let every virtual PBX to share the same numbers of EXT. Ex. (PBX ID 10 Extensions) 10-101 10-102 10-103 (PBX ID 20 Extensions) 20-101 20-102 20-103 I use some rules in the dialplan to let users of this pbxs to dial only EXT to call other extensions of their PBX. So i add the pbx ID statically to the called number if it is an extension. Now i have to realize the pickup call and blind and attended call transfer. As i read in the documentation the standard codes *2 for transfer give me the dial tone to dial the extension to transfer at but i need to transfer the call not to the EXT but ID-EXT . Have you any idea on how to realize this for Pickup and call transfer? The end users don't have any idea of the ID part of their extension so i have to let them dial something like *2101 and transfer the call to 10-101 . For the pickup i need them to dial something like *8101 and answer the call ringing for 10-101. I already thought a solution for the Pickup to use the application Pickup joined to a custom pickup code and a custom string in the command it self, but i dont have a solution for the call transfer. Thanks in advance for help. Bye, Marcello ___ -- Bandwidth and Colocation Provided 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 errors
Justin Johnson wrote: Hi All, I have centOS 4.3 installed and have attempted to install asterisk separately. I have installed all the modules as suggested on Asterisk downloads, more (via SVN) However, on the zaptel install I am getting the following errors. centosbug is, like, a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h make[3]: *** [/usr/src/zaptel/torisa.o] Error 1 make[2]: *** [_module_/usr/src/zaptel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.9-34.0.1.EL-i686' make[1]: *** [linux26] Error 2 make[1]: Leaving directory `/usr/src/zaptel' make: *** [all] Error 2 Any one have any ideas how I can solve this? Thanks in advance, Justin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone Tollfree Port
John Kington wrote: I tried to get an update from NuFone but Has anyone gotten their tollfree number ported to another provider by NuFone? Should I just forget it and move on? Regards, John Yes we have ported our number out of there service. You need to go and sign some papers with the other provider you want and they take care of the rest for you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreePBX in production?
Rich Adamson wrote: Has anyone attempted to use FreePBX for a business in production mode? Yes it works great in business applications. Initial take is there are lots of things scripted but a lot of limitations in terms of supporting basic business functions. Inability (or lack of flexibility) is handling multiple incoming pstn lines, dialplan limitations, poor/no documentation, etc, to mention a few. Yes it does have limitations, which you can get by with some use of there custom.conf files. Documentation for asterisk and freepbx is done via the users and there is not much on it. But it's getting better. You can see lot of info for it on http://aussievoip.com.au/wiki/ . Also there is a new update that you can route on the Zap channel number now. Maybe its just me, but it appears its no where near usable even with the latest beta1 code. If your able to work with asterisk without a GUI it's better due to you can do more. But remember Freepbx has asterisk as it's main part and it works just the same. It's easyer for many to use it but again this comes with some short commings. But all around for the price is the best GUI out there. Is it just me or what? Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Welltech USA? and Wellgate Products?
I normally don't like talking bad about products. But I would like to say that the Welltech/Wellgate are not products that are support to work with asterisk. I have invested many hours of work in getting there device to work with Asterisk. They do not. And also as of Last Nov. They told me that they did not plan on supporting Asterisk. Good luck if you are able to get them to work since they go and sell there product with other names please post the settings you get for them to work. I have 2 of them as paper holders. And since there really bad I will not even sell them on ebay. kevin ling wrote: Hi, Have you try to search on eBay? I found some welltech devices for sale. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Tuesday, February 07, 2006 3:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Welltech USA? and Wellgate Products? Any feedback on this brand and in particular on doing business with WelltechUSA? I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I am hoping to replace the near worthless Grandstream HT-488. This company is telling me that I need to wire $ directly into there bank account. Most unusual. Thanks for any feedback on this, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXTEL??
Iaxtel has been down for some time now. But to get in contact with digium via your asterisk box all you need is to set this dialing rule up. exten = 500,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ;Call Digium exten = 500,2,Congestion Kerry Garrison wrote: Is IAXTEL still around? I needed to call Digium and figured I would set it up to save some miinutes when talking to them but I can't get it to register. -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which Asterisk GUI?
Ken D'Ambrosio wrote: There are a bazillion GUIs out there (as http://www.voip-info.org/wiki-Asterisk+GUI will attest). However, I'm not sure which to use. A lot seem to be fairly comprehensive... but until I kick the tires, it's trial-and-error. And that would be a *lot* of trial-and-error. So, here's what I'm looking for: - GPL (not a dealbreaker, but I like being able to tweak things if they don't work the way I want) - Comprehensive (does the substantial majority of configuration) - Decent documentation - Wishlist: comes with CLI tools for easy automation Other then writing your own the best one I have found so far is AMP. And belive me you can do allot with it. There are lots of ways to do things in AMP with it's custom config files. And it's GPL and you can write your own changes and even add them to the project. You should look at how there working on version 2.0 of AMP it's going to be a major change. I've used AMP, and found it to be reasonably decent, but there are a lot of things it doesn't do, too. So: which GUI do -you- like? Thanks! -Ken D'Ambrosio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Creating conf files from db
Yes that is how AMP works. It's a very nice setup. - Original Message - From: Douglas Garstang To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, December 22, 2005 5:07 PM Subject: [Asterisk-Users] Creating conf files from db Just wondering if anyone here has tried the approach, where all config files are stored in a database, maybe using the ast_static table structure. Rather than using realtime to access the database live, you have scripts that read the contents of the db, and generate the .conf files from that., and then do a 'reload'. Anyone tried that? How'd it work for you? Doug. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 help please.
OK we need some help in setting up a good wiki-info page for setting up the Mediatrix 1204 to work with asterisk. If anyone has set these unit's up and have them working please post your settings here so we can create a page on the wiki. These unit's are being sold to be used via sip format with asterisk and there is no real information on getting them working. At present there one of the worst I have run into to get correctly working. These are very expensive and some of us can't afford to send them back for a restocking fee. If someone working with Mediatrix has a white paper on getting these unit's working please let us know the link for it. It would be very helpful for many asterisk users. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk using tdm400p has echo
[EMAIL PROTECTED] has the Zaptel from head. Just need to update the zaptel drivers from CVS head you don't have to upgrade the asterisk. Matthew Fredrickson wrote: On Oct 27, 2005, at 12:02 AM, Mark Quitoriano wrote: i tried doing the instruction from voip-info[1] anyway here's my comment with that instruction. when i tried doing /usr/src/zaptel/fxotune -i 4 it gives me this Tuning module 1Failure! Tuning module 2Failure! Tuning module 3Failure! Tuning module 4Failure! how can i debug this? i look at my /var/log/messages and it gives me many of this line Oct 26 17:36:06 sloan kernel: — Set echo registers successfully Oct 26 17:36:25 sloan kernel: — Setting echo registers: I'm using AAH 1.5 stock no modifications. I think they're using versions 1.0.9 for all(e.g. asterisk, zaptel). Don't use 1.0.9. It's old. Try the version of fxotune from head. Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple calls per phone
Asterisk wrote: Hello, I am new to this list and to Asterisk. I am using Asterisk @Home, but have begun to be comfortable editing the scripts. I have a Grandstream GXP-2000 with 4 line buttons. Is there any way I can set Asterisk to send more than one call to the phone without setting up multiple accounts? *70 will turn on Callwaiting via the Amp setup that comes with [EMAIL PROTECTED] Such as: Receptionist answers line Another call comes in and she puts the first one on hold and answers the second one. Currently if she is busy, the call is sent to the queue instead of giving her the option to answer. Thanks! Greg ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bulk Buys/Group Buys
Nathan Pralle wrote: Hey folks, Anyone know of companies selling bulk SIP adaptors (phones, adaptors, etc.) or has the list ever considered doing something like a bulk buy? Give a call to VoipSupply.com 800-398-VOIP (8647) I was just curious...I'm looking to get another 5-6 Grandstreams or similar and I figured I'd ask the list. If we found something that lots of people wanted, it probably couldn't hurt to contact a company and ask for bulk deals. Whadya think? Anyone tried this before? Nathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to send fax using BroadVoice
They do not support faxing over there network. IP faxing has always been a problem. ulaw is the only codec you can use to do this with. And for that matter of fact it's hit or miss. Ariel - Original Message - From: Rene Nelson To: asterisk-users@lists.digium.com Sent: Thursday, September 29, 2005 2:31 PM Subject: [Asterisk-Users] Unable to send fax using BroadVoice Has anyone had success sending faxes via a broadvoice byod account? Everything 'looks' to go as expected, but then my fax hangs up and I get a printout with Error 351. I am wondering if it is a codec issue or something.Any help will be great.Neri ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pinging ...
Alan Bunch wrote: Ok, if I missed something in the wiki please point me there with the correct search terms. Asterisk 1.0.7 (AAH really) 4 co lines from Bellsouth into a Diguim T400P. Polycom 501 x 4 on the desktops. My problem is on calls to or from the CO I hear a beeping every 12 seconds. Sounds like your main line has the Alarm monitor on it. Have you check that out? You can set your watch to it. Could this be a call recording in progress tone. I have not made any effort to turn on call recording. If it is I'll go figure out how to turn it off. Could this be echo cancelation retraining Any guesses from anyone here. I just want the beeping to stop. Well I don't really care but my users sure do 8-) One more thing, this machine is remotely managed and is in a distant city. If I really have to go I can. Thank in advance Alan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Most desireable Linux distribution for Aster isk?
Colin Anderson wrote: There is no real answer to your question. just use one you're most familiar with. I use RH allot so I am now using CentOS mostly. It's Red Hat EL GPL code and so far everything I have runs on it without issues. Great OS. Second that, using FC2 for me, and it's the Devil You Know, right? I have a good handle on the RedHat Way which the Debian guys say is stupid, but it's not, it's just different. All of the little RedHat-y things you get used to and when you use a different distro, it's like, WTF?? I like YUM and RPM's, so RedHat it is, for me. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP 1.10.009 released!
canuck15 wrote: That is great news Ryan. I don't know what I would do without AMP. One question. How can I upgrade from 1.10.008. I downloaded it and followed the procedure in the UPGRADE file but it hung at: Fri, 9 Sep 2005 16:28:55 -0700 - Unable to connect to manager 127.0.0.1:5038 (111): Connection refused [FATAL] Cannot connect to Asterisk Manager with admin/amp111 You need to have asterisk running to do the upgrade -Original Message- From: Ryan Courtnage [mailto:[EMAIL PROTECTED] Sent: Friday, September 09, 2005 1:42 PM To: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Subject: [Asterisk-Users] AMP 1.10.009 released! Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP: https://sourceforge.net/mail/?group_id=121515 AMP 1.10.009 changes: - Optional separation of Devices and Users. Devices are endpoints (ie: phones), and can be Fixed (to a user), or Adhoc. Users are extensions, with options like voicemail. A user can log in to one or more Adhoc devices by dialing *11, and log-off by dialing *12. - Custom device technology support - this means devices that are not configured directly in AMP's admin can still be used (ie: SCCP) - Asterisk Recording Interface (ARI). ARI is a php interface to Voicemail and Monitor recordings. (written by littlejohnconsulting.com) - RingGroups now use strategies: Ring All (default), Hunt, Memory Hunt - DID Routes re-written as Inbound Routing. This allows for DID specific fax emails and call answering options. - Queues can now play a welcome message to callers upon joining. - HINT priorities for FIXED devices - Interface translated to French, German, Italian, Spanish - FOP .21 - FOP button layout can now be sorted by last name or extension number Regards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04b and echo
Matt Fredrickson wrote: On Wed, Aug 31, 2005 at 08:45:34PM -0500, chris gamble wrote: the echo isnt horrible most of the time, and seems extremely random in that i can call a number once without echo, then dial the same number a second time and get echo. things i am currently considering (and would like to know if these might be useful) 1 upgrade from 1.09 ( asterisk at home ) to 1.2 cvs code base PS you dont' need to upgrade asterisk to CVS Head to use the Zaptel from CVS head and the new Echo setup. I just installed it on 3 systems and they all improved. Using the KB1 That is worth a shot. There are a few new echo-related features that have been added: 1.) fxotune - try this first. There is a file called README.fxotune that explains how to use it. It is primarily for doing echo related line tuning (which in your case possibly won't help). 2.) Also, there is a new echo canceller in CVS-HEAD zaptel that has received a lot of positive feedback. Look in zconfig.h for ECHO_CAN_KB1 for further information. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registrar only setup
have you tried in the sip.conf for the devices canreinvite=yes - Original Message - From: Tomas Florian To: asterisk-users@lists.digium.com Sent: Tuesday, August 30, 2005 8:48 PM Subject: [Asterisk-Users] Registrar only setup Hello, Im having trouble figuring out how to setup Asterisk so that its only a registrar not passing any RTP data during phone calls. So far I got this far: Asterisk server holds registration information for phones Phones register with Asterisk giving it their ip+port where they can be currently contacted NAT doesnt seem to be a problem because STUN seems to take care of it nicely for me. The hard part that I dont understand is this: Phones can call each other BUT all the RTP traffic is passed through Asterisk I dont want this, I need that the phones call each other directly based on the registration info stored in Asterisk. Im having hard time wrapping my head around this I think Im missing some key part but the way I understand Asterisk is that it listens for requests on the SIP channel, when it gets a request it handles it appropriately using its dial plan. But in the dial plan the only thing that makes sense to use is dial and once I do that all the RTP is sent through asterisk (in-out) to the other phone right? Or maybe the problem is on the phone setup? I tried to make sure that Im not specifying any outbound proxy but I do have to specify proxy otherwise it will not know where to register right? Or maybe Im all messed up 8-P I thought I understood asterisk at least a *bit* until I came across this :-) Thanks for any clarification, Tomas ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] looking for failover ideas
Sergio Serrano wrote: If I use hearbeat I need a failover system for ISDN Lines, not? I waould like that if Server A crashes, Server B Control SIP Registration and ISDN Lines. Do you know about this? There is a new product form Redfone that will help provide a failover with the T1/PRI connections. There available from VoipSupply.com http://www.voipsupply.com/product_info.php?products_id=1002 regards, srsergio -Mensaje original- De: Senad J [mailto:[EMAIL PROTECTED] Enviado el: martes, 23 de agosto de 2005 22:10 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] looking for failover ideas [EMAIL PROTECTED] wrote: How do you do monitoritng? How Server B knows that Servar A is down? I just do a rsync and MySQL Replication, but I try to do a C program that monitor Server. If you know how can I do this monitoring I will be pleasant with you. 1. use heartbeat for failover between A and B. Setup correctly failover is fully automatic. 2. u can use www.nagios.org or similar installed on C to monitor A and B Senad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office setup/using analog lines w/ Asterisk
jennyw wrote: Dave Cotton wrote: This was your first experience with *, was it also the installers? Only sharing with the next busiest card in the machine the one feeding the IP phones. Yeah, I know, in retrospect it sounds really odd that we did that, but at the time he thought there was a chance it would work. I'm not sure why -- he's done several installs before. It might be because we ended up taking more time than intially planned to do the the install (Murphy's Law and all that). The hardware we have, by the way, is: Well it's funny due to I think what you have is over kill. I have run into problems with raid and smp systems. Depending on how many users you have and what you planned on for growth. Intel does make some nice cards but I would not change that out yet as it will be a better solution to see if you can fix the irq's settings. Adding an addtional card that takes an IRQ will only add the to problem. AMD Sempron 2400+ 3Ware Escalade 8006-2LP (2 channel SATA RAID) Asus A7N8X-E Deluxe 1 GB RAM Jetway Radeon 9000 64MB 128Bit 4X AGP DVI/TV Out Dual Head We're using the on-board NICs. From what people have said, that could be a problem (in addition to the IRQ sharing issue), so I'm off buying Intel NICs now (I don't think the store has eepro 100s -- I think all the Intel cards they have are gigabit now, but that shouldn't make much difference should it?). I'm not sure if I can change the IRQ setting on the Radeon, but will try. Can the Digium cards change IRQs? Thanks! Jen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office setup/using analog lines w/ Asterisk
Jennyw, I have setup about 8 Asterisk systems with The TDM400p boards in them. Yes allot of them had at the beginning some echo and other things. But I have been able to work and get them fixed. 1) Make sure your motherboard is able to assign it's own IRQ for the board. This is one of the most important things. 2) There is a tool on the system that lets you set the txgain and rxgain. It's called ztmonitor which you can use to see how the volume is setup then you can make some adjustments. 3) It's important to make sure your system has good network card. I have had some problems with Realtech with echo. 4) You also need to make sure you have the phones on connect to a switch. 5) Asterisk system does save allot of money in the long run. I am sorry that the person you got was not able to help. But I know that the TDM400p boards have there well critic's but they do work. 6) For us to give you more help we are going to need to know more about you system. What is the server your using? What phones? How is your network setup? If you want you can email me directly. I will try to help you out with your setup. Ariel Batista jennyw wrote: Hi, We recently tried installing Asterisk for a small office. We figured the safest way to go would be to buy from someone who sold equipment specifically for Asterisk and to use a consultant that they recommended. However ... it didn't turn out so great. Sound quality is terrible -- the echo is pretty bad, and there are popping noises, too. Callers say that people on the Asterisk end sound very faint, while people on the Asterisk end hear people maybe too loundly (might be related to the popping noises -- sounds like when you have stereo turned up too high). The reseller and the consultant both say that the most likely cause for this is using Digium cards w/ analog phone lines. Apparently, they say, sound quality can be pretty bad. I called Digium and they gave me some suggestions for settings, but nothing has worked well. So I wanted to ask others ... has anyone had good luck with using analog phone lines and Asterisk? Especially with Digium cards (we use the TDM400P)? Although from reading articles on the net it sounds like people do have a lot of echo problems, it also sounds like some people are using analog phone lines with some success. FYI, what I've mainly done is try changing echotraining, echocancel, echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard from the reseller that what might work better is to trade the Digium cards in for VegaStream gateway. It's more expensive, but apparently has a DSP built in that should increase voice quality. Of course, they say there are no guarantees with this. They also mentioned (after the fact) that Asterisk systems don't necessarily save money. So far, the experience has been very frustrating and I'd love to hear some success stories from others (or more info on what I can realistically expect from an Asterisk system)! And, of course, some ideas on how I can get things to work better. One of the next tests will be using Asterisk with a VoIP provider to see what the sound quality is like with digital on both ends. PRI sounds like it'd be even better, but for an office w/ 5 people, it sounds pretty expensive. How do other people do this? Thanks in advance for any pointers! Jen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need Help Troubleshooting Broadvoice Connection
Tim P wrote: [EMAIL PROTECTED]:mypass:[EMAIL PROTECTED]/2068660133 You need to add the number to the back so you can route it with asterisk. Ok I can register with BV fine (as far as I can tell from asterisk - see below). I am able to make outgoing calls but all incoming calls get a fast busy. I have opened and forwarded the following ports to my pbx: 5060-5063 UDP + TCP 69 UDP (BV claims they need this) 1-2 UDP I tried switching proxies as well, tried both LAX and CHI with the same problem. Called BV they said they can conenct andd call it with a softphone so it must be a configuration issue. Here are some outputs that might be helpful: Asterisk -r sip show registry asterisk1*CLI HostUsername Refresh State sip.broadvoice.com:5060 [EMAIL PROTECTED]23 Registered asterisk1*CLI sip show peers asterisk1*CLI Name/usernameHostDyn Nat ACL Mask Port Status bv/2068660133147.135.12.128 N 255.255.255.255 5060 Unmonitored /(Unspecified)D 255.255.255.255 0 Unmonitored 1005/1005(Unspecified)D 255.255.255.255 0 Unmonitored 1004/1004(Unspecified)D 255.255.255.255 0 Unmonitored 1003/1003(Unspecified)D 255.255.255.255 0 Unmonitored 1002/1002(Unspecified)D 255.255.255.255 0 Unmonitored [Kasterisk1*CLI sip show peer bv asterisk1*CLI * Name : bv Secret : Set MD5Secret: Not set Context : from-pstn Language : FromUser : 2068660133 FromDomain : sip.broadvoice.com Callgroup: (0) Pickupgroup : (0) Mailbox : LastMsgsSent : -1 Dynamic : No Expire : -1 seconds Expiry : 900 Insecure : Very Nat : Always ACL : No CanReinvite : No PromiscRedir : No DTMFmode : inband LastMsg : 0 ToHost : sip.broadvoice.com Addr-IP : 147.135.12.128 Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Username : 2068660133 Codecs : 0xc (ulaw|alaw) Codec Order : (ulaw|alaw) Status : UNKNOWN Useragent: Full Contact : (not sure about that Status = UNKNOWN, is that a problem?) Get full output on outgoing calls and they connect sucessfully Get zero output on incoming calls, pbx never seem to get them Here is my sip.conf [EMAIL PROTECTED]:mypass:[EMAIL PROTECTED] [sip.broadvoice.com] username=2068660133 user=2068660133 type=user secret=mypass nat=yes insecure=very host=sip.broadvoice.com fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband context=from-pstn authname=2068660133 Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Phones
Chris Gamble wrote: Just got in a bunch of polycom phones for use on my shiny new asterisk box, but found 2 small issues I was wandering if someone could help me with. Are you using AMP or Asterisk @ Home? First, though the phones support 2 call appearances, if I am on a call, the second call does not ring through -- it goes to voicemail instead of letting me put the first on hold to talk to the second. Is there a way to fix this? If you are then you need to turn call waiting on * 70 The second is: a lot of my phones will not ring for internal extensions. They show up on the screen as a call ringing in, but the phone itself wont ring. About 50% however do ring. What could cause this? Are the phone registered correctly? What are the settings you have on them. As usual, thank you all for your kind support in getting this far! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer to outside line.
This is simple since your using AMP,you can create a ring group to dial that number out for you. First create your ring group lets put number 200 for it (you can call it any number you want). where the extension number goes just put there the phone number you want like 301212# don't for get the # key after the number. Then if it does not pickup you can send it to a voicemail box or any other place you want it. Ariel - Original Message - From: Tim King To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, August 03, 2005 10:12 AM Subject: [Asterisk-Users] Transfer to outside line. Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the call to a offsite phone number utilizing my Zap Trunk. Im sure someone has done this already. Anyone want to point me in the right direction? Tim King ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer to outside line.
I would need to see what the CLI displays when this happens. I am using at many locations and it works. How many dialing rules for trunks have you setup. Do you have any passwords setup? We have a user to user support area for amp on the freenode #amportal You can find me there as well. Ariel - Original Message - From: Tim King To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, August 03, 2005 5:24 PM Subject: RE: [Asterisk-Users] Transfer to outside line. I tried this solution, although ti acts like it is working it only rings once and than the call is just dead air. The number I am forwarding to never rings. Anything else I may need to try? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ariel BatistaSent: Wednesday, August 03, 2005 3:40 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Transfer to outside line. This is simple since your using AMP,you can create a ring group to dial that number out for you. First create your ring group lets put number 200 for it (you can call it any number you want). where the extension number goes just put there the phone number you want like 301212# don't for get the # key after the number. Then if it does not pickup you can send it to a voicemail box or any other place you want it. Ariel - Original Message - From: Tim King To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, August 03, 2005 10:12 AM Subject: [Asterisk-Users] Transfer to outside line. Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the call to a offsite phone number utilizing my Zap Trunk. Im sure someone has done this already. Anyone want to point me in the right direction? Tim King ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to create a secret code to use [EMAIL PROTECTED] server's long distance plan from a public phone
Adrien Laurent wrote: Hello everyone, I have an IAX server ([EMAIL PROTECTED]) with a FXO card. I have a trunk connected to a voip provide, asteriskout. When I call my server from a public phone, I want to route this call to the asteriskOUT trunk so that I can make long distance calls. Your going to have to add an extenion in the extensions_custom.conf for a DISA setup. Depending on what your extensions look like it's fairly easy to do. Your system should be answered by the Digital Reception for this to work correctly or a dedicated DID routed to it. This is a quick sample. lets give an extension like 300. exten = 300,1,Goto(custom-disa,s,1) [custom-disa] exten = s,1,Authenticate(1234) ; any number you want for a password here. exten = s,2,DISA,no-password|from-internal exten = s,3,Hangup How can I setup a secret password in the extension.conf, so that my asterisk server can allow me to make long distance call ? Thanks a lot, Adrien ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Auto-Answer problems
C F wrote: The IP-501 AFAIK comes shipped with SIP1.5.2 which does NOT use ipmid.cfg. You have to get new *.cfg files for the ip-501 or the older phones that run 1.5.2. Sorry to tell you but that is not a correct. The IP-501 I have about 15 of them new and they came with 1.4.2 also they do use the ipmid.cfg. But it can use the newer version which is an all in one. You tell the phone which files to use via it's configuration file mac.cfg It sets up which files to load. On 7/14/05, Chad Osmond [EMAIL PROTECTED] wrote: CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans) exten = 301,2,SetVar(ALERT_INFO=Ring_Ans) # Tried both combinations exten = 301,3,Dial(SIP/5001,15) exten = 301,4,Hangup Sip.cfg for Polycom phone alertInfo voIpProt.SIP.alertInfo.2.value=Ring_Ans voIpProt.SIP.alertInfo.1.class=4/ Ipmid.cfg RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ Asterisk Log: -- Executing SetVar(SIP/5002-6e20, _ALERT_INFO=Ring_Ans) in new stack -- Executing SetVar(SIP/5002-6e20, ALERT_INFO=Ring_Ans) in new stack -- Executing Dial(SIP/5002-6e20, SIP/5001|15) in new stack -- Called 5001 -- SIP/5001-f735 is ringing -- Nobody picked up in 15000 ms As you can see it just rings, and then hangs up. Any one have an idea? Chad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Gui?
Michael Felder wrote: Is [EMAIL PROTECTED] as functional as full blow Asterisk. I am using this for my business. Yes I feel that the actually name should be [EMAIL PROTECTED] or SOHO. I have a few clients running there business on it. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ariel Batista Sent: Friday, 15 July 2005 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Gui? Michael Felder wrote: Can anybody recommend an Asterisk GUI to help a newbie confg ? Try [EMAIL PROTECTED] it's a complete ISO with everything you need to start with. If you already have the OS installed then check out AMP which is by the way included in [EMAIL PROTECTED] Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: [EMAIL PROTECTED] http://www.ITMedic.com.au Keeping your computer systems healthy. ure Asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Gui?
Michael Felder wrote: Can anybody recommend an Asterisk GUI to help a newbie confg ? Try [EMAIL PROTECTED] it's a complete ISO with everything you need to start with. If you already have the OS installed then check out AMP which is by the way included in [EMAIL PROTECTED] Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: [EMAIL PROTECTED] http://www.ITMedic.com.au Keeping your computer systems healthy. ure Asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Colocation/Telehousing
Sahil Gupta wrote: Hi, Is there anybody on the list that recommends anyone for colocation/telehousing in the US? I'm after 2 Servers to be hosted in the US, preferably on the west coast. I would suggest www.race.com Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux Distribution for Asterisk server use
Wow, I just want you to know I am and have been a Networks Engineer for many years. I started back when Novell was king for networks. Window and many others have come by and I have setup shop with them. I still manager and maintain several of my Clients Windows networks. Almost 3 years agoour boss/owner decided to give asterisk a try. We got a Consultant and they set the system up for us. I at the time did not know Linux distro's and was under the impression they were all the same. We I quickly learn there not. (Yes Linux kernel is the main part.) But the distro's are very different. We were setup on Mandrake.I learn as much as I could about asterisk and for the 1st 6 months never did anything with the OS. Well soon I needed to do some changes and add something's like reporting and other options that required me to start learning the OS and what It could do. I looked at about 10 major distro's, Red Hat 9 was current back then, Mandrake, Debian, Gentoo, Slackware. There all good I spent about 3 months going through them to fine the one that would fit me best. I decided due to all the books out to learn and setup Red Hat 9. Which by the way asterisk works great on. I setup all our 3 different servers on RH 9. Very soon RH changed there Linux distro's to the Enterprise editions and create in there Open Source Fedora. I tried Fedora Core1 and Core 2 until about 1 year ago. And at that time I found CentOS. It's Red Hat Enterprise without there labels. And Now I am using for all my setups CentOS. For Production systems I am using there CentOS 3.4/3.5. Works great and is easy to setup. For my testing and other servers that run on Linux I use CentOS 4.1. I have started to move some of my customers off Windows. It's a great stable OS which is mission critical product. Hope this helps if you need more info please feel free to email me off list. Ariel - Original Message - From: TWV To: asterisk-users@lists.digium.com Sent: Sunday, July 03, 2005 2:45 AM Subject: [Asterisk-Users] Linux Distribution for Asterisk server use Hello, My question is about which Linux distribution to choose for Asterisk. (/me holds breath) OK, hopefully youre still reading, because whatever you were thinking now, youre thinking wrong! ;) First of all, I want to make clear that I have read EVERY message and reply that I could possibly find about this topic, so that includes the dozens of messages here on the Asterisk mailinglists, on the Digum forum, and even Google search results! Still, my question was not answered! Mainly because the same answer always came back: Use the one you are most comfortable with. Well, I already knew that (linux is linux), but it doesnt apply to my situation at all! Lets make things clear and concrete now: In my professional life, I work as a windows system network administrator and as a developer on the NET platform, and have a long and extensive experience with telecom and VoIP. Working for a telecom company, I now have accepted the challenge to extend our offering with Asterisk. During the past 6 months, I have learned as much as I possibly could about the Asterisk PBX, successfully set up a complete test environment, developed IVR systems, and now we feel the time is right to put these services into production. BUT, before we can go live, one important problem remains: as I said, I am a Windows guy, I have a VERY profound knowledge of Windows and manage almost 20 Windows 2003 Servers that run mission-critical applications on a 365/24/7 basis, and support a large number of Windows applications and Web services (some of the applications I have developed are used by more than 25000 users every day!) Why am I telling this? Well, because I want to make it clear that I am perfectly happy with my platform/OS (windows), and have no intent whatsoever to ever change servers or application platforms to linux (let alone my workstation). Whats more, I have NEVER come in to contact with linux/unix before, so I have never worked with ANY distribution. Having explained all this, it should sound logical that I chose the AsteriskWin32 version for learning Asterisk. Of course, I realize that we cant put any production system on AsteriskWin32 So before we can go live with Asterisk servers and services, this last issue remains to be resolved: what Linux distribution should I choose (and learn)? As I explained, I see Linux merely as a necessary evil (because of my lack of knowledge) for running Asterisk. So Im asking about the best linux distribution only to put up asterisk servers. Im NOT asking for the easiest one or so (I always enjoy challenges and learning new things), Im asking for the best choice to build a carrier grade telecom system, having to support thousands of users each and every day.
Re: [Asterisk-Users] HT-488 vs. SPA-3000?
Brian Roy wrote: On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote: In other words, the further the spa3000 (or TDM card) is from the central office, the more difficult it seems to be to set gain values that are acceptable. That's apparently why many people find its use is okay while others seem to think its objectionable. I've read all of these reports and thought... Wow, I must be lucky. The audio on mine is perfect. My wife uses it ALL day and has never complained about the voice quality. She is very picky too.. Just for comparison's sake, here is the info on mine I have 2 of them and the work great no sound problem. And yes my wife also uses it all day long. Product Information Product Name:SPA-3000 Serial Number: 88012DA02506 Software Version:2.0.11(GWg) Hardware Version: 2.0.1(96a3) MAC Address: 000E08CAF559 Client Certificate: Installed Now, it very well could be distance to CO, but I doubt that I am that close. I live out in the woods. Sorry you guys have all these problems, but mine is perfect outside of the occasional talk off. -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MCI vs. XO/Allegiance
Title: MCI vs. XO/Allegiance we have been using XO/Allegiance for over 3 years and have had no problems. I can't compare to MCI but we also had a sprint t1 that we had to get remove due to them being bad in billing and also not very reliable for faxing. - Original Message - From: Wiley Siler To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, June 13, 2005 6:59 PM Subject: [Asterisk-Users] MCI vs. XO/Allegiance Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best BootRom SIP Code for Poly600?
Justin Ellison wrote: Hey all, Just getting started playing around with my Polycom 600. According to the wiki, it looks like it's recommended to run BootRom 2.6.1 and SIP 1.4.1. Is that info still current, or is it safe to upgrade to 3.0.1 and 1.5.2? I am still running BootRom 2.6.1 with Firmware 1.5.2 works great. I don't want to upgrade the rom due to not being able to down grade. Justin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B
[EMAIL PROTECTED] wrote: Hi, I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. This board is FXO which you plug incoming phone lines into it. So plugging in a handset unless it's a butt set it will not give you any dial tone. In fact you damage the port doing this to it. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP-500 600 Nat settings.
I have looked at the wiki and the mailing list. But I need to find how do we setup the external IP address and the rtp ports for the Polycom IP-500 and IP-600. There web interface has a nat setting but can't find instructions on how to set this up. I would like to set this up via there ftp file setup instead of via there web setting. Also There QoS settings are set to 5 and 2 but there it does not say if you change it to 7 or to a lower number which one gives you better priority. Main problem I am having is that the polycoms work great as long as there on the same LAN. once they go through a Nat router even if all the ports are open we get one way audio or no audio. The asterisk servers are on a real world IP address and the Phones are behind a Nat firewall called m0n0wall. We have all ports open going out to where the asterisk box is setup. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@home
Asterisk @ Home This CD will install everything you need to get your Software PBX going. Its a complete ISO CD that brings together the OS (CentOS 3.4) Asterisk Software version 1.0.7 stable AMP Asterisk Management Portal Web GUI FTP TFTP Plus many more items. Every pre-configured to install and run out of the box. Just put it into your CD drive it will format and setup asterisk for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Quintin Sent: Tuesday, May 24, 2005 6:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] [EMAIL PROTECTED] Hi Can any one tel me what is [EMAIL PROTECTED] Thx Q ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog Lines
Sean, We setup a support department via just that way. In fact it's about the only real way to get modems working correctly. We used T100p card attached to Adtran 750 units. We got them on ebay for around $ 500.00 each. Which is well worth the cost. Good luck. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cook Sent: Tuesday, May 24, 2005 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Analog Lines I am looking for a cost effective way to drop analog lines from our asterisk system to support modems and faxes. More than would typically be done with TDMxxB cards. I have looked at going with a T1 interface to Channel Bank, but that just seems like a very expensive way to solve this problem. ($1500 - $2000 ). Any suggestions? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell PowerEdge SC420 for Office Implementation???
They work just fine. I have a few of them out at customers and there working without any issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Geoff Manning Sent: Friday, May 20, 2005 3:01 PM To: Asterisk Users (E-mail) Subject: [Asterisk-Users] Dell PowerEdge SC420 for Office Implementation??? I was wondering how the Dell SC420 will perform under normal office to office communications. We would equip each server with a T1 card to make office to office SIP calls. They will integrate into our existing PBX systems. Does anyone on this list use this hardware currently Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell PowerEdge SC420 for Office Implementation???
Well, if you are only making office to office calls, save the $500 per T1 card and just use NICs. The T1 card is only required if you are using a voice T1. If you are doing IAX to IAX for example between offices, then Asterisk is your friend. Avoid SIP altogether as it is not needed and just use IAX bwteen the boxes. Integration to legacy PBX can be found on the Wiki www.voip-info.org Wouldn't we need the TE110P card to interface with the T1 card of our legacy PBX. That will give us 24 channels to work with. Am I on the wrong track here? Yes you do need the TE110p to connect the T1 part. I just wanted to add a few things about the SC420 I have been using. First I am had to use the 2 lower PCI channels the ones away from the AGP card. This way they don't share an IRQ with the Nic card. The TE110p works great but again you need to use the 2 lower PCI slots so that your assigned it's own IRQ. Other then that it works great. Love the server. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Who knows where voicepulse has their asteriskservers?
I know that you can contact www.race.com they have rolls of rack space available in One Wilshire in LA. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of InternetMarketingMan2001 Sent: Friday, May 20, 2005 6:25 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Who knows where voicepulse has their asteriskservers? I want to collocate an * box somewhere, where better than where voicepulse chose to put their servers? They probably did their homework and selected someplace where good handoff to the pstn can be found, right/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Phone Recommendations?
Just want to let everyone know that even if there changing it out to the new 501 it's still on of the best. Remember that people are still buying the Cisco 7960G which is being phased out as well. The IP-500 works and works very well. I know that there price will be going down soon once there are some supplies of the IP-501. But if you need a phone now it is a very good one for the price. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Thursday, May 19, 2005 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Phone Recommendations? On Wed, 18 May 2005 22:29:40 -0500, Kristian Kielhofner wrote: Ariel, It's probably not a good idea to reccomend the IP 500/300 anymore. They are being phased out by Polycom because they (and the IP 300) only have 2mb of flash, and Polycom is looking to standardize on 4mb for their firmware (which the IP 600 has had since day one). If you are going to buy a Polycom now, get an IP 600, or, wait for the 301's or 501's. Don't say I didn't warn you! Good advice!. BTW, I LOVE my IP600's. I also kinda like the Zultys 4x4/4x5.The hardware and software is good but their support arrangement is terrible. They provide no end user support at all. Period. They rely upon their dealers to provide all support, but then they're ok with signing up dealers that know nothing about the products. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Two TDM04 with Poweredge
I have 2 of them working on a SC420 server and also another one the SC400 and older one that has 4 TDM boards on it. Both systems have been working fine. I did not have to do anything special on them to get them working. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Hayden Sent: Thursday, May 19, 2005 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Two TDM04 with Poweredge Has anyone on this list succesfully managed to get two (or more) TDM04 (with four FXO each) working on a Dell PowerEdge server? If so, which model? Was it a hassle? I'm doing a seven-line installation and a callbank seems like overkill, I just don't want to get suck with a PowerEdge that gets into an IRQ mess. Thanks in Advance, Tom Hayden ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How do you put someone on hold on a zap channel?
Hold on asterisk is not possible But you can park the call. If you have setup parking extensions in the features.conf the default is 700. You would flash then 700 it will give you an area where it parked like 701. Then when your ready to get back to the call just pickup and dial exten 701 and you got the call back. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Gabrielson Sent: Thursday, May 19, 2005 4:58 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] How do you put someone on hold on a zap channel? Ok, this is probably a stupid question, but I can't seem to find anywhere where it tells how to put someone on hold on a zap channel. Flash gives me a dialtone and # tells me to enter a new extension, how can i just put the caller on hold. Pressing # then hanging up drops the call. Is there a simple way of doing this without transfering the user to a parking lot? Thanks, Jon. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Small office setup with Asterisk @home, IAX and analog termination
I would say it would be batter to the the TDM11b since it will have your inbound analog line for 911 and faxes and the FXS port you can plug your fax machine in. This is what I do for most of my SoHo setups. Which I also use AAH for. Ariel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Howell Sent: Wednesday, May 18, 2005 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Small office setup with Asterisk @home,IAX and analog termination I'm setting up a small office with about 8 SIP phones. Incoming and outgoing lines will be through IAX. We would also like to use an analog line for 911. Is the TDM01B a good option for this kind of configuration? Are there gotchas I'm missing? Finally, we would like to be able to use analog fax machines in the office. Would it make more sense to purchase the TDM400 card with 1 FXO and 1 FXS port and use the FXS for the analog faxes, or to split the analog line before it goes into the Asterisk box? Or something else entirely? Thanks! --TWH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nearing my wits end....bad switch???
Yes I have customers using this switch and the 2324 as well. They work fine even with the IP-500 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Wednesday, May 18, 2005 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Nearing my wits endbad switch??? Grasping at straws here...is anyone using a Dell PowerConnect 2224 24-port unmanaged 10/100 switch in a deployment? I have two separate asterisk installations with bad one-way audio where the only common elements left are the Dell switches and Polycom IP-500 phones. Two different ITSPs - one location uses IAX2 to the outside world, the other SIP. Two different bandwidth providers, one SDSL 1.5 Mbps and the other T1. Both locations had Dell servers (different models though), but I've since replaced one of them with a clone with no improvement. In both locations the computers were plugged into the switch port on the phones and I've tried separating them with no luck. The problem occurs with or without a high network load present and can occur with only one person in the office and using the phone... I'm grasping at straws here, but maybe both switches are in some way either faulty or not suitable for the task? Both were purchased in the same timeframe in March 2005... The remote party hears my users fine, but my users get bad audio inbound from the remote party...below are the specifics for each location: Location 1: - SDSL 1.5 Mpbs with static IP, Netopia 4652 SDSL router (enabled Prioritize Delay Sensitive Data to recognize tos=lowdelay per Netopia support) - Dell PowerEdge SC420 with TDM04B (currently only using one port. the single analog line is call forward on busy to my IAX provider) - Asterisk CVS-v1-0-02/22/05(AAH 0.6) using IAX to connect to my provider over the public internet - I have run pings for an extended period of time against my provider's server and get no packet loss. - In IAX.conf: tos=lowdelay, jitterbuffer=yes, also enabled Prioritize Delay Sensitive Data on the Netopia to support tos=lowdelay per Netopia support - Average ping time to my provider: 160 ms with no packet loss - 8 Polycom IP-500's running SIP 1.4.1.0040 and bootrom 2.6.1 using ulaw only - Small Business Server 2003 set up as DC for the network - Two network laser printers -Dell 222424-port unmanaged switch (all phones are home run back to a patch panel, patched from there into a switch port. The DSL modem, printers and server are patched into the switch in the same way) - 8 pc's running XP Pro, all plugged into the switch port on the back of the IP-500's Location 2: - Full rate data T1 - Dell PowerEdge SC1420 (Since replaced with a clone pc)with no TDM hardware at all (this location connects SIP directly to the T1 providers Broadsoft switch and does not go over the public internet) - Asterisk 1.0.7(AAH 0.9) - using SIP to connect with my provider (not across public internet, not natted since the Cisco IAD does the SIP mangling for us) - Average ping time to the broadsoft switch: 42 ms - 8 Polycom IP-500's running SIP 1.4.1.0040 and bootrom 2.6.1 using ulaw only - Small Business Server 2003 set up as DC for the network - One network printer -Dell 222424-port unmanaged switch (all phones are home run back to a patch panel, patched from there into a switch port. The DSL modem, printers and server are patched into the switch in the same way) - 8 pc's running XP Pro, all plugged into the switch port on the back of the IP-500's Can anyone throw me a bone??? I'm willing to try anything, my next steps are to run a new cat5 along the floor from the switch to one of the phones to eliminate the wiring in the walls, and if necessary to purchase a managed switch that I can do QOS on... Thank you Marty ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pickup other ringing phone
You need to go into the extensions setup and put the pickupgroup and callgroup to the same on both. That way when you hear the other extension ring you just dial *8 send and you can pickup the ringing phone call. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Brown Sent: Wednesday, May 18, 2005 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Pickup other ringing phone Hi everyone, Is there a simple way of answering a different ringing extension from a sip phone using AAH? I have absolutely zero technical know-how when it comes to modifying conf files etc. Still working on figuring it all out. ;) That brings me to my second question... where the hell does one find an extensive manual of sorts that explains all conf files and what the strings all mean etc? Cheers All Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Phone Recommendations?
The receptionist phone is going to be a hard one. We use Flash Operator Panel. Works great. Now about the phones for all around great phone we are using the Polycom IP-500 which is in my view one of the top of the line phones. For el cheapo well we are using one that is yes cheap but also pretty good. Sipura 841 is filling this bill for us. As well as for some of the users where we want something a little better then the Sipura we use the Polycom IP-300. Ariel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Mensel Sent: Wednesday, May 18, 2005 10:25 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] SIP Phone Recommendations? Hi all. I'm in the process of putting together a new Asterisk system as a proof-of-concept, and wanted to see which SIP phones all of you had the best luck using with Asterisk. I've just come off a very trying experience with some Cisco 7960s, and am looking for something else to round out the phones on our network. This is a small setup, for no more than 20 users total. We need at least one of them to be a receptionist phone, the sort that calls can be routed from throughout the network, and then several more garden-variety handsets for regular users. A couple of el-cheapos to stick in out-of -the-way, little used spots would also be nice, if there are any good ones out there. Any phones that you've loved? Any that you've hated? Your input will be most welcome. Cheers, John Mensel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Credit Card Machines
I don't have any problems using a pots line with the credit cards. In fact I have in some locations a Sipura that is attached to the cc machine. Works just set it up just like a fax using ulaw only. Ariel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Coulthurst Sent: Tuesday, May 17, 2005 5:12 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk and Credit Card Machines I am planning to deploy an Asterisk server at a local restaurant and was thinking: I hear a lot of troubles using fax machines with IP trunks. What about using Credit Card readers? Same basic technology right? A slow modem to negotiate the transaction. Does anyone have any caveats? Suggestions? Incidentally, the credit cards might be on POTS lines with a Digium TDM22B. Any concerns using this arrangement? Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoipSupply.com
I have gotten Digium boards, Sipura and Polycom phones from them. There very good and I have not had any problems with them. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Manjit Riat Sent: Tuesday, May 17, 2005 8:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VoipSupply.com I am going to buy some IP phones from them but I sent them an email couple of weeks ago and got no reply. Has anyone ordered anything from them? Any other places that I can buy from? Sorry if it's a wrong post. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with extensions - can't dial 700
Check features.conf for parking extensions. There default is 700 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason Sent: Monday, May 16, 2005 8:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Help with extensions - can't dial 700 I have been working on integrating some FXS ports into my dial plan delivered via a channel bank and testing with an analog handset. The receptionist is on Extension 700. All other SIP phones are 7XX. From a SIP phone I can dial 700 and all other extensions. From the analog handset I can dial any other extension but not the 700 number. Weird? Yep. The CLI does not show any dialing when I dial 700. Extensions.conf [office] exten = 700,1,Dial(SIP/700,20) exten = 700,2,VoiceMail,u700 exten = 700,102,VoiceMail,b700 exten = 701,1,Dial(SIP/701,20) exten = 701,2,VoiceMail,u701 exten = 701,102,VoiceMail,b701 exten = 702,1,Dial(SIP/702,20) exten = 702,2,VoiceMail,u702 exten = 702,102,VoiceMail,b702 exten = 703,1,Dial(SIP/703,20) exten = 703,2,VoiceMail,u703 exten = 703,102,VoiceMail,b703 .. [zap-in] exten = s,1,NoOp(Starting Zap-in) include = office == zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes context=zap-in group=1 signalling=fxo_ks faxdetect=incoming channel = 1-12 Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet anyone?
It would be nice if you post how you set this up to either the wiki or right here. Just a few lines would do nicely. There seems to be allot of people who use voipjet and aah and both are good products. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Friday, May 13, 2005 10:48 AM To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voipjet anyone? I use AAH with VoipJet and it works perfectly. Setup was a breeze with absolutely no hand coding of configs required. VoipJet is without a doubt the best outbound provider I have come across. No problems at all yet. knock on wood And the call quality has been awesome. Anyone having trouble getting their AAH setup with voipjet may contact me off list for assistance if they want. Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Friday, May 13, 2005 5:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voipjet anyone? Personally I thought that VOIPJET has the best service and documentation including simple up to date CDRs also. They do not offer incoming, at least not to me If you follow the instructions on their site it will work, if you are useing AAH then maybe you should look into editing the files by had. Checking the server ping time will help. Everyone with a nonroutable IP address will be surprised which ones are faster. On 5/13/05, JD [EMAIL PROTECTED] wrote: Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m, [1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received reject Outbound settings: notransfer=yes auth=md5 context=from-pstn host= 66.246.220.19 secret= md5hashstring type=friend ; also tried peer and user username=1234 Im using [EMAIL PROTECTED], but that shouldnt matter; people have this working or is it me? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 3 x TDM400P in one PC ??
I just setup one Dell SC420 with just one SATA drive and 512mg Ram ($ 404.00) with 2 TDM04B in it that is 8 FXO ports. And a second system for another customer with 3 TDM 2 TDM40B 8 FXS ports and one TDM01B for 4 FXO's. Both systems are working just fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joerg Wleklik Sent: Friday, May 06, 2005 8:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 3 x TDM400P in one PC ?? Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? Ciao Joerg -- _ Don't PANIC ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 10 digit dialing in Ft Lauderdale, FL?
Yes it's used in Ft. Lauderdale. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo Jojo Sent: Wednesday, May 04, 2005 10:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 10 digit dialing in Ft Lauderdale, FL? Does anyone know if 10 digit dialing is used in Ft Lauderdale, FL? -- Private Label Wholesale Internet Access! http://www.YourOwnISP.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?
I just setup a SC420 with two TDMO4b cards in it and it works just fine. No problems what so ever with it so far. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Hencke Sent: Friday, April 22, 2005 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P? I have head that the SC prefixed Dells are not good to use with Digium hardware. Considering how picky my TE405P cards were in other low end Dell servers, I would suggest using an 1850 instead. OTOH, if it does work, please let me know :-) If you go to small biz, you can get the SC1425 trimmed down with dual 2.8hgz for under $1k - Ben On 4/22/05, Greg Boehnlein [EMAIL PROTECTED] wrote: Hello, I've been asked to build a couple of Gateway servers for a client w/ TE405P hardware, and have been looking around at various 1U options. I've been looking at SuperMicro and Tyan barbones boxes as possible platforms, but then was directed to Dell's SC1425 by a friend. Short story, is that you can purchase a 2x3.0Ghz/1GigDDR400/1xSATA box in a 1U form factor for $1,498.00. This seems almost too good to be true, so I'm asking if anyone has had any experience with this box? I'm not up on my PCI terminology, but as I understand it, the TE405P can only be used in a 32 bit 33Mhz slot at 5.0 Volts. This SC1425 lists a 1x 64-bit/1xxMHz PCI-X slot under it's expandability information. I'd venture to guess this is probably NOT going to work with a TE405P. That being said, if it works, great. If not, what 1U boxes are people using IN PRODUCTION w/ TE405P cards? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recommended Linux Dist. for Asterisk
If your used to RH keep using it. Since I am a person that has used RH for many years I have gone with CentOS which is RHEL via GPL. It's great and there yum servers are always up and running. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Shiflet Sent: Wednesday, April 20, 2005 11:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Recommended Linux Dist. for Asterisk I'm trying to find out what flavor of Linux people are choosing for their asterisk boxes. I have been using RH, but i'd like to try some different ones. It seems that RH is the common denominator in this rash of line noise problems. So some suggestions for what dist to use would be great. Thanks, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet?
From what I have heard it works but has still some issues. It's on sale from VoipSupply for 114.95 http://www.voipsupply.com/product_info.php?cPath=95_111products_id=331 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andre Normandin Sent: Wednesday, April 20, 2005 10:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet? http://www.grandstream.com/y-gxp2000.htm Looks like the phone is $139 from DigitNetworks.. Price looks good.. If anyone has one working with Asterisk, how does it sound/work? Also, does it have caller ID with name? The Budgettones only support plain old callerID number.. Very annoying!! Thanks, - Andre ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] queue - transfer calls
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dov Bigio Sent: Monday, April 18, 2005 9:16 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] queue - transfer calls Hello, I am setting up an ACD using *, but found a an issue that I am not being able to resolve, and this might impact our * implementation. We have a call center with 4 agents, which should receive calls from their queue. But we also have a call center management team which should be able to talk to end customers in case the first level call center is not able to solve the problem. There are two issues there: The agent cannot use the soft-phone TRANSFER button.. she has to press the pound key to transfer. This is not a 'terrible' issue, since it is just a matter of educating agents. This one can be fixed if you want by going with the paid xten pro software. It has a transfer button. Attended transfer: If the agent transfers the call to someone in the management team, the call is immediately transferred, and the agent is not able to talk to the manager before. Is there a way to allow an agent to talk to the management befora actually transferring, so that he can explain the issue in advance In stead of transferring to the next level support have your agents park the call to lets say 700 it should give you something like 701 then call the next agent tell them what the problem is and to pickup exten 701. Thank you very much Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT]: Wiki Etiquette
If there is a mistake that you can fix then do so. If it's your option to do it differently then add a note to it and put your text after that. The Wiki is an open and should always be open. But as you stated Etiquette goes 2 ways and you should not go around removing text because you don't agree with it. Just edit the mistake and add your comments. Thank you for having good Etiquette. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Thursday, April 07, 2005 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [OT]: Wiki Etiquette Hi folks, I recently registered with the wiki site to fix a few things I've noticed, and I had a question: Is it proper to delete other people's additions if they are obviously incorrect? My main concern is for the content, which is ( well, was ) false. On the other hand, I do not want to start a pissing match with anybody because of bruised egos. Further, in some cases that I've seen, the OP might have a valid point, but it is not one shared by the general populous. In my mind, that view should be respected, but on the other hand, I feel there should be a correction to the wiki regarding it. Any input on this would be greatly apprecaited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel bank question
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Monday, April 04, 2005 6:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Channel bank question Hi all, Quick question regarding channel banks, I managed to confuse myself ( monday...daylight saving time...no coffee ). If I have 10 copper wires coming in from the phone company, and I want to get a channel bank that will turn those into a t1 to feed into an * box with appropriate hardware, do I want an FXS or FXO channel bank? You need a channel bank that has at least 10 or 12 FXO ports. I recommend an Adtran 750 or 850. You can get them on EBay for around $ 400 to 500. But most are pre-configured with FXS. You will need to either switch some of those card out. Then you just put in a T110p card into the asterisk. Also if you get this C/B with 12 FXO you can have the other 12 with FXS for normal analog extensions. While I'm at it: Are there specific features I should be looking for? Is there a specific company everyone's had good luck with? Any recommendations on this or otherwise? Thank you. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hello Everyone
Welcome, Yes I have used it. It's great to get started. Give it a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bagan Jermal Sent: Friday, March 25, 2005 6:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hello Everyone would like to test this e-mail list. anyway, have anybody here install and run [EMAIL PROTECTED] how was it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor System for T1 failure.
Hello I am looking for 2 things to add to our Asterisk servers. I would like to know if there is any way to monitor the PRI/T1 lines via the Asterisk Server to see if they go down. If they go down then email a notice to us. Also would like to extend this to if the Asterisk goes down that it also can email a notification. This software should be able to either run on the Linux box running Asterisk or for our safety on a 2nd box. Thank you, Ariel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie pointers
We all at one time were in the same boat. Well my suggestion is do allot of reading on the wiki for asterisk, 2nd get your self a PC load [EMAIL PROTECTED] and start working with it. It has everything you need to get started. And it works. Now you don't have to keep it with the supplied .conf files you can change them and add to them. But it will get you started the right way. Also the chat on freenode #asterisk is very good for information. Wiki = http://www.voip-info.org/wiki-Asterisk [EMAIL PROTECTED] = http://asteriskathome.sourceforge.net/ Good luck and welcome to our club. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fred Blaise Sent: Thursday, March 24, 2005 6:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Newbie pointers Hello all I have come to Asterisk with no previous telco experience. As I will be playing with Asterisk really soon, I would like to have some pointers as to some tutorials in telco that could help me get into all this. I am quite a beginner, don't forget :) Thanks a lot! Best, fred ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hi there..
All the samples are on your system /usr/src/asterisk/configs/ the files have a .sample on them. Also there is allow of information on the Wiki http://www.voip-info.org/wiki-Asterisk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bharat M. Sarvan Sent: Thursday, March 17, 2005 6:21 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Hi there.. Hello Everybody, This is Bharat here. I am on the way of learning Asterisks, and I just wished to know how I go about if got to write dailplans for outbound calls and inbound calls. If you could provide me with a simple example, I could get thru. Waiting for your response Regards Bharat M. Sarvan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Kernel 2.4 or 2.6 for the latest asterisk ??
This question really has no one reply. The different Linux builds all have there reasons. If your used to Fedora Core 1 then that is what you should use. I use CentOS which is a clone of RHEL 3. They have just released there Version 4 which is based on RHEL 4. It works and since I am used to the way RH does there settings I like it. But it's really up to you. Fedora is good and works. I just don't use it do to it's mainly for RH to develop there newer system from it. But I know that many use without problem Debian, Fedora, Slackware, Gentoo and many more. There is even a group that is working with FreeBSD. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Tuesday, March 15, 2005 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Kernel 2.4 or 2.6 for the latest asterisk ?? Hi there Just a quick question. I have been playing around with asterisk CVS-1.0.02 on fedora core 1 (2.4 kernel) and I would like to have a look at asterisk v 1.0.6 but am still a little uncertain which linux kernel is best to run on ?, can I use Fedora Core 3 (is it the preferred kernel) or should I stick with FC1 Ps - the only additional hardware in the box will be a digium single port E1 Any advice would be greatly appreciated Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom phones do not talk to each other and cannot answer when we pickup
You need to call PBXware it should not have anything to do with the phone. If a phone registers there gui should put it in the correct context. If it does not then get your money back due to there non standard setup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Thursday, March 10, 2005 4:09 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Polycom phones do not talk to each other andcannot answer when we pickup We have bought PBXware GUI from Bicom systems and configured extensions with Polycom Phones as UAs. The Polycom Phones can dial out and make calls but I cannot make extension to extension calling. Googling did not help much. As you may be aware PBXware is a closed source software GUI from Bicom Systems for configuring extensions. It is a good tool to configure and manage users and phones but it does not allow to do any of the customization tasks that are possible by directly editing the .conf files, which may be required in for Polycom. However if this is an issue of configuration on the Phone itself, we want to be able to make changes and fix this problem. Any tips? Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] any good redhat 9.0 rpm reposiroty?
Try this site: http://fedoralegacy.org/ they have most of the things there for RedHat 7.1 on to Fedora Core 1 items. - Original Message - From: Muhammad Muzzamil Luqman [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, February 18, 2005 1:48 AM Subject: [Asterisk-Users] any good redhat 9.0 rpm reposiroty? I have been googling for the RPM kernel-source-2.4.25-040218.i386.rpm or kernel-source-2.4.25-040218.i686.rpm for the last 59 hrs and couldn't succeed. Can someone suggest me some good Redhat Linux 9.0 rpm repositories. And will the Debian deb work with redhat or not? Kindest Muhamnmad Muzzamil Luqman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk@home greek letters and suggestions
Great setup for the [EMAIL PROTECTED] .06. I have a few questions about the console mode. If you go to the Ctrl Alt F9 area you see asterisk loaded but it's displaying some funny Greek letters. I did the following but it did not help. Eliminating some internationalization errors: In /etc/sysconfig/i18n, the first line reads something like LANG=en_US.UTF-8. Change it to read LANG=en_US, and reboot. I did this but it did not help. Next is some request and maybe some help. 1) I would like to see an upgrade path for your product from .5 to .6 and so on. (I don't like having to rebuild the complete server every time). 2) I would like to know of a good way to change the name of the server from asterisk1 to something else. (will this affect the installation). 3) Maybe a built in dyndns.org program for updating the IP address. Just a few items other then that it's a great product. Thank you for all your work on it. Ariel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP!!!!!!!!
Yes turn off silence suppression. xlite - Menu - Advanced - audio settings - Silence Settings - transmite Silence: (change to yes) - Original Message - From: Julius Kidubuka To: asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 10:04 AM Subject: [Asterisk-Users] HELP Hi, I have installed two X-Lite phones and theyre able to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX. From the Asterisk CLI I see the following errors; i) Unknown RTP codec 72 received ii) RFC3389 support incomplete Anyone got ideas on how I can go about this? Thanks in advance. Julius Kidubuka "When you do the common things in life in an uncommon way, you will command the attention of the world" ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.
Daniel Eboa wrote: I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. I just burned the CD and it installed just fine on my test box. Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: samedi 12 février 2005 06:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. 1. There is a help file you can run from the Linux command line help-aah. This will tell you how to change the passwords. On a clean install it tells you this in the motd. 2. Not sure about this second one. I made some big changes in asterisk for this release. It now runs as asterisk not as root and it uses amportal to start not the startup files in /etc/init.d I think only a clean install will fix this. 3. A lot of changes in FOP too the config files are in a different place could cause this problem. Sorry about all the changes. As we get closer to a 1.0 release of [EMAIL PROTECTED] a lot of this will stabilize. --- Ariel Batista [EMAIL PROTECTED] wrote: Hello, Great job on the [EMAIL PROTECTED] project. Looks great this new version is really nicer looking. But I have a few questions. 1) For the new web access http://localIP/maint how and where do I change the password. 2) Since I don't use the Amp section for setup the .conf files I use my own. How do I get the asterisk server running status up. I have it running and works but shows up as not running on the web page. 3) I upgraded my system from the older .04 by downloading the new tar and running your script. Then I copied my .conf files back and rebooted. I had already changed my password and logins names before this. Asterisk is up and running without any issue's. But the Flash Operator panel comes up flashing and I can't seem to get it to work. I feel you have done a great job and I would like to thank you for your setup to us. I will be sending you a donation soon. I am at a small self employed computer consultant that has limited funds at present. This is one of the best setups for Asterisk that I have seen. I feel your name does not do it right due to it can be used for SOHO's and other setups. It's great keep up the good work. You actually make AMP work. P.S. one more question do you have an area in the freenode for chat? If you don't I would love to help out in it. Something like Asterisk-athome would be nice. Ariel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.
Roger Hanson wrote: I've downloaded 2x and burned 2 cds and get an error invalid compressed format (err=2) system halted message both times. It'd be nice to have a MD5 to verify my download is OK. It'd narrow down the problem to either the download or the burn, wouldn't it? The other day I was getting problems with downloading files over 12mg in size. They all were failing the checksum. Found out it was my driver for the nic card in my Linux box. I was using an RealTec. Changed the nic to an Intel and no problems after that. - Original Message - From: Roderick A. Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 11:55 AM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. Daniel Eboa wrote: I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. Not as of approx 5:)) PM yesterday. I downloaded, burned, and in last stage of the install ( compiling * ) right now. I don't remeber if there was a md5sum for the iso, but a binary error in hte download or bad hardware ( cd burner ) are the twom main causes of this problem. Try another download. Rod -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home 0.5 Released today
[EMAIL PROTECTED] wrote: We are releasing a new version of our one-button Asterisk install, [EMAIL PROTECTED], today. This release includes a redesigned web interface and auto-detection of Digium fxo and fxs cards. We have also fixed a lot of bugs and added numerous customer requested enhancements. [EMAIL PROTECTED] is now more secure with passwords on the web pages and better Linux security. Is there an upgrade for current users of .04? http://asteriskathome.sourceforge.net/ __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@home .05 release questions on setup.
Hello, Great job on the [EMAIL PROTECTED] project. Looks great this new version is really nicer looking. But I have a few questions. 1) For the new web access http://localIP/mainthow and where do I change the password. 2) Since I don't use the Amp section for setup the conf files I use my own. How do I get the asterisk server running status up. I have it running and works but shows up as not running on the web page. 3) I upgraded my system from the older 04 by downloading the new tarand running your script. Then I copied my conf files back and rebooted. I had already changed my password and logins names before this. Asterisk is up and running without any issue's. But the Flash Operator panel comes up flashing and I can't seem to get it to work. I feel you have done a great job and I would like to thank you for your setup to us. I will be sending you a donation soon. I am at a small self employed computer consultant that has limited funds at present. This is one of the best setups for Asterisk that I have seen. I feel your name does not do it right due to it can be used for SOHO's and other setups. It's great keep up the good work.You actually make AMP work. P.S. one more question do you have an area in the freenode for chat? If you don't I would love to help out in it. Something like Asterisk-athome would be nice. Ariel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.
Thank you that worked for the passwords.. Thanks - Original Message - From: dean collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, February 11, 2005 10:06 PM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. If you log in via cosole and type in the help-aah command it brings up the new security requirements for changing passwords I cant help you with the rest as I installed from scratch using the iso. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ariel BatistaSent: Friday, February 11, 2005 7:55 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. Hello, Great job on the [EMAIL PROTECTED] project. Looks great this new version is really nicer looking. But I have a few questions. 1) For the new web access http://localIP/mainthow and where do I change the password. 2) Since I don't use the Amp section for setup the conf files I use my own. How do I get the asterisk server running status up. I have it running and works but shows up as not running on the web page. 3) I upgraded my system from the older 04 by downloading the new tarand running your script. Then I copied my conf files back and rebooted. I had already changed my password and logins names before this. Asterisk is up and running without any issue's. But the Flash Operator panel comes up flashing and I can't seem to get it to work. I feel you have done a great job and I would like to thank you for your setup to us. I will be sending you a donation soon. I am at a small self employed computer consultant that has limited funds at present. This is one of the best setups for Asterisk that I have seen. I feel your name does not do it right due to it can be used for SOHO's and other setups. It's great keep up the good work.You actually make AMP work. P.S. one more question do you have an area in the freenode for chat? If you don't I would love to help out in it. Something like Asterisk-athome would be nice. Ariel ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autodetecting faxes
Asterisk wrote:
I have managed to get spandsp working, and if I dial a specific
extension I can receive faxes. WhooHoo.
However, I was wanting to use the fax detect option in order to
allow individuals to receive faxes, but can't get that to work.
Given the following extensions (mainly copied from examples on the
wiki), why is the call simply passed onto the sip device rather than
being detected as a fax ?
Asterisk CVS-HEAD-02/02/05-07:11:04 built by asterisk on a i686
running Linux
Spandsp is 0.2pre9
Incoming lines are E1 line 30 channels PRI.
Many thanks.
Julian.
===
exten = 442781,1,Goto(fax,1,1) ; dialling this number works
exten = 442781,2,Hangup()
exten = _4427XX,1,Answer() ; dialling any number in here does not
add this:
exten = _4427XX,2,Wait(3)
\/
exten = _4427XX,3,Macro(dialsip,${EXTEN:3})
exten = _4427XX,4,Hangup()
exten = fax,1,Goto(fax,1,1)
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(3)
[fax]
exten = 1,1,Macro(faxreceive)
exten = h,1,system(/usr/sbin/mailfax ${FAXFILE} ${EMAILADDR} \
${CALLERIDNUM} ${CALLERIDNAME})
exten = h,2,Hangup()
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP with SUSE9.2 (Apache2)
Title: AMP with SUSE9.2 (Apache2) I think you should post this information on the Wiki. It's our main location for all of these things. Thank you. - Original Message - From: Keith Burns To: asterisk-users@lists.digium.com Sent: Thursday, February 03, 2005 2:09 PM Subject: [Asterisk-Users] AMP with SUSE9.2 (Apache2) Hi all, After pinging the AMP userlist at SourceForge, I got a great step by step explanation as to how to set up AMP for Apache2 (some maybe obvious stuff that wasnt in the Newbie Guide). Thanks to Jason Becker of Coalescent Systems. If anyone needs me to post Jasons instructions here, I can, but they are in a thread called AMP noob issues with Apache2/Suse9.2 at SourceForge. Again my thanks Jason, looking forward to using your software. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Sipura-841 phone.Mike volume problem.
Well I just need to say I got my phone last week. Here is my quick review of the phone and hope that someone has a possible fix for it or I will be sending it back. First the phone is nice looking in my view and it's heavy so it feels like a real desk phone. But it has these stick, gummy or I really don't know how to describe the bottoms on the phone. There good size but when you press them they feel like they are dirty with some sticky stuff on them. They don't get stuck but feel that way. Here is my problem. The mike is really bad on the phone. It's not the hand set or the plugging via the 2.5 plug on the side. It's something to do with the phone hardware internal. I can tap on the mike and I hear a faint tap on the other end. But unless you scream into the handset or mike they can't hear you. I need to see if there is some type of fix for this. Registration and setup is just like all of the Sipura devices via the web. In fact most of the setting are almost line by line like the Sipura 2100. Looks great on how they did that. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Certification
I just have to make my view known about this. 1) I agree that one is needed but! 2) I feel that there should be a way to get a self study course which will lead to a way to take a test for the Certification. 3) Cost and who set this up is really something that I think should be done first from digium and not out side sources. 4) Which Linux Distro will this be based on is important to make this all work seamless. 5) Certification must only be for Stable released version and not head. (Released is the onlyproduct that should be used for real worldinstallations). Can't be oh we have a new thing so we are now saying you need a new certification this is wrong. 6) Support group for the Certification. 7) A low cost way to get this Certification. 8) If we do get Certification our very own advanced support from Digium. 9) Real documentation is a must and needed if this is to work. Before you Certify people there must be some manuals and I mean real ones. Not the wiki or people just say well go to the wiki and read for your self. This is not the point and should not be. If you want to be considered real to have an Asterisk Certification then you must pay the price to have proper documentation and support behind it. There are people like my self that have been working with Asterisk for more thena year. Some of us have even made a biz out of installing and supporting Asterisk systems. There should be a way to get these people a quick and easy way to get Certified. Like many of us we have been helping via the freenode and other sources others this will be a problem in the future due to with Certifications you will have people who feel since there certified the don't give help unless there paid for. This will be a step backwards for our great community. So in my view we are not ready for this Certification and I do not support the current view that is posted about it. Lets get real and do something professional not halve done. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using SPANDSP for faxes
Eric Hall wrote: I installed spandsp on our asterisk server to get faxes. It works however the images are a little off. Sometimes a few pages will be together, pages missing and sentence missing. Is this normal for this program? Yes it is with some fax machines. We had to make our own program that take the image and sets it correctly for viewing. It's not a GPL program it's one we got as a test. If all goes well we will post what we are doing to fix the problems. The programmer says it has to do with the libtiff library's. Any input would be great. Thank You Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Voice Prompt Info
Warren Burstein wrote: One more thing about prompts, it's better to say for sales press 5 than press 5 for sales, because by the time you hear sales you've already forgotten what number it was. If you add the sounds all you need is For Sales recorded the new sounds have press # already. So you don't need to get any additional recorded items except the one that says For Sales by Allison. If you want have her record Press as an additional recorded item. So record for sales press and the digits (you could use the digits that come with *, but a sentence in two voices sounds very funny, I know, the user directory on an old IVR of ours works that way). That way when you need to change the numbers the menu you can do it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-Re: [Asterisk-Users] help with detecting fax.---fixed.
Original Message
From: Ariel Batista
I have Spandsp working fine. Asterisk sees a fax on the zap port and
redirects the call to the fax-in area.
This works if I have a simple dialing rules that goes answers first
and waits 10 secs then goes to the next item. If it hears a fax it
goes to the right place. Here is a sample that works. Also you need
to have the fax lines in the context that all calls from the pstn
side come in. In my case it was inbound.
[incoming]
exten = 2019,1,Goto(test,s,1)
[test]
exten = s,1,answer
exten = s,2,wait(5)
exten = s,3,Macro(stdexten,Sip/101)
exten = fax,1,Goto(fax-in,s,1)
[fax-in]
I am adding the notes as what we did to fix this problem in case
someone else needs these fixes. Note this problem is due to what I
feel is a bug in asterisk. But there is a work around. Here you need
to add an extra line which is not really used. Asterisk will work if
you add this to it.
exten = s,1,NoOp
exten = s,2,Dial(zap/1)
Instead of
exten = s,1,Dial(Zap/1)
We now actually have it going to spandsp now and it's working from
the macro.
But if I use this following macro it just detects the fax then goes
to congestion instead of the fax extenstion. [macro-followme]
;
; Standard single line follow me then to voicemail
; $ARG1 first device to dial to. $ARG2 2nd device to dial.
;
exten = s,1,Answer
exten = s,2,Wait(5)
exten = s,3,Dial(${ARG1},20)
exten = s,4,Dial(Zap/g1/${ARG2})
exten = s,5,Voicemail(u${MACRO_EXTEN})
exten = s,104,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?105:107)
exten = s,105,Dial(Zap/g1/${ARG2})
exten = s,106,Voicemail(b${MACRO_EXTEN})
exten = s,107,Voicemail(u${MACRO_EXTEN})
;
I am calling it with this setup.
exten = 2019,1,Macro(followme,Sip/101,16502468900)
I have fixed the problem. See notes below.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help with detecting fax.
I have Spandsp working fine. Asterisk sees a fax on
the zap port and redirects the call to the fax-in area.
This works if I have a simple dialing rules that
goes answers first and waits 10 secs then goes to the next item. If it hears a
fax it goes to the right place. Here is a sample that works.
[incoming]
exten = 2019,1,Goto(test,s,1)
[test]
exten = s,1,answer
exten = s,2,wait(5)
exten =
s,3,Macro(stdexten,Sip/101)
exten = fax,1,Goto(fax-in,s,1)
[fax-in]
exten = s,1,Dial(zap/1)
But if I use this following macro it just detects
the fax then goes to congestion instead of the fax extenstion.
[macro-followme];; Standard single line follow
me then to voicemail; $ARG1 first device to
dial to. $ARG2 2nd device to dial.;exten =
s,1,Answerexten =
s,2,Wait(5)exten =
s,3,Dial(${ARG1},20)exten =
s,4,Dial(Zap/g1/${ARG2})exten =
s,5,Voicemail(u${MACRO_EXTEN})exten =
s,104,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?105:107)exten = s,105,Dial(Zap/g1/${ARG2})exten = s,106,Voicemail(b${MACRO_EXTEN})exten =
s,107,Voicemail(u${MACRO_EXTEN})exten =
fax,1,Goto(fax-in,s,1);I am calling it with this
setup.exten =
2019,1,Macro(followme,Sip/101,16502468900)
Here is the error I get. And yes I have exten
fax,1,Goto(fax-in,s,1) in the inbound context as well.
-- Executing Macro("Zap/1-1",
"followme|Sip/101|16502468900") in new stack002 -- Executing Answer("Zap/1-1",
"") in new stack003 -- Accepting call from '6502468901' to '2019' on channel 0/1, span
1004 --
Executing Wait("Zap/1-1", "10") in new stack005 -- Redirecting Zap/1-1 to fax
extension006 == Spawn extension (macro-followme, fax, 0) exited non-zero on
'Zap/1-1' in macro 'followme'007 == Spawn extension (inbound, fax,
1) exited non-zero on 'Zap/1-1'008 -- Executing
Congestion("Zap/1-1", "") in new stack009 -- Channel 0/1, span 1 got
hangup010
== Spawn extension (inbound, fax, 2) exited non-zero on
'Zap/1-1'011 -- Hungup 'Zap/1-1'
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceMail Problem or bug?
Ok I have a question about the voicemail program with Asterisk. This is with the current head CVS as of 7/28/04 and every other one before it. When apending to a message that you forward, to stop recording you press any key. But it take however long you record for it to save the message then return to a menu. If your add 2 minutes of recording it takes that long to return to you. If you press the # key it will delete the message and your out of there. This is a major problem with people how use cell phones. You get dead air while your waiting. Is there a fix for this. Does someone have a better voicemail program than this one. - \ \\_ Ariel Batista // / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Doublehash transfers
John Todd wrote: At 12:00 PM -0400 on 7/23/04, mattf wrote: Hello, I recently tried an upgrade of CVS on my test server today and found that the res/res_parking.c file is completely gone. This is where I had to go into the code every time I do an upgrade and change the code to allow for doublehash transfers instead of single hash transfers: If someone is currently using doublehash on a recent CVS could you be so kind as to email me a copy of your patch? I hate being a me too poster, but the double-hash patch I have implemented four times now, and I know at least three other people who have also gone well out of their way to put that patch into their system. Making this an official modification would be ideal, in my opinion, especially if it is selectable (default=single #, to retain previously expected functionality.) I secound that motion. This is something that should have been done many moons ago. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anybody recommend a good T1/PRI provider?
We use Allegiance it's been very good to us. It's now X/O but still we have had no outage. We have a ATT and a Sprint line as well. Sprint is evil and sucks. ATT is good but for the price I can get more from Allegiance down here in South Florida. - Original Message - From: Deon Rodden To: [EMAIL PROTECTED] Sent: Thursday, July 22, 2004 8:53 AM Subject: [Asterisk-Users] Can anybody recommend a good T1/PRI provider? We're in South Florida, right now we primarily use Xpedius PRI and 2 IDS PRI's. We were looking at getting a MCI PRI but upon reviews from a few people I know that have them, I decided against it. Can anybody recommend good/reliable T1 PRI providers for South Florida? Should I stick tobig namesonly (ATT, Sprint, etc.)? 550 Fairway DriveSuite 210Deerfield Beach, FL 33441Online: www.webunited.net Deon Rodden Toll Free: 1-877-538-5969 x 208Phone: 954-418-8884 x 208Fax: 954-418-8635E-mail: [EMAIL PROTECTED]
Re: [Asterisk-Users] where can I get toll-free number?
Arick Davis wrote: www.Kall8.com Expensive .068 that is 6.8 cents per minite. And they support SIP termination. Arick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: Monday, June 14, 2004 9:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] where can I get toll-free number? On Jun 14, 2004, at 9:06 AM, Jay Milk wrote: I'm using zoneld numbers which I can terminate on any US number -- http://ld.net/mu has various options. You basically get your incoming voicepulse, broadvoice, etc line, then get an 800# to terminate on those lines and you're in asterisk. Through this, I also have tollfree numbers to my cellphones and fax... Yeah, but that's more expensive then NuFone's $0.029/minute 800 rate. If you're looking for an 800 number that points to an existing device, then ld.net probably a great way to go. If you're looking for 800 VoIP services, then there's no reason to stack services like this. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Binding to 2 nic's for trunking two asterisk servers
I have a problem in that when you use IAX2 for trunking and have 2 nics one is used to connect directly to 2nd Asterisk server how do we get the outside Nic card to take IAX connections? Is there any way to get this working via two paths? There is only one bindipaddr=10.1.1.1 for internal trunk but outside address section? - \ \\_ Ariel Batista // / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialogic D/41E
Kubat, Philip wrote: I have an old Dialogic D/41E card. I searched the mailing list and it looks like there was or could be a module for it. Although the posts never specified where or how. Is a D/41E usable w/Asterisk? If so how does one obtain the drivers? Or is it a better pots adapter for the wastebasket? In my view it's better sold on ebay and buy a good TDM04e from Digium Thanks Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disable blind xfer
Timothy R. McKee wrote: My SIP users need to transmit the # key as part of data entry. Asterisk intercepts and initates a transfer function. I'm almost positive I've seen this discussed somewhere, but none of my searches are finding it. In your dial plan take the Tt out of it. exten = 123,Dial(Zap/1,20,Tt) to exten = 123,Dial(Zap/1,20) Anyone have a handy answer? Tim McKee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] example of mulity company extension.conf needed.
I am trying to get a building that has 3 company's on one asterisk server. I need to make the IVR via DID take them to there right menu. So far I have everything working except when they goto via standard_marco to an extension and are sent to voicemail they are dropped off in the first menu and not the one they came from. In other word sent to another company's menu. If it happens to be the first company it work fine. I do not have a defaultcontext= setup due to I want each company to have there own context. But I am using standard_macro for all the extensions. My extension.conf file is extermly large so it is not a good idea to post it here. - \ \\_ Ariel Batista // / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom 200 phones.
I have about 5 snom 200 phones working fine with everything. Voicemail, Transfers and all. Except I can't seem to use them to pickup parked calls nor place a call on park. I also have sipura-2000 with analog phones that are able to pickup parked calls and to park them. Most of them are on firmware 2.04g I have upgraded one to 2.05c for testing but this did not fix the problem. I get no error message on the CLI and I am at a lost of where I can begin to look for a problem. I have other Sip phones working fine. Cisco 7960'g, IpDialogs They all work fine. ATA 186 and Sipura-2000 are also working fine they all can park a call and pick them up. - \ \\_ Ariel Batista // / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA 188 and fax
Osvaldo Mundim wrote: Hi, Does anybody have ATA 188 working with any kind of fax machine? I've tried many different configuration following the Cisco Online Manual and I couldn't get this working with Asterisk. I don't know what the difference is between the 186 and 188 other then the extra nic port. But we gave up on the 186 for doing any fax or data calls. We switched to Sipura-2000 and using Ulaw faxing works. Data calls well we can get them working but only at 28,800 bps. Good luck I were trying do change the ATA Connect Mode and Audio Mode reading the (http://www.cisco.com/en/US/products/hw/gatecont/ps514/ products_configuration_example09186a00800d698e.shtml) and allowing all codecs on Asterisk and did not work either. best regards Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with greek leters in CLI
I have been having a major problem with after some installations of Asterisk about every 3rd one the CLI will come up in some strange looking greek letters. This problem does not happen all the time but once it happens I was not able to clear it up. Well with the help of a unix/linux expert we have found a fix for it. If this happens to your system here in the US you need to change the following lines (This mainly has been happening on RH 8/9 and Fedora core 1). edit file /etc/sysconfig/i18n and make the following changes. LANG=en_US SUPPORTED=en_US:en SYSFONT=lat0-16 Save it and you then need to reboot the system. - \ \\_ Ariel Batista // / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip software
James Moran wrote: Anyone have any suggestions on free sip phone software for windows?? Only have one IP phone and want to have one other computer hooked up to my Asterisk box for testing. xten x-lite. Works great free and works just plain works! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_motv: Request for Comment
Mark Spencer wrote: Any feedback on: a) The idea itself -- is it a good one or is it stupid? Now this is just my views. No I do not feel we need to be sending any information back unless we want to. Like someone else said a sub job that is turned off by default. My preference would be no communication back. I would like to see on you web site more information on stable builds, bugs and easyer way to determine the version your running. Also maybe some feed back form that we can fill out and sumit to you. But all of them are manual and not automatic. b) The way to make it deployed without sneaking a call home in on anybody that doesn't want it? Thanks! Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
Bisker, Scott (7805) wrote: I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI-T400P-Asterisk-T400P-Adtran 750(L36 Firmware)-RAS Server. I have 4 Zap channels signalled FXO_KS to the 750 with FXS_LS channels, On-Hook messaging disabled, the rest defaults for the channels. In zapata.conf I've tried with both busydetect=yes and busydetect=no busycount=6, busycount=10, callprogress=yes, callprogress=no all combinations. We have 4 750's and one TSU 600 working with PC anywhere for data communications for our support department. We have on this system 2 T400P's. The only thing I can say is who are you getting your timing from. We are able to get modem calls and faxes without problems. But this is only using PRI from Allegenice. We also have a LD service T1 from Sprint that is in no way able to handle any data calls. Our Adtrans are out of the box without any changes to them. This is our settings in our zapata.conf. ; Enable echo cancellation echocancel=yes ;echocancelwhenbridged=yes immediate=no ;adsi=yes usecallerid=yes hidecallerid=no ;callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes musiconhold=default signalling = fxo_ks Hope this helps. The weird thing is, that if I forward the incoming call from the PRI out another channel on the PRI into a POTS line hooked into the RAS server, the connection is fine. In my view, that rules out the PRI and points the blame at either how the adtran is configured, or the how the channel itself is configured. Can anyone with a _working_ configuration similar to this chime in with some config info on the Zap channel and the channel bank config? Thanks in advance. -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
Bisker, Scott (7805) wrote: Same as mine. Do you know off the top of your head what firwmare you're using? Also, what RAS card do you have on your PCAnywhere side? I have firmware L36. Ras card is a Digikey 4 port board on one NT server and others are using the normal serial ports on the servers. The desktops are using there modems connected to there PC's via Serial cables. All our modems are USR Sporters 56K we have about 20 of them. Except for 3 USR Courier 56K. For our fax board we are using BrookTrout I4P on a Windows 2000 server with ZataFax. Everything is working off the timing from the PRI line. Asterisk is older on this installation. This installation is still using .5 from CVS 12/05/03. I belive if it works leave it along! And it works just fine! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
Bisker, Scott (7805) wrote: I'm timing off my PRI from Verizon as well. This is mind boggling. All my Fax machines are fine. The modems connect, but drop the calls after about 1-2 minutes regardless of busydetect. That was our exact problem with Sprint when we had there T1 line. We decided to switch to Allegence and problem was gone. I just tired a data connection through the Sprint LD line we have and after 2 minutes we got dropped. If we tell the modem to only use lower speed like 28.8 it will stay connected for longer time. At 19.2 no problem they will stay connnected. Verizon might have the same problem with there lines like Sprint. That there not data lines but voice only. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ariel Batista Sent: Wednesday, April 07, 2004 12:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750 Bisker, Scott (7805) wrote: Same as mine. Do you know off the top of your head what firwmare you're using? Also, what RAS card do you have on your PCAnywhere side? I have firmware L36. Ras card is a Digikey 4 port board on one NT server and others are using the normal serial ports on the servers. The desktops are using there modems connected to there PC's via Serial cables. All our modems are USR Sporters 56K we have about 20 of them. Except for 3 USR Courier 56K. For our fax board we are using BrookTrout I4P on a Windows 2000 server with ZataFax. Everything is working off the timing from the PRI line. Asterisk is older on this installation. This installation is still using .5 from CVS 12/05/03. I belive if it works leave it along! And it works just fine! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RPM packages
Christopher C. Howard wrote: Andrey McRory built a RPM dist for * but I can't seem to find it anywhere.. Any hints where I might be able to find this package that has matching kernel? This is what I found for rpm. http://www.voip-info.org/wiki-Asterisk+RPM Hope this helps. Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk server lockup
Gary Franczyk wrote: Hello, We are trying to deploy a new asterisk server with a Wildcard T400P (quad T1) card. It uses a custom voice recording app written in the perl AGI. Now that the machine has been in production, it seems to lock up within 24 hours of reboot! When it locks, we can ping the machine, but we cannot log in using telnet or ssh. Asterisk stops answering the phone and our Big Brother monitoring scripts stop sending data. Nothing is shown in the messages log. The script we are using seems to be ok, since it exits after every We have seen the exact problem with a production system. We found out that it was due to the monitor application connection. We feel it has a memory leak and there is never any error other then it starts to slow down in playing back message and other small signs. We have stopped using the Gastman and other monitor applications. The problem has gone away. As a test we started using the Gastman again and after 24 to 36 hours it would do the same slow down then lockup. We also have set up a schedule event to restart the system once a week. And have not used Gastman nor any application that uses the monitor. Sorry this is not a fix but we are looking into either redoing the monitor application or waiting till someone fixes this. My first guess is that it could be running out of memory (since I can still ping it, the kernel must be working to some extent). I don't have a console attached to this machine yet, so I can't tell you what is displayed on the screen yet. (I will get one soon) Has anyone seen any asterisk lock ups like this before? Thanks Gary Franczyk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
