[asterisk-users] How to busy out zap channels
I know this topic came up many months back and some discussions were being had on how to do this within the Zaptel drivers. However, I'm looking for even a crude hack that someone has put together to get this done. We have PRI's and LD T1's that are load balanced on two boxes. The hunt order goes from box to box as far as the spans are concerned. There are times that I would like to busy one out so that calls gradually role to the new box and I can eventually take one out of service. What I was thinking is to create a script that I could tell the specific channels and it would go through and initiate zap calls to an empty meetme. Basically bridging all of the available zap channels on a given span together. Then the trick is monitoring the hangups so that it can initiate a subsequent call immediately following. Once all of the channels in a span have been bridged, I can then bring the box down. Nasty huh? Anyone have a better idea? Or do they have anything like this so I'm not putting it together? Thanks, -Brian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel Bank Recommendations
On 8/29/07, Mark Bell [EMAIL PROTECTED] wrote: Need to add some fxs and fxo ports behind a fonebridge2 box any recommendations a channel bank We're using a Rhino here and haven't had one problem with it. It's connected to an analog fax server and lights up for hours at a time. Probably been up 300+ days since we bought it. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where is 1.4.12?
On 8/29/07, Steve Totaro [EMAIL PROTECTED] wrote: Kind of harsh for am employee of Digium on a public Asterisk mailing list, don't you think? Enough The Digium/Aseterisk bashing seems to be at an all time high recently. You seem to be involved in a lot of it. Russell has given most of his time and life to this project over the years and to see someone say inability to fix bugs I'm sure frustrates him. Not that I view his remark uncalled-for, because I agree that was very trollish. Let's get back to real problems. Like Russell says, if you got a bug, submit it or fix it. If you got hardware problems call Digium. If you don't like Asterisk/Digium go on to something else. The bashing is ridiculous. My .02 ___ --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] Channel Bank
On 5/6/07, Doug Lytle [EMAIL PROTECTED] wrote: The Adit 600 is a favorite of mine. Doug Would second the Adit too. We are running a Rhino now and have had no problems with it. -Brian ___ --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] Cacti/Nagios monitoring, what do you want graphed.
On 4/12/07, Brandon Kruse [EMAIL PROTECTED] wrote: Hey guys, What are some of the numbers you guys want graphed? Curious how you are going to do this and will it be backwards portable. One of our engineers wrote an app that queries the manager interface to build RRD data. That's sent over to Cacti to monitor active calls on a box. I could think of many things queue related that would be good to have, but then again, shouldn't that be done somewhere else? Being able to break out calls in things like Zap(trunks), SIP, IAX, etc would be very useful. I have an 8 port T1 card and it would be nice to see how many calls I have on each port. Thanks for getting this going Brandon, I'll follow closely. We are heavy Nagios and Cacti users here. -Brian ___ --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] Any quiet 24 port POE switches out there?
On 1/3/07, John French [EMAIL PROTECTED] wrote: I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The 8port Netgear switch on my desk doesn't have any fans. FS108p. Not sure if they make a 24port switch or not. -Brian ___ --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] webvoicemail
On 12/13/06, Ed Nuñez [EMAIL PROTECTED] wrote: I've been trying to find where to download the Web Vmail application and instructions on how to install it for Asterisk BE. Any ideas? Is this any different than the vmail.cgi that comes with the open version? Otherwise, you will just need to grab a compiled copy off of another box. Only needs vmail.cgi and a couple of supporting graphics. -Brian ___ --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] Polycom MyStat
On 12/13/06, LST [EMAIL PROTECTED] wrote: I think that is strictly a Polycom to Polycom thing (Buddywatch). I do not believe it affects Asterisk (i.e. Busy = DND). With that being said, I don't think it works very well even with all Polycom phones. I can change my status to Busy and look at the other Polycom Phones and they still show me as Online. (Yes, I have bw set to 1.) Does that mean that DND doesn't show up as a hint either? -Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Re: Voicemail Press '0'
On 10/10/06, LJ [EMAIL PROTECTED] wrote: In my Asterisk 1.2.9.1 installation I use the following:in voicemail.conf include the following:exitcontext=vmloginoperator=yes Sorry to revive a month old thread but here was the easy button solution for me. With debugging on I did a reload app_voicemail.so from the CLI I saw the following Nov 15 06:28:16 DEBUG[24613]: app_voicemail.c:6012 load_config: VM Operator break disabled globally That happened even with the operator=yes in my voicemail context. So I moved the operator=yes up to the general contextandthat message went away, and my 0 option worked fine. Hope this helps, -Brian ___ --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] Digium vs. Sangoma
On 10/23/06, Unmetered Pipe [EMAIL PROTECTED] wrote: I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ? That you are a troll? -Brian ___ --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] Rpath PoundKey 1.2
Is there a way to correct the problem or can the files be generated? Did you run the registration program? Asterisk won't start unless it's registered with Digium. -Brian ___ --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] Frappr mapper
OK, so this list claims to have1000's of users on it. Let's see where they are I was putting myself on frappr map for something else and found an unused asterisk map. I saw Olle on that (imagine that) but no mention of it on the lists. So. Frappr yourselves. Let's see how many Astriholics there are out there. http://www.frappr.com/asteriskusers -Brian ___ --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] anybody get experience with dell powerconnect 3424 and QOS for asterisk traffic?
On 4/22/06, Jean-Louis curty [EMAIL PROTECTED] wrote: Hi,Anyone already configured a powerconnect 34xx to prioritize voip traffic over the lan ?I just bought a 3424 for the lan and I like to give priority to voice ,thanks in advance, jean-louis Hi, do you have another layer 3 QOS switch on premise? The 3424 is QOS aware, but does not support QOS tagging and prioritization. In essence, it will just honor and pass the tags. However, depending on what you have going on in your network, I doubt you will need QOS if you just have one switch. MHO of course. -Brian ___ --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] Compatible Asterisk Connectivity Cards : Sangoma
On 4/2/06, Heidi Mendoza [EMAIL PROTECTED] wrote: Hello List!I wanted to share to everyone the following compatibleconnectivity products that my company installed in our Asterisk based soft switch. I already sent these tothe Asterisk.org site many days ago but for somereason they still have to post it. This feels trollish to me. Please don't feed them. -Brian ___ --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] cdr_odbc appears to have fields missing
On 3/29/06, Nathan Bowyer [EMAIL PROTECTED] wrote: When I look at the code, in this case calldate is actually thecdr-start value.I'm working on a patch to record answer and end as well. Thanks Nathan. Are you going to post this to the bugtracker? -Brian ___ --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] cdr_odbc appears to have fields missing
I'm currently using Asterisk running version 1.2.5 and trying to use cdr_odbc to connect to a Microsoft SQL database. I have everything running, but the insert statement being sent to database doesn't appear to have the start, answer, end information in it. Below is the insert statement that MS Profiler shows being sent. As you can see those fields are missing. INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES (@P1,@P2,@P3,@P4,@P5,@P6,@P7,@P8,@P9,@P10,@P11,@P12,@P13,@P14) Here is the record that shows up in cdr-csv ,4718,2599576,default,Asterisk2 4718,SIP/4718-af52,IAX2/visioniax-1,Dial, IAX2/visioniax/[EMAIL PROTECTED],2006-03-29 10:29:23,2006-03-29 10:29:25,2006-03-29 10:29:34,11,9,ANSWERED,DOCUMENTATION that record looks fine. Please let me know if I'm missing anything here. Thanks, -Brian ___ --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 Users Mailing List Traffic
On 3/19/06, James Harper [EMAIL PROTECTED] wrote: That being said, a mailing list with a forum interface (or a forum witha mailing list option) might be a reasonable compromise as it should meet the needs of both mailing list lovers and forum lovers (assuming itis implemented properly!) All- if you haven't tried gmail for mailing lists you are missing the boat. I have over two years of this mailing list in my gmail account now and I can't imaging not having that resource at my fingertips. vi-like keyboard navigation 2+gig storage browser accessible (including wap) threading google search capabilities rule based archiving etc etc etc -Brian ___ --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] Toshiba Strata DK-280 support?
On 3/15/06, Charles Marcus [EMAIL PROTECTED] wrote: Can anyone provide any feedback on using this system with Asterisk? Am Iwasting my time even thinking about it? I run Asterisk partnered up to the 280 (424 for us). We have a 6 cabinet installation of the Toshiba so I understand you dilemma. There are some quirks with the 280 that make it a challenge to use Asterisk with, but it's do-able. Keep in mind, you can move to the CTX or the CIX and still keep a lot of your investment and those systems play much better with Asterisk. -Brian ___ --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] IAX2 + Sonicwall
On 3/10/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote: Hi all,Has anyone else run into this, or figured out the rationale for it? I've noticed the same thing on my TZ150. I'll try that setting this weekend and see if it makes a difference. -Brian ___ --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] IAX2 + Sonicwall
On 3/10/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote: OK apart of my beleive that sonicwall is a piece of crap (personal), try to do a port forwarding for the IAX port (4569) Some people always subscribe to the Get what you pay for theory. Since they are usually priced on the low-side of the market you will hear this from time to time. We've been very pleased with Sonicwall for years. I usually don't take the flame bait from the trolls, but what the hell.. -Brian ___ --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] MixMonitor Problems -- sssshh, don't be too loud
On 3/3/06, Gary Richardson [EMAIL PROTECTED] wrote: I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811.. I'm running 1.2.1 and most of mine get cut short too. I posted this on the list a few months ago and nobody had any suggestions. BJ said I should probably post a bug on it but I haven't had time to continue to troubleshoot it. I will go to 1.2.4 (now 5 probably) and see if mine goes away. I've been watching change logs and hadn't seen anything surrounding mixmonitor so I've let it go. Please continue to update us if anyone gets some resolution. I'm glad to know there are lots of us experiencing this. That should be the catalyst to get it fixed. -Brian ___ --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] Problem with T1 installation
On 2/24/06, Doug Lytle [EMAIL PROTECTED] wrote: T1s require a D (Data) channel, unless connecting to a channel bank, Itshould be 23 voice 1 data.Also, I would strongly suggest moving to 1.2.4 Not totally true. A PRI is 23b 1d. A DS1 (US) is a 24 channel circuit. Nitin- When you stop/start asterisk does it load all 24 channels? Any errors? How about zap show channel 1 in the CLI? -Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell PowerConnect 2724 Switch and QoS for VOIP?
On 1/20/06, Sean Tempesta [EMAIL PROTECTED] wrote: Has anyone had any experience with the Dell PowerConnect 2724 GigabitSwitch and VoIP traffic?It seems to support Class of Service (CoS), but not full Quality of Service (QoS). Honoring, and tagging, are completely different things. I believe that the 2724 will just honor QOS frames. The 6000 series Dell switches are the only true Layer 3 switches with QOS. You will probably get less than stellar reviews from this crowd but it's not a bad switch. I've used a 6024 as a building core switch and not had a lick of problems. -Brian ___ --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] SPA-3000 is translating vocal sounds into DTMF
On 1/6/06, Brian Capouch [EMAIL PROTECTED] wrote: I would hope perhaps there's some kind of setting that has to do withthe way it detects inband DTMF?I'm pretty sure it's an artifact of this particular ATA; my other SIP devices are just fine. FWIW, I have the same prob on my spa3k. It's peircing and pisses my wife off greatly. -Brian ___ --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] Mixmonitor
On 11/14/05, BJ Weschke [EMAIL PROTECTED] wrote: There is a known issue right now where using mixmonitor withchan_local is going to cause an unintentional disconnect. Are you using Local/ with this setup? BJ, Thanks for the response. No, I've got nothing going though chan/local at all. It's a real straigh-forward zap to sip bridge. Nothing fancy. I'm going to try and route my calls over to another box via iax today and see if that makes any difference. The mixmonitor will be looking at sip to iax then. Let me know if you think I should file a bug on this. -Brian ___ --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
[Asterisk-Users] Mixmonitor
Hello, I recently switched over to using Mixmonitor versus Monitor to see if it would clear up some warble that I was getting in my recordings. It did indeed clear that up, but a new problem was introduced. The recordingsfor no reason will just end abnormally. There is no rhyme or reason as to when they will end, but usually after a minute or so. Here is my current setup. Asterisk v. 1-2-0rc2 Outbound calls being made via Sip Polycom phone Bridged call to Sangoma Zap channel. Asterisk doesn't report anything about it ending abnormally in debug. When you go to pull the file up, it was just cut off abruptly. Any ideas? ___ --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] dell and digium hardware
On 11/9/05, Craig Guy [EMAIL PROTECTED] wrote: Works well.I am running 1.0.9 stable on this with FC2 on kernel 2.6.9Thekernel needs patching to pick up the onboard SATA (ICH7), or we use a pci express SATA raid controller with a TE110p. Which pci-e SATA controller are you using? The one that shipped with my dell was pci-x -Brian ___ --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] Who is going to AstriCon (The Asterisk Conference)?
On 9/16/05, Steven Sokol [EMAIL PROTECTED] wrote: Hi,I'm taking a straw-poll to see who out there is planning on going toAstriCon. Enjoyed it last year, but putting it on the west coast seems to be pretty restrictive. I won't be making it. Atlanta was a good compromise. Maybe consider moving it to a more central location next year and I'll be back. -Brian ___ --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] Make asterisk call out
On 9/11/05, Andreas Moroder [EMAIL PROTECTED] wrote: Is it possible to use asterisk to call automatically a list of number. Yes, it's possible. It will require a little effort to do some scripting, but not much. They key to making Asterisk call out, will be using the call files. Here is the wiki page http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out With a little effort you can do this yourself, or request someone write something for you on the -biz list. Someone would do it for a few bucks. -Brian ___ --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] Craig R. Saxton/PACE/US is out of the office.
On 8/18/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I will be out of the office starting08/18/2005 and will not return until08/19/2005. I will respond to your message when I return. Tell me we're not going to get these until Craig gets back to work tomorrow? ___ 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 w/ XML browser
On 7/9/05, Dan Perik [EMAIL PROTECTED] wrote: PJ, You should check out the Polycom 500/501/600. I'm quite sure it has all that (although I don't use all of what you listed). IIRC, the 500's browser is crippled. I think you have to go up to the 600 to get that functionality. -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
Re: [Asterisk-Users] RE: res_cepstral
On 6/22/05, Memon, Nauman [EMAIL PROTECTED] wrote: I was told that the project has already been released in to the CVS head, and is available to us now, but not available yet for the business edition. Nothing on the cvs mailing list as of yet. -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
Re: [Asterisk-Users] HT-488 vs. SPA-3000?
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 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
Re: [Asterisk-Users] how to limit simultaneous calls
On 6/11/05, Juan Pablo Abuyeres [EMAIL PROTECTED] wrote: Hi, There is one asterisk server, and there are several locations. On each location there are 100 (SIP) extensions. The idea is to set up a limit of 10 concurrent calls for each location (because of bandwidth issues on each location). How can I do that? Check out setgroup. See if that will accomplish what you are after. http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup -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
Re: [Asterisk-Users] Opinions of Sphinx?
On 5/31/05, Alistair Cunningham [EMAIL PROTECTED] wrote: Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Alistair, Well, it's been a couple weeks and no answers on the list. That isn't encouraging, but I'm hoping to accomplish some of the same thing. Have you made any progress on your own with this? Let me(us) know... Thanks, -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
Re: [Asterisk-Users] Play MP3 during Record
On 6/9/05, Phuong Nguyen [EMAIL PROTECTED] wrote: Hi all, Does Asterisk support multi thread? I mean: Is it possible to do one of the 2 following scenarios: 1. Play a low background music when the user record his/her voice I don't know why you would want to do that, but here is a hack. Throw two calls into a meetme. One with chan/local playing the mp3, and the other your call that you want to record. Monitor the second leg out and there you go. This would take some tweeking with MeetMe flags but totally possible. Just a hack, there is probably a better way. TIMTOWTDI -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
Re: [Asterisk-Users] Queue Log
On 6/7/05, Johann [EMAIL PROTECTED] wrote: Hugo, 1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||Ray Balbin 25 (716)250-3405 2nd column is not really sure...maybe the duration? Asterisk UniqueID of the call. -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
Re: [Asterisk-Users] TDM400P vs SIP3000 x2
On 5/26/05, Andres Paglayan [EMAIL PROTECTED] wrote: I am about to start building my first ever * production server and would be nice to have some input from the list. My personal vote would be for the Sipura's. Pro's - It would make failing over to standby box much easier. You could run a small 1u box and not have to worry about PCI requirements. Lightens the load (especially interrupts) on the * box PSTN doesn't have to be located by the * box, just by an ethernet port I think if you poll the archives, you would find problems with both of them. I run a SPA3k and have had no problems with it at all. Just my .02 -Chuji ___ 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] Broadvoice delivers CID even when restricted?
On 5/24/05, Johnathan Corgan [EMAIL PROTECTED] wrote: I can call my Broadvoice DID from a outbound caller-id blocked phone, and BV happily delivers the CID to Asterisk (and then on to my IP phone display.) I've tested with the *67 prefix from a PSTN phone to make sure it was supposed to be blocked. It's not like this on all of their switches. I just tried my Nashville number, and it came across restricted. Or you let the cat out the bag! -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
Re: [Asterisk-Users] Home Usage
On 5/16/05, Nathan Pralle [EMAIL PROTECTED] wrote: Hi all. I'm curious to hear about other people's HOME usage of Asterisk. Do you have a really neat setup for home use? Fun stuff with VM and/or forwarding and custom scripts? If you are married, the goal is spousal approval. If you can achieve that, bliss Here is my setup PSTN from Bell. This line is connected to a SPA-3K to convert it to SIP and into Asterisk. The line has call forward busy on it from Bell and directed to my Broadvoice line. So when someone is on the PSTN line, it forwards to the Broadvoice line and comes in as a new call. Broadvoice line used to call adjacent counties that would normally be toll. Also, I have a phone number terminated in Rhode Island so my friends and family can call a local number to reach me. VoipJet for outbound long distance. SPA-3k fxs port has replaced all the analog wiring in my house. When you pick up an analog line somewhere, it's on the SPA. All my callerID is logged to a database where it is identified as a call for my wife or myself. This is used when activating a follow-me feature. One cool thing I did was ask the caller what their music preference is when trying to located us. They got to choose from 1. Rock 2. Country 3. Classical 4. RB That is stored in the database so every time they call that is the music they get when trying to located us. People think that is cool. My system has been rock solid. My wife doesn't even know she's using a PBX. We have lots of little bells and whistles setup with our home automation (X10) stuff and lots of little scripts for things. Good luck! -Chuji ___ 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] Large Asterisk Setup (~500 Concurrent Calls + Scalability)
On 4/21/05, Matt Roth [EMAIL PROTECTED] wrote: Daniel, I would be interested to hear if anyone knows of a method to completely offload the Monitor command from the master server. It is the missing piece of the puzzle to optimizing the digital recording process. You might want to talk to the folks at aheeva. www.aheeva.com They built their platform around * very much like you are. They offloaded quite a bit (including recording calls) to other boxes. Now, they built their solution around a much less stable Asterisk build, but they have some great experiences. I talked to them quite a bit last your at Astricon. I do remember them saying that after about 60 concurrent monitor's the * box would get unstable. Looks like you have some good research going you just need a little more proof of concept. -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
Re: [Asterisk-Users] Manipulate Asterisk Database from manager?
On Apr 11, 2005 10:16 AM, Matt [EMAIL PROTECTED] wrote: Hi, Is there anyway to manipulate the asterisk internal database from the manager (the one you can telnet to)? And if so.. how does one do it? (ie for enabling call forwarding, etc) Not that I'm aware of. You can do a 'listcommands' from the manager to verify everything that is avaiable to you. You can use the asterisk -rx dbput from a shell script if that suites your needs. -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
Re: [Asterisk-Users] app_darthvader.c?
On Tue, 29 Mar 2005 11:33:16 +, Andreas Anderson [EMAIL PROTECTED] wrote: Hi, on Alias the badguy(tm) on the phone usually sound like Darth Vader thanks to some cool device from marshall :-) Is something like this possible with asterisk, or, asked a little more generic, can i somehow pipe an rtp-stream to an application via STDOUT and read it back via STDIN? Have you ever tried LPC10 codec? Sounds like Darth Vader to me. -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
Re: [Asterisk-Users] Fail over
On Tue, 29 Mar 2005 09:40:08 -0400, Chris Mason [EMAIL PROTECTED] wrote: No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. The Sipura 3000 does this. That is what I use at home. -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
Re: [Asterisk-Users] Digium T1 Card Questions
On Thu, 24 Mar 2005 20:57:15 -0500, Matt Roth [EMAIL PROTECTED] wrote: I have a couple of questions about Digium's T1 cards, such as the TE410P. Any answers would be greatly appreciated. 1) Do they support standard T1s or are they ISDN-only? Yes they support DS1's and PRI's on the same card. They can be mixed on the TE410 as well. 2) Do you know of anyone offering support for configuring T1s for Digium cards, and if so at what cost? Digium themselves support the configuration of the card if you buy through them. Hope this helps, -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
Re: [Asterisk-Users] Follow-Me Script
On Sun, 20 Mar 2005 16:36:08 -0800, Kerry Garrison [EMAIL PROTECTED] wrote: I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. Kerry, I'm more of a fan of anthm's patch that does this. You need to be running CVS-Head to get it though. http://bugs.digium.com/bug_view_page.php?bug_id=0002905 -Chuji ___ 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] BroadVoice configuration changes for Outbound
On Sat, 5 Mar 2005 12:13:08 -0500 (EST), Dan Weber [EMAIL PROTECTED] wrote: Today, We have added INVITE Authentication. Thanks for the warning. You pissed my wife off. If she can't make calls, she's an unhappy camper. Maybe next time you warn us? Sheesh -Chuji ___ 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] queue_log and exitwithkey
Hello, I am using Asterisk stable and have a question about the queue_log. It seems like in the past (although I can't find my old logs) that the exitwithkey produced a wait time entry. It would seem logical that you would want to track this. Right now it only shows the key they pressed, and the position they were in. I want to know how long they waited before they bailed. Right now I am circumventing this by having the keypress call an AGI that determines this by the epoch and sending it to the log, but it seems like it is much better suited for inside app_queue. Is this by design? Would anyone want this as a feature? It seems like an easy thing to do, I'm just not up to the challenge of doing it. I'm sure that I could find someone who would though. Any ideas? Thanks, Chuji ___ 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] Advanced FollowMe or Forwarding Application Suggestions
On Mon, 28 Feb 2005 15:23:13 -0600 (CST), Kevin LaFata [EMAIL PROTECTED] wrote: There is one final application I've been trying hard to find to replace something we already use with another provider. It's kind of an advanced FollowMe application. (freedomvoice.com) What I was wondering is if there is already something like this available, or if it would be possible to create/simulate something like that using existing applications and/or extensions. Kevin, Well, yes this is already implemented. Only problem is that it's in the CVS Head version right now and not in stable. I don't know if/when it will make it to stable. So you have a couple options... A. Run CVS Head (In a production system this is a little risky) B. Patch stable for the dial arguments. (This is assuming there aren't many dependencies). Ba. if you patch stable for this, I would advise talking to anthm about it. He would probably take some consulting fees for making sure you are safe to do this. C. Wait for it to come out in the stable version D. Hack some kind of funky meetme/call file/AGI/etc to make this work. If you are interested in what's in HEAD, check out this bug where anthm put it in. http://bugs.digium.com/bug_view_page.php?bug_id=0002905 -Chuji ___ 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] MOH clicks
On Mon, 21 Feb 2005 02:42:41 -0600, Anton Krall [EMAIL PROTECTED] wrote: Guys.. Ive noticed that I have 2 mpg123 processes running, is that ok? also... can you make MOH random? Yes, this if fine. Please read the archives. Use google. Use the wiki. Again, on the random, read the samples, use the docs, check the wiki. Come on man, this info is readily available. Also, I dont know if there is a problem with my config but when listening to MOH, every 3 or so second I get a click sound which notices because music gets a hickup every 3 or so seconds... is this ok? Check the version of mpg123 you are using. There are some specifics in the wiki on which version works. Most of us would recommend version Version 0.59r (1999/Jun/15) Otherwise, watch your CLI when in it hiccups. Could be something else going on. We don't mind helping, but it does get old answering very well documented configs/problems. If you are going to do much with Asterisk, you will have to spend a lot of time on the wiki. You won't always get spoon fed. -Chuji ___ 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] * Call Monitoring
On Mon, 21 Feb 2005 08:00:40 -0500, Daniel Corbe [EMAIL PROTECTED] wrote: I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? There aren't any specific tools that do exactly what you want afaik. It wouldn't take much to taylor a few things yourself though. As for the PRI processing calls. You could always drop a call file in from the cron every 10 minutes that makes a call out and back in. Then you you can run a script that looks over your CDR to verify that the call was received. Have it call a specific context or application to look for. As for calls failing this could be a challange. What do you consider failing? You could use something like my-swatch to tail the log file looking for certain patterns. PRI alarms would be an obvious. Might take you a day or so to get these things going, but it would be well worth your time and piece of mind. -Chuji ___ 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] FAX
On Sun, 20 Feb 2005 23:16:00 +0900 (JST), Isamar Maia [EMAIL PROTECTED] wrote: Ok. I will be burned in fire.. :-) Or better.. I won't go to the heaven... You are probably right. But in the the mean time, while you are here on earth, you will probably spend some time in the legal system too. Spam faxing is a punishable offense and enforced per incident. War dialing for fax machines fall under the same category. Spend a little time here before you get too far into the project. http://www.junkfax.org/index.html If you impede someone's ability to get the e911 system by clogging their lines that goes beyond illegal. Find another get rich quick scheme. -Chuji ___ 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] Re: Integration Panasonic PBX
On Tue, 15 Feb 2005 17:12:01 -0500, John Novack [EMAIL PROTECTED] wrote: The real problem with Panasonic and anyone's voice mail is tenant sharing. Calls to VM or returning to VM don't contain the trunk number information. The only way to handle is to dedicate extensions and ports for each company, a real waste of resources. John Novack Does the Panasonic in question support SMDI? If so, maybe someone will finally get around to coding the SMDI interface to *. There was/is a rather large bounty for this. I would be interested in contributing to this if there is still an interest. My Toshiba does SMDI. -Chuji ___ 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: How do I match a D? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
In my case, I need it because it's how my PBX does disconnect notification to the voice mail system. When the line is hung up, it sends a D. I think that my PBX does this too. Is there any way I can get the Zaptel drivers to disconnect on that tone too? I would love to replace my existing voicemail with * but I can't get my PBX to signal a disconnect properly. I have to use busycount=10 but every voicemail has an annoying busy signal tacked onto the end of it. -Chuji ___ 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] Fastagi question
On Tue, 8 Feb 2005 22:09:59 -0800 (PST), Paul Chan [EMAIL PROTECTED] wrote: Hi All, I have a question about Fastagi because I can't get it to work for some reason. Everytime I execute the fastagi command, i get an error: my extensions.conf: .. exten = 1000,1,agi(agi://some_ip_address) .. try this exten = 1000,1,agi(agi://some_ip_address:some_port) Here is the exact line from my extensions.conf I am running on 1.0.5 against a JAGI server as well. exten = 5282,2,agi(agi://10.10.2.250:4573) Hope this helps, -Chuji ___ 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] 911 and Cops knocking on my door
On Wed, 2 Feb 2005 15:51:53 -0700, Andrew Niemantsverdriet
[EMAIL PROTECTED] wrote:
So it looks to me like something else went wrong.
If you took your dial line right from the samples you likely still
have a ${EXTEN:{TRUNKMSD}}
That variable TRUNKMSD is probably stripping off the first digit. If
that is there, you need to get rid of it.
Your dial line should be Dial(${TRUNK}/${EXTEN},30)
-Chuji
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[Asterisk-Users] Volume on Zap channels (T1)
I'm having some problems with the volume when bridging two zap channels together. Here is my config. Asterisk(with TE410) -- PBX (Toshiba) -- PSTN (Via T1) When a call comes from the PSTN, in to the pbx, and into Asterisk it sounds great. When a call is sent from Asterisk to the PSTN (via our pbx) it sounds great. When I try to have a call come in from the PSTN, to the Asterisk server, and then back out with a Dial command (bridging Zap/1 to Zap/24) the volume is terrible. Now, to add confusion to this, multiple calls in and out on these same zap channels to a meetme sound great! I've had 10 calls from the PSTN into the meetme with no problems what so ever. So this semi-proves that there isn't degradation when having multiple Zap channels together. Does anyone have a clue why the actual bridged calls would suffer from volume problems? Thanks, -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
Re: [Asterisk-Users] Damn DTMF Beeps on my calls
On Mon, 24 Jan 2005 15:16:45 -0500, Mark Eissler [EMAIL PROTECTED] wrote: It's a common problem with VOIP and tends to happen when certain voices hit tones that mirror a DTMF soundwave. Some CPE's may be more sensitive and therefore more likely to cause problems in this regard. -mark My SPA-3000 does this constantly. It really pisses my wife off. I've been meaning to email sipura to see if there is anything they can do. This is a really annoying trait as it's generally very load when it happens. -Chuji ___ 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 VPN server for * and woad warriors using windows?
On Sun, 23 Jan 2005 10:33:14 +0100 (CET), Remco Barende [EMAIL PROTECTED] wrote: I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. OK, take a hard look at this before you get too far. Installing VPN software *on* the Asterisk box is not a good idea. Now, you haven't explained the volume of users on the box, or the availability needs of the box, but either way, this is bad practice. The term roadwarriors' makes me think this is for a business. There are numerous vpn server daemons around and I found many messages about some of them using tcp/udp etc and instead of trying them all out hopefully someone can recommend one? If you want IPSec, take a look at OpenWall. If you must run this on your asterisk box, so be it. Now, if I were you, I would take this opportunity to install a good Linux based firewall solution that sits in FRONT of the asterisk server. I can't stress this enough. Take a look at m0n0wall. It has vpn support (ipsec and pptp) built in, and it will run on nearly anything. Put this on a machine by itself. http://m0n0.ch/wall/ (I guess this would make a useful wiki page too). Thanks!! Remco Hope this helps! -Chuji ___ 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] Re: API Call Bridge?
On Fri, 21 Jan 2005 03:24:26 + (GMT), taf taffey [EMAIL PROTECTED] wrote: Is there a way to dial two outbound/external numbers and bridge them together using the Asterisk API manager method instead?? Cheers, Taff. Use the wiki luke! http://www.voip-info.org/wiki-Asterisk+manager+API -Chuji ___ 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] Ring an incoming call in multiple extensions
On Thu, 20 Jan 2005 21:32:37 -0600, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi asterisk users! Here's my issue, I've deleted the s extension cause I don't want any action to be taken on incoming calls as my pbx is for home use, but I would like to ring all my VoIP extensions at the same time the PSTN line rings and to be able to pick up the call in any extension, honestly I don't know if this is possible, some ideas ??? Use the between your dial targets. From the sample config ;exten = 6245,1,Dial(SIP/Grandstream1SIP/Xlite1,20,rtT) wiki page is here http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial Have fun, -Chuji ___ 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 monitoring with Nagios and IAX
On Tue, 18 Jan 2005 14:07:08 +0100, Jens Vagelpohl [EMAIL PROTECTED] wrote: Hi *, Does anyone have a lead on a Nagios plugin that speaks IAX or a small app to do so? I'm trying to set up remote monitoring for my Asterisk server and only IAX2 traffic is allowed through the firewall. I don't believe it has been done yet. There is a small Nagios plugin out there but it doesn't do much. We have a very large Nagios installation and I have been struggling to find time to write some Asterisk plugins. I would be interested in helping out (though my IAX knowledge is limited). Also, RoyK on IRC is very involved in the Nagios and Asterisk communities. I'm sure he could be a resource. -Chuji ___ 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] sipura 3000 mwi stutter problem
On Tue, 18 Jan 2005 13:30:03 -, Chris Stenton [EMAIL PROTECTED] wrote: May be I have fiddled too much with my sipura settings but I can't get it to give the stutter tone when there is a new voice mail waiting on the asterisk box. I can either get a stutter tone all the time or not at all. Anyone got this working. Mine works fine. If you don't get this working let me know and I can save the HTML off of my Sipura. It's at home and I am at work. I can only get to it over lynx right now. -Chuji ___ 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] Connecting Asterisk to a Toshiba Strata system
On Mon, 10 Jan 2005 10:03:58 -0500, David Brodbeck [EMAIL PROTECTED] wrote: I'm interested in setting up Asterisk as a voice mail system for our Toshiba Strata CTX100 PBX. Our current voice mail system connects to the PBX with four analog extensions. Does anyone have information on how to integrate this kind of system with Asterisk? I've looked at the legacy integration section of the Wiki, but it doesn't seem to have anything about Toshiba systems. David, I do this with a Toshiba Strata DK280. The model before the CTX. You shouldn't have any problem doing this with the CTX either. Assuming you don't have SMDI integration with your voicemail (you didn't state that you did), you just need the phones to send trailing digits, and you need to negotiate the lighting of voicemail lights on phones. All manageble from Asterisk. As for hardware, you mention you only need 4 ports now. You could get a tdm400 card, but you would max that out with the first install. Like most of us here who have messed with Asterisk, this is only the *first* application that you have. There will be more, I promise! Do yourself a favor and look at a T100p and an Adit 600 with 8 (or more) FXS ports. You're intial investment will be about $800 bucks, but it will prepare you to do more with the Asterisk-Toshiba integration. Good luck, and do some wiki reading. -Chuji ___ 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 site asterisk installation question
On Thu, 6 Jan 2005 20:15:09 -0500, Ronald Hartmann [EMAIL PROTECTED] wrote: Good Day list, I have a friend who is interested in implementing an asterisk implementation at his offices. I'm assuming you will be doing the implementation or you friend can't post to the list himself? snip The configuration would consist of the following Site A Asterisk Box With 12 incoming lines and 15 phones Extensions 101-115 Site B Asterisk Box With 4 incoming lines and 7 phones Extensions 201-207 Site C Asterisk Box With 4 incoming lines and 6 phones Extensions 301-307 He would like to have this system setup such that anytime a call comes into Site B or Site C, that it is forwarded to an operator group at Site A. Possible. IAX2 is the magic word. When Site A determines that the call needs to go to a sales rep for site B, they want to be able to forward the call to the Queue_SiteB and operate as such. First Ring Agent1, then Agent2 (Both of which are local to SiteB) then Ring Agent3 who is at SiteA, finally Ring Agent4 who is at SiteC. Tricky, but doable.Might be better if you queue timeout and go to SiteC Will Asterisk allow me to have agents to log into ACD Queues on different Boxes? Feasibly agents would be logged into 3 different ACD Queues 1 which is local to them and Yes, agents can log on to multiple queues The other two which are located on a separate box. Finally, does Asterisk have the ability to share dialplans between the boxes? Yes, this is not for the faint at heart though. IAX can do left and right sharing of dialplans. Any feedback as to the feasibility of such a creation would be greatly appreciated. You have a pretty advanced setup here. The queues will take some creativity. Also, you might be able to enlist the help of DUNDI to create your meshed enviornment. PS They need separate Asterisk Boxes at each site incase the connection between sites goes down, they want to be able to operate independently. That's a good plan. Ron Final thought I don't recommend you tackling this on your own at this stage of your asterisk knowledge. This sounds like a pretty critical production enviornment that needs a properly thought out implementation. You will probably want to look for a seasoned * consultant to help you with this design. Take a look on the wiki for asterisk consultants in your area. -Chuji ___ 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_log
On Tue, 4 Jan 2005 12:19:31 -0500, John Bittner [EMAIL PROTECTED] wrote: Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net John, I wrote a little perl routine that monitors the log and sends it to MySQL as it's written. Now, my perl skills are very weak. The app works fine and we have been running it in production for months, but keep in mind it was written by a total non-programmer type (me). If you don't find what you are looking for, I would be glad to send this to you. Just let me know. -Chuji ___ 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] phones with two ethernet ports
On Sun, 2 Jan 2005 16:35:12 -0500, Erick Perez [EMAIL PROTECTED] wrote: Hi there, what phones are available that have two ethernet ports? Check out the Polycoms. IP500 and IP600. -Chuji ___ 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] Call Queue Question
On Sun, 2 Jan 2005 21:48:47 -0500, Glenn Dalgliesh [EMAIL PROTECTED] wrote: When I first started looking at a similar problem started out on the same path with app_queue but even having access to a friend of mine who actually help write some of the Queue code we decided it wasn't the right tool for the job. I agree with that. For a much simpliar solution, try anthm's recent patch in CVS. http://bugs.digium.com/bug_view_page.php?bug_id=0002905 I use this with much success. -Chuji ___ 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] Final call for departments
On Wed, 29 Dec 2004 01:51:16 -0800, Alspach Family [EMAIL PROTECTED] wrote: I am getting ready to submit a list of department names to be recorded. This is what I have so far: QA or Quality Assurance. -Chuji ___ 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] Matching Caller ID against a database of known callers
On Wed, 22 Dec 2004 17:12:56 +1100, Eric Bishop [EMAIL PROTECTED] wrote: Hi All, Is it possible to match caller ID on incoming calls against say text file of know numbers and diaplay the name rather than the numerical caller ID? Eric, Check out the following. http://www.voip-info.org/wiki-Asterisk+cmd+LookupCIDName and http://www.voip-info.org/wiki-Asterisk+tips+managing+CID+names Hope that helps, -Chuji ___ 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] Toshiba DK-40 and Asterisk...possible?
On Mon, 20 Dec 2004 17:08:55 -0500, Kanwar Ranbir Sandhu [EMAIL PROTECTED] wrote: They currently have a Toshiba DK-40 and 19 DKT 2010-SD phones. The owner has told me that the phones are digital (not in the VoIP sense, obviously). Kanwar, I have a Toshiba DK 280 that is connected to our Asterisk server here. We have it connected in two ways right now. A T1 card going from the Toshiba to a T100p on the * box. And we also have analog extensions from the Toshiba going into a Adit 600. The analog extensions are for doing transfers from the Toshiba to the * box. Unfortunately, Tie lines on the toshiba do not support unsupervised trunk to trunk transfers. Anyway, you will definitely be able to integrate the two, it just might take a little effort. -Chuji ___ 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] Bugtracker Karma Hall Of Fame
On Wed, 15 Dec 2004 17:25:59 -0800, Paul Crick [EMAIL PROTECTED] wrote: The Karma Hall Of Fame is now available at: http://bugs.digium.com/karma_halloffame.php Well the fact that Kram sits at +14 tells me this system if flawed. Not that I'm unappreciative of Anthony's work, but come on 14? -Chuji ___ 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] Queueueueuueue position
On Thu, 16 Dec 2004 15:18:10 +0100, E. Versaevel [EMAIL PROTECTED] wrote: When I call in (with an agent logged in) I get to hear the MOH on the client side, hover no matter how high the hold time is, I NEVER get an announcement over my queue position or my estimated wait time? Both the incoming call and the agent are on SIP channels. What is wrong ? Kind regards, E. Versaevel Would that be because this is the only call in queue? Try putting another call in queue and see what you get. -Chuji ___ 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] XML to monitor queues on Cisco display ?
On Sat, 4 Dec 2004 13:08:19 -0600, Joe Dennick [EMAIL PROTECTED] wrote: I, too would be very interested in this application. We are also building an application to handle this. The desktop app is built in Java and will have a java proxy component (running in websphere) that talks to the manager. We are probably 3 weeks away from putting anything usable out there, but I would be glad to give back once it does. BTW, ours also has screen pop functionality so that it calls our vertical software package. I've also created a perl app that reads the queue.log and pipes all that info in to SQL server. -Chuji ___ 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 dabbling...
On Sat, 4 Dec 2004 14:12:04 -0800 (PST), Ray Jender [EMAIL PROTECTED] wrote: Newbee here Ray, You should be fine with your setup. BSD can be a little finicky to get working sometimes, but if you're familiar enough with it you will be OK. I have a P133 w/ 128mb ram running my home * box and I don't have any problems with it. My wife doesn't even complain. For dialtone checkout any of the following. Nufone, Voicepulse connect, broadvoice, voipjet. All of them have varying strengths. You will be able to connect to any of them over your broadband. Cuddle up to the wiki for a while. There is more information there than you could possibly need. Asterisk is an adventure. Hope you're not busy for the next couple months! -Chuji ___ 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] Broadvoice outbound 404 error
On Sat, 4 Dec 2004 17:22:38 -0500, Reid Forrest [EMAIL PROTECTED] wrote: Is anyone else experiencing 404 errors on outbound dial with Broadvoice? I've followed the instructions posted by Broadvoice to configure sip.conf, and inbound calling works fine. Every time I try to dial out, I get a 404 Not Found error. [bv-home] type=peer host=proxy.dca.broadvoice.com Change the above line to host=sip.broadvoice.com Give that a try. -Chuji ___ 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] Using Pocket PC over cell phone connection?
On Sat, 04 Dec 2004 14:25:25 -0700, Paul Fielding [EMAIL PROTECTED] wrote: Anyone tried using a pocket pc with sjphone or x-ten over a cell phone connection? Uhh, good luck. Latency, lack of bandwidth... Nice idea, but I would stick with the cell phone when you're on the road. -Chuji ___ 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] Ring all Configured Extension
On Thu, 2 Dec 2004 10:56:06 -0600, Eric Rees [EMAIL PROTECTED] wrote: I was afraid that someone would suggest that. Check out app_queue then. -Chuji ___ 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] SPA-2000 Dropped calls
On Wed, 01 Dec 2004 16:01:16 -0800, Mike Benoit [EMAIL PROTECTED] wrote: Don't run LISa on the same network as any SPA-2000 or SPA-3000. (maybe even any Sipura device?) I have a problem with mine locking up, but not while talking. When it sits idle for a period of time I come back to it and it's dead. No ping, nothing. I have to unplug the power to get it back to life. Could this be symtompatic of the lisa thing? I do have lisa on a lot of my *nix boxes, but none of them appear to have it running (ala ps -aux |grep lisa). -Chuji ___ 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] Exiting out of Voicemail with a '0'
I'm having a problem getting this feature to work. I've tried this both on latest CVS Head and on stable. In the same context that I send the call to voicemail I have this. exten = o,1,Congestion(2) exten = o,2,Hangup() When I'm in voicemail and I press zero, I get Playing 'vm-sorry' (language 'en') in the CLI and then hangup. If I change the 'o' target to an 'a', and then press * during the voicemail, it does exactly what I expect it to do. If I remove the 'o' target out all together, zero doesn't do anything at all. This all tells me that I'm putting it in the correct place in the context. Why is it not working for me? Any ideas? ___ 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 - ACD.
On Fri, 26 Nov 2004 13:56:20 -0200, Jefferson Carvalho [EMAIL PROTECTED] wrote: Hello list , Why not create call groups in sip.conf ?! We have this on Zap Channels !!! For example : We have XYW users ( sip1 , sip2 .. etc ) = g3 When we create an extension we will refer to that group . :) exten = 100,Dial(SIP/g3 ... ) :) Hey there, Check out the appliction queue. There is a lot of good information on the wiki for you. It will do what you are wanting and more. http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Queue -Chuji ___ 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] multiple arguments in system
On Fri, 26 Nov 2004 13:48:58 -0600, James Taylor [EMAIL PROTECTED] wrote: I'm trying to: exten = 5551212,1,Answer exten = 5551212,2,System(/usr/local/snpp.pl,-p Xpage -m some_text -r 8005551212) What is that comma doing after .pl? That will likely get replaced by a pipe. Take a look at the CLI and see the shell command it's running. If it is the comma, surrout it in single quotes. -Chuji ___ 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 broadvoice outbound plz... ;)
On Fri, 26 Nov 2004 22:25:56 -0600 (CST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: *sigh* [broadvoice] type=peer nat=yes username= (phone number) secret=secret disallow=all allow=ulaw dtmfmode=inband fromuser=xx fromdomain=sip.broadvoice.com ;host=147.135.0.128 canreinvite=no insecure=very srvlookup=yes context=broadvoice Kevin, I would change it took like so [broadvoice] type=peer nat=yes username= (phone number) secret=secret disallow=all allow=ulaw dtmfmode=inband fromuser=xx fromdomain=sip.broadvoice.com host=sip.broadvoice.com canreinvite=no insecure=very ;srvlookup=yes context=broadvoice Are you really behind a nat? If not, lose that too. Hope this helps. -Chuji ___ 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 Phones-Receptionist Setup
On Sat, 20 Nov 2004 15:58:48 -0800, Tracy R Reed [EMAIL PROTECTED] wrote: I proposed something like this to a client but the receptionist has other duties for her computer and does not want to have to have the operator panel up all the time or go searching for the window in the taskbar every time a call comes in. Nor does she want another computer on her desk dedicated to just this. People don't like to change and to achieve maximum success * needs to be able to replicate what the client is accustomed to. I would look at putting a dual monitor on her desk. You can pick up a 15 flat panel and a video card for about the same cost as the SNOM. Not to mention, you get quite a bit more benifite from the FOP controls than you do busy lamp fields. It's a a new era here folks. Asterisk is not your dad's pbx. -Chuji ___ 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] Answer confirmation on non-Zap channels?
On Fri, 19 Nov 2004 18:02:51 -0600, Brian West [EMAIL PROTECTED] wrote: http://bugs.digium.com/bug_view_page.php?bug_id=0002905 bkw Sexy I've been waiting for this. As always, thanks Anthm! -Chuji ___ 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] Call ID Mini-Popup?
On Wed, 17 Nov 2004 17:05:55 -0300, Thomas Hutton [EMAIL PROTECTED] wrote: Question: Does anyone know of a lightweight popup method to put an incoming call ID string on a client machine? Something as simple as winpopup would work great- for example: I have a call coming in on Zap/4 but the phone on Zap/4 doesn't have a call ID display. Could I somehow configure Asterisk to call a script that uses a SMB winpopup (or other method) out to a specified computer sitting next to the phone? Thanks very much for any ideas, or knowledge of something already in existence. I use a program called YAC. http://sunflowerhead.com/software/yac/ The client basically sits on a listening socket. I run a little perl AGI that streams the CID to the YAC client. I also have this running on my Tivo so that I can get callerID on my TV. All of the vb source is available. One thing I have been meaning to do is rewrite it to accept a UDP broadcast on my whole subnet so that I don't have to specify indidual IP addresses in my Perl AGI. Hope this helps. If anyone tweeks it better, let me know. I just haven't had time. -Chuji ___ 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] Directory app and extension
On Mon, 01 Nov 2004 16:21:46 -0500, David Filion [EMAIL PROTECTED] wrote: So, the question is does anyone know of a way to get the extension number when the dial plan context is entered via Directory(), and if so how? David Filion David, You could always right your own directory application (AGI if you want) that reads from Postgres. It would be pretty easy. -Chuji ___ 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] $AGI-stream_file
On Fri, 29 Oct 2004 16:59:56 -0400, Victor Cartes [EMAIL PROTECTED] wrote: Hello everybody! I've got a problem here. I writing an AGI in Perl and when I used the stream_file method It did not work. Then I realized that the next line has no waited for the streamed file end, the program has just gone on. Victor, Make sure you get a good readparse of all the variables sent in first. If you don't do this, you could get some wacky behavior. i.e. something to the effect of my %stuff = $AGI-ReadParse; That will fill the array stuff will all of the call variables. Hope this helps, -Chuji ___ 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] Digium TheVoice recordings' sound terrible
However, the format the customer ordered was WAV, whereas all the included recordings are of course GSM. Has anybody had similar experiences? I tried to convert the WAV files to GSM using sox but since I don't know what parameters are best in this case, the results weren't satisfactory. Any suggestions? Benjamin, Don't know if this helps you or not, but this is taken right from Jtodd's wiki page. **my disclaimer** Might want to backup your sounds before doing this though. #!/bin/sh tmpfile=/tmp/rescale$$.wav for i in *.wav; do scale=$(sox $i /tmp/foo.wav stat -v 21) if [ $scale != 1.000 ]; then echo -n Rescale $i... cp $i $tmpfile sox $tmpfile -v $scale $i echo fi done The wiki page is at http://www.voip-info.org/tiki-index.php?page=Asterisk%20sound%20files Hope this helps, -Chuji ___ 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] cheap gig switch? smc, netgear, or 3com?
On Thu, 21 Oct 2004 09:22:46 -0500, Jay Milk [EMAIL PROTECTED] wrote: Afaik, they use Linksys for their OEM. So, however you trust linksys, that should be par for the course on DELL. That said, do not expect usable support from DELL. We have purchased in the neighborhood of 5,000 computers from them in the last four years, and while most of them perform well, when there IS a problem, the friendly population of Bangledesh just isn't very helpful in getting things up and running. Sheesh, 5000 computers and you don't get silver or gold support? You have been doing business with the wrong account team. With our gold support I have a dedicated team of engineers to support our company. We call directly to their desks and talk to the same guys all the time. You might want to talk to your rep. For those of you who have been through the offshore support offerings, you might appreciate this. http://www.bordergatewayprotocol.net/jon/humor/web_animations/india_tech_support.swf -Chuji ___ 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] cheap gig switch? smc, netgear, or 3com?
Bleh, what group said this? The 3Com group? Dell is the WalMart of the hardware world. Their pricing is better because they build efficiencies. I have 33XX 54XX and I just bought my first 6024 Layer 3 QOS ready switch. These things are nothing but Ciscos in sheep's clothing. They have been rock solid for me and others that I know who use them. -Chuji On Wed, 20 Oct 2004 14:47:21 -0600, Matt Hess [EMAIL PROTECTED] wrote: Remember, you pay for what you get.. especially with Dell networking equipment. I have heard about several groups who tried the dell switches only to give up on them because the dell switches just didn't perform. Yes, price-wise they look good.. but as far as performance goes.. (that is assuming you want high/solid performance) I'd look elsewhere. ___ 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] Sample advanced call routing standard extension
On Fri, 15 Oct 2004 13:26:12 -0500, Eric Wieling [EMAIL PROTECTED] wrote: [default] ; ; Eric Wieling Eric, Great stuff! I wish more people would post their configs. A lot can be learned from examples. Maybe find a home on the wiki for this! -Chuji ___ 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] Seeking a VoIP Solution for a big company
Knowing that we are decided to make the move to VoIP, can somebody tells me the feasibility of deploying such a solution in an environment that has the following technical requirements: - 250 Users for the Headquarter (100 Mb LAN) - Around 50 remote sites ( WAN Technology: Leased lines/ISDN/VPNADSL/Wireless) - Unified messaging - Small call center (10 users) - CTI Applications - Interoperability with the existing carriers ( Phone companies/ 64 lines) - Security Asterisk can handle all of this. Not all of it is canned (i.e. CTI applications) but all very possible with a little effort. You might want to look in the wiki for for consultants in your aread http://www.voip-info.org/tiki-index.php?page=Asterisk%20consultants as this would give you a head start. That is a pretty big implementation to take on without any experience. Hope this helps, -Chuji ___ 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] Astricon
Be carefull with assumptions... some of these arrangements require the web browser access to open the channel again during that 24 hour period. I was at one last week, and used a hub in the room thinking I could use a snom 200 for making calls. Didn't work. PC worked fine has long as I started with the web browser, then x-lite would work; but the snom never did. (The snom has been used in lots of hotel rooms around the country, and I'm quit comfortable with its ability to handle nating, firewalls, etc.) I wouldn't make any assumptions relative to sip and iax2 though. Rich These services often use MAC level ACL's. In order to effectively do this, you must start out behind a NATted connection. So if you are going to use multiple boxes in your room start with a router of choice so that the Hotel system sees that MAC address. Then you will be NATted behind it. Creates some SIP difficulties, but this is a better method. -Brian ___ 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 newbie questions
On Sat, 11 Sep 2004 10:01:27 -0400, John Stegenga [EMAIL PROTECTED] wrote: yesterday, and since I'm only getting Digest I figured I'd see a response in a day... [sarcasm off] Some people may have a filter in their inbox that has newbie in it going directly to trash. Just kidding, it's been a busy year! Hi everyone. I'm a bit of a Linux newbie, but I've been doing tech stuff for ages. I'm also brand new to *. This is a dangerous combination. Asterisk/Linux isn't point and click, and it takes a lot of effort to get running and keep running in a production environment. Everyone has to start somewhere, but just know that you have a long road ahead of you with your limited knowledge. I've been reading the Voip.org wiki, and perusing the list archives for a while since I've been asked to investigate using IP telephone / soft phones for a call-center type scenario. GREAT START! The Wiki is your friend. Also, try IRC. You won't always get questions answered right away, but it's a great place to lurk and learn. People (marketing folks) have pointed me at Cisco, but I really don't wanna. I'd rather be the hero and pull this off with a much smaller budget. Very possible, but just remember, you won't have the ability to call 1800 - Go Cisco if things break. Here is a scenario - 40 person call center, all with PC's (windows) and soft-phone. -any recommendations on hardware to run *? soft phones? 90% of calls would be IP / IAX coming to the center. Hardware - Don't go too cheap here. Maybe grab a Dell server (entry level P3 or Xeon). Depending on your expansion needs, be careful getting a 1u with 1 PCI slot. Software - Take a look at XTen. It's easy to use. It's a freeware, and an inexpensive buyware. SJPhone is another one for you to check out. I read in the list archives about an ACD application / extension to * that would probably to what I need in that regard. - thoughts? Are you speaking of ICD? If so, I don't know that it would be necessary for what it sounds you are doing. You should be fine doing all of your ACD inside of *. In remote locations I would also run *, and hook it up to an extension on an existing PBX. Excuse the complete newbie question, but how many 'wires' do I need to bring between the PBX and the * box to support multiple simultaneous calls? These calls would come from any extension on the TDM pbx to asterisk to the call center. This depends on your PBX and how many calls you will want to have. If it's 8 or fewer, you should probably go with analog extensions off of your PBX. That will require Analog ports (and cards possibly) and Digium card(s) to connect to them. Or, if it's cost effective, you can get a T1 or PRI card in your PBX and interface that to *. Either directly, or through a channel bank. This all varies depending on your needs and what's available to you on the PBX. How would / could? one configure * at the remote location to communicate with * at the call center? IAX between two * servers. You can even share the dialplans (a bit challenging for a newbie). How would / could? one configure * at the remote location to use the existing TDM PBX as failover to call the support center via 1-800 if the IP circuit died? This would be done with solution I spoke of above. I know you're all banging your heads on your desks saying OY! another newbie. Thanks in advance for your wisdom and guidance. John John - I'm relatively new myself. I came into this with loads of Linux and teleco experience so my path is a little easier. What you really need is a desire to achieve your goals, and a lot of patience. If you have that, and a lot of time, you can pick this up and be successful with it. If you are in a hurry though, I advise you to head back to the WIKI and possibly look for a consultant to get you a head start. They are relatively inexpensive and can get you started in the right direction. Hope this helps, and good luck on your journey. ___ 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] Legacy Toshiba Phones
Not necessarily so. Recently I discovered that Artisoft's Televantage Soft PBX can support Toshiba Strata CS digital phones (DKT 2000 and 3000) through a PCI 16-port digital station card (Toshiba part #CS-DKTU-TV). Apparently, the Strata CS is an OEM licensed version of Televantage. It would be quite cool if an Asterisk driver can be developed for the 16-port digital station card. I'm a Toshiba Strata user too. My asterisk is interfaced over T1 doing EM Tie lines. I would love to be able to use the phones on * if we every make the leap. We have 150+ extensions though. -Brian ___ 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] users
Here you go Travis. Just about everything you need on the wiki. http://www.voip-info.org/wiki-Asterisk+non-root -Brian - Original Message - From: Travis Conway [EMAIL PROTECTED] Date: Thu, 5 Aug 2004 18:38:54 -0500 Subject: [Asterisk-Users] users To: [EMAIL PROTECTED] Hello Guys, I just setup an Asterisk server here at work and have just a question that I was hoping you could help me with. How do I run asterisk as a user other than root? It seems that if I try to start it as a user I created it doesn't actually start. -- Travis Conway [EMAIL PROTECTED] FWD: 414668 +1 334 220-7519 (T-Mobile) ___ 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] broadvoice/asterisk
Jeff, Thanks for jumping in and working with the Asterisk community directly. Hopefully people (myself included) will recognize the effort you guys put in to make sure that Asterisk users are happy. I was nearly driven to another service due to the outage, but I'll stick with you guys as long as there is support. Thanks, -Brian On Tue, 27 Jul 2004 03:24:22 -0400, James Jones [EMAIL PROTECTED] wrote: Ok we have found a better solution. Put everthing back the way it was and make sure that you have this line in your general section of you sip.conf file: srvlookup=yes We have added a SRV entry in the correct place now. So everyrthing should go the correct servers. -james ___ 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] broadvoice/asterisk
James, Sorry, was reading your reply about Jeff and misquoted the name. Anyway, thanks again! -Brian On Tue, 27 Jul 2004 07:50:24 -0500, Brian Roy [EMAIL PROTECTED] wrote: Jeff, ___ 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] Broadvoice problems again
I am having issues also. I called them and I was told that they are doing upgrades on their network. Zac Upgrading their network with all of this downtime? That is pretty pathetic for a VoipCo. I've considered going to VoicePulse. Maybe this will help me move. Mine has been down for over 12 hours now too. -Brian ___ 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
