Re: [asterisk-users] EVERY toll free number appears to be in e164.org??
The way that I understood this to work was that e164.org lists all toll-free numbers to make it free to call those kinds of numbers (instead of using one of your own trunks). Since ENUM can provide priorities, if I own a toll-free and enter it into the system, the route that I specify will be returned with a lower priority than the default e164.org route. Anyone who doesn't list their number will have the default e164.org route provided if someone queries for that number. Perhaps troublesome that the system is opt-out instead of opt-in, but at least you have the ability to override the provided route for toll-free numbers that you own. Also, they do list on their FAQ page that they list routes for toll-free patterns for several countries (and they state that the call makes the provider money - there is no mention that e164.org makes money off the call themselves). Cheers, AR -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: July-24-09 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] EVERY toll free number appears to be in e164.org?? ENUM lookups at e164.org return a IP route for ALL toll-free numbers. I was surprised to observe that ALL toll-free numbers get a hit at e164.org. It appears that ALL toll-free prefixes have been delegated, thereby publishing an IP route for YOUR TOLL-FREE NUMBERS, my toll-free numbers, and even toll-free numbers that have not been allocated. :-) See below Should I care? Even though this whole thing doesn't directly cost me anything, it seems like bad netizenship to be publishing circuitous routes to others' numbers. Their route may be inferior; less direct (higher latency), and may also introduce more transport over best-effort networks (artifacts and jitter). Is this a reasonable thing for e164.org to do? Based on what I see at e164.org, one might make the reasonable assumption that ALL routes are published after a token test that the routes are sanctioned by the 'owners'. Perhaps their test is not perfect test, but you are free to do what you want with that information. There is however no indication (that I can find) that ENUM lookups for toll-free numbers are actually what I might call 'bandit routes', leading ITSP's and PBX admins to make routing decisions they might not otherwise make. Perhaps CHASE BANK or MAYO CLINIC should be concerned when end-user callers (via their office PBX, or ITSP) unknowingly send their media ( secrets) through a stranger's gateway. Maybe Coca-Cola should be worried that a poor quality call could be attributed to them, their brand. Am I correct in assuming that this basically a toll-free revenue share model to fund e164.org?? If so, it seems to me should be disclosed at e164.org, perhaps right next to where they ask for donations. Can anybody speak to this? -Karl p.s. for example try: ast-chi43*CLI !dig 0.0.0.0.0.0.0.6.6.8.1.e164.org ANY Naturally this number on the PSTN returns a reorder tone. The e164.org route actually rings instead of giving a reorder, answers and plays 'dead air' forever if you let it. All resolve to one of the same three gateways: tf.voipmich.com sip.tollfreegateway.com tollfree.sip-happens.com ;; QUESTION SECTION: ;3.5.3.6.4.1.3.7.7.8.1.e164.org.IN ANY ;; ANSWER SECTION: 3.5.3.6.4.1.3.7.7.8.1.e164.org. 60 IN NAPTR 200 10 u E2U+SIP !^\\+1877(.*)$!sip:1877...@tf.voipmich.com! . 3.5.3.6.4.1.3.7.7.8.1.e164.org. 60 IN NAPTR 200 10 u E2U+SIP !^\\+1877(.*)$!sip:1641641877...@sip.tollfreegateway.com! . 3.5.3.6.4.1.3.7.7.8.1.e164.org. 60 IN NAPTR 200 10 u E2U+SIP !^\\+1877(.*)$!sip:1641641877...@tollfree.sip-happens.com! . ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ghost ??
What are your trunks like? If there are any analog lines there's the possibility of crossed lines or interference. Not with digital trunks though... Are you able to hear what the third person in the call is saying, or is it just noise? -- Alex Robar alex.ro...@gmail.com On Tue, May 19, 2009 at 2:16 PM, David @ULC ucoms2...@gmail.com wrote: We are using asterisk and sometime when our guys are on call , they hear some voice of person and amazingly that person is NOT from our center. Any one faced this kind of thing ? ___ -- 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] Newbie trying to make calls outside via digium card and POTS line
On Mon, Mar 30, 2009 at 5:16 PM, Bruce Thayre br...@mipscomputation.comwrote: Up to this point, all i have set up are two SIP phones, my POTS phone, and 1 ring group. My POTS line is connected to channel 1, and my POTS phone is connected on channel 3. Perhaps my understanding of how the calls are handled is flawed, but it seems to me that: 1. I dial a number on my POTS phone 2. Using the number, asterisk should match it against the dialing rules i have set 3. Having matched the number to an outbound dialing rule, it routes the call to the outside trunk and bingo bango i'm talking on the phone with someone outside my office However in this situation, it doesn't seem to work. And lines like [18585300...@from-internal:6] Congestion(Zap/3-1, 20) in new stack are a mystery to me. If any additional information is needed just let me know what, and i'll post it. Any help would be greatly appreciated as i'm kind of stuck on at this point. Thanks http://lists.digium.com/mailman/listinfo/asterisk-users Hi Bruce, I can't be sure without looking at your dialplan, but based on your description it looks like you are routing calls out the wrong port. Asterisk is trying to dial 1-858-530-0400 on port 3 of your Digium card. You've stated that your POTS line is plugged into port 1, so there's likely an error in your dial command. Do you have Dial(ZAP/3-1... instead of Dial(ZAP/1-1... ? AR -- Alex Robar alex.ro...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CentOS and BAT File
Linux incorporates this functionality using shell scripts. You're likely using BASH as your shell, check out this link for a decent BASH scripting tutorial: http://tldp.org/HOWTO/Bash-Prog-Intro-HOWTO.html -- Alex Robar alex.ro...@gmail.com On Sun, Jan 25, 2009 at 11:34 AM, David @ULC ucoms2...@gmail.com wrote: In windows, we use BAT file to execute few series of command , which help us in not writing each command manually everytime we want to execute those commands. In CentOS, I want to do the same thing. Any Advice ? ___ -- 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] IAX2 client for eee pc 1000
Hi Joseph, Not directly related to your question (it's more an answer for the something better part of your plan), but I've loaded Ubuntu onto my Asus eeePC 4G Surf, and I've found that ZoIPer works pretty well. Cheers, AR -- Alex Robar [EMAIL PROTECTED] On Sat, Nov 15, 2008 at 5:49 PM, Joseph [EMAIL PROTECTED] wrote: What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)? I'll eventually replace this crippled Linux with something better but I don't time to play around with it as most divers and modules are still too new and not fully available in all distros. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] IAX2 client for eee pc 1000
I installed eeeUbuntu ( http://www.ubuntu-eee.com/ ). I followed instructions on their wiki to create the USB-stick installer from another Ubuntu PC that I have. It looks like they've made the process even easier now, with a GUI-based application that will prep a USB stick with any ISO that you choose. Once you've got Ubuntu running, the 1000 series should have more than enough horsepower to run any number of great IAX2 softphones. As I mentioned, I'm running ZoIPer, but it's not the most light or stable application out there. As David suggested, KIAX2 is pretty good. If SIP works as an alternative, there's a plethora of other great applications. -- Alex Robar [EMAIL PROTECTED] On Sat, Nov 15, 2008 at 8:13 PM, Joseph [EMAIL PROTECTED] wrote: Well, I've tried to find Ubuntu but so fare I'm not sure which one. I have eee pc 1000 (one with 40GB SSD so plenty of room for any modern distro. Which ubuntu did you loaded? It has to be something that loads onto USB bootable stick (and not through Windows as I don't have one). -- #Joseph GPG KeyID: ED0E1FB7 On 11/15/08 19:01, Alex Robar wrote: Hi Joseph, Not directly related to your question (it's more an answer for the something better part of your plan), but I've loaded Ubuntu onto my Asus eeePC 4G Surf, and I've found that ZoIPer works pretty well. Cheers, AR -- Alex Robar [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to notify an event to every user
Hi Olivier, What type of handsets are you using in-house? I ask because there are a bunch of handsets that allow paging/broadcasting through their speakerphone mechanisms. This could possibly work in your scenario, even if all handsets don't support paging (it would generally be loud enough to hear, depending on the size of the office). Cheers, AR -- -- Alex Robar [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FWD $30 membership-fee
I stand corrected, I finally received a few of these yesterday. They're not unclear about the process; The $30 yearly charge is mandatory. The messages do state that you can link as many accounts to one payment as you'd like though. -- Alex Robar [EMAIL PROTECTED] On Thu, Aug 7, 2008 at 3:05 PM, Alex Robar [EMAIL PROTECTED] wrote: FWD has had paid membership options for years. The paid memberships help to improve the network and increase it's reach. As far as I've heard (and as far as the site mentions), paid membership is not a requirement. That would sort of go against the talk... for free... for good slogan. AR -- Alex Robar [EMAIL PROTECTED] On Thu, Aug 7, 2008 at 2:48 PM, SIP [EMAIL PROTECTED] wrote: From what I can ascertain, this is a way to essentially fund Jeff Pulver's political agenda. I remember writing something a couple of years back ( http://neil.ideasip.com/2006/03/08/von-coalition-and-the-ideals-of-the-little-guy/ ) about how the VON Coalition, which is meant to be a political action committee to help foster new communications, has a somewhat high barrier to entry (minimum $10,000 per year). As far as I can tell, this FWD membership is a less expensive way for people to put their money behind a similar agenda (well... okay, Jeff's agenda, whatever that may be). The only real issue I see with it is that, a political action committee is a committee. The FWD membership seems a little less transparent. It could very well be a way to fund Jeff Pulver's personal vision. While he's done some great things in the community, I still feel awkward with the idea of funding the whole One man. One voice. One decision. No oversight idea. I'm eager to see how it pans out, though. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Semi-OT: ServerBeach for VoIP
Les.net hosts a significant chunk of their services in a few of the ServerBeach data centers. I've had great quality with Les.net. ServerBeach picked Les as their Geek of the Week last year: http://www.serverbeach.com/aboutus/geek_of_the_week.php?id=8year=2007 . -- Alex Robar [EMAIL PROTECTED] On Fri, Aug 8, 2008 at 1:31 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: Hello, I'm looking at getting a dedicated server from ServerBeach to host some light Asterisk/VoIP/SIP stuff. Has anyone used them for this before? I'm pretty sure I've heard good things (in general) about them but VoIP is a very different animal than web hosting - especially for the network (obviously). ServerBeach uses the Peer1 network which looks pretty good. In fact that's how I found out about them in the first place. Or maybe I can do better than ServerBeach? Does anyone know of a dedicated hosting provider that meets the following specs: - Multiple physical datacenters available by request - Well peered network with multiple Tier 1's (Level3, ATT, Qwest, Verizon Biz, etc) - Dedicated servers running Linux (preferably CentOS) Ideally I'd like to be at $150/mo or less. Bandwidth/peering is important but transfer isn't really an issue - SIP/RTP is just a bunch of small packets! :) Other hardware specs don't matter much either. I'd rather have a Pentium 2 running on an awesome network than have an Athlon 5000 with nothing but dirty bandwidth. Any ideas? -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FWD $30 membership-fee
FWD has had paid membership options for years. The paid memberships help to improve the network and increase it's reach. As far as I've heard (and as far as the site mentions), paid membership is not a requirement. That would sort of go against the talk... for free... for good slogan. AR -- Alex Robar [EMAIL PROTECTED] On Thu, Aug 7, 2008 at 2:48 PM, SIP [EMAIL PROTECTED] wrote: From what I can ascertain, this is a way to essentially fund Jeff Pulver's political agenda. I remember writing something a couple of years back ( http://neil.ideasip.com/2006/03/08/von-coalition-and-the-ideals-of-the-little-guy/ ) about how the VON Coalition, which is meant to be a political action committee to help foster new communications, has a somewhat high barrier to entry (minimum $10,000 per year). As far as I can tell, this FWD membership is a less expensive way for people to put their money behind a similar agenda (well... okay, Jeff's agenda, whatever that may be). The only real issue I see with it is that, a political action committee is a committee. The FWD membership seems a little less transparent. It could very well be a way to fund Jeff Pulver's personal vision. While he's done some great things in the community, I still feel awkward with the idea of funding the whole One man. One voice. One decision. No oversight idea. I'm eager to see how it pans out, though. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 transfer feature
Adam, We have the exact same issue occurring on one of our networks. I haven't had time to dive into it much, but here is what I've found out: Calling from a softphone via IAX2 or SIP, transfers work fine. Calling from a Polycom 501 outside the network, transfers work fine. However, the Polycom I used to test outside the network was a Rev. E, the ones inside the network are Rev. C's, so I'm not sure this is a valid test. The configs that these phones are using are identical to the configs used by our other networks, and transfers work fine elsewhere. I haven't had time to put a Rev. E inside the network and test that, but that's my next step. Beyond that, it would have to be something with regards to DNS, routing and how the re-invite works (although the Asterisk full log shows no mention of that type of problem). If you find more details or get anywhere close to a resolution, please post back and let us know. Alex -- Alex Robar [EMAIL PROTECTED] On Thu, Jul 17, 2008 at 5:05 PM, Adam Moffett [EMAIL PROTECTED] wrote: Thanks for responding Kate. I do have a transfer button on the phone, and I follow the transfer process as described in the user's guide. When I press transfer the first caller is placed on hold and then I call the party I want to transfer to. At this point I'm supposed to press transfer again to connect the two parties together. Instead absolutely nothing happens, I can still press cancel to return to the first caller, but that's it. We have 3 of these phones and it used to work on all 3 of them. At some point we noticed it wasn't working any more on any of them but I'm not sure what changed. Any ideas? Thanks, Adam I think it should work standard (i.e. no special setup) Do you have a transfer button on the phone? Kate Adam Moffett wrote: I can't transfer calls with my polycom 501's. Do I need to set up something in particular in the asterisk dialplan to make the feature work? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tests in VMWare
If you leave all of the modules enabled, which one does it have a problem with? You should be able to run asterisk -vc to see where it stops loading. The last line or so should give you the module that it tried to load before it failed. Based on the last time I tried to install under Ubuntu, you're probably failing to load the Zap module. Since you're in a VM and it's unlikely that you're using Zap for anything, you can disable chan_zap.so and see if your Asterisk starts properly then. Cheers, AR On Sun, 2008-03-30 at 20:50 -0400, Ein Bielaczyc wrote: I'm just wondering if any one else has tried to successfully install Asterisk on Ubuntu inside VM. I've installed Ubuntu without incident or error. Even the install of Asterisk is relatively straightforward as it is maintained in one of the repositories. But when I attempt to start Asterisk I get a nice Segmentation Fault. I've narrowed down the problem somewhat. If I disable modules from automatically loading in modules.conf, e.g. autoload=no, Asterisk will start. If I keep the default, autoload=yes, Asterisk fails to start (seg fault). I can't find in any of the other config files where Asterisk may be trying to load a module and therefore crashing the system. I'm really just trying to experiment with different features and configurations of multiple Asterisk machines and would prefer to do that in virtual space. I'm willing to make my configs available. I just thought I'd drop this email on the list hoping for the chance that someone has dealt and corrected this problem. :-) Thanks much in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk VOIP Jobs version 2 Launched!
Please start a new thread for messages. Replying to an old one messes up the archives and puts your conversation in with the old one for those of us with threaded mail clients. Also, please use a descriptive subject. This is a high volume list and without a descriptive subject that applies to your topic, a lot of people won't even read the thread. Regards, AR On Mon, Mar 17, 2008 at 9:57 AM, Rony Ron [EMAIL PROTECTED] wrote: Hello all, please, is it possible to which party has hangup a call? if yes, please tell me how ? thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Graceful Asterisk Shutdown
Hi Jeng, From the Asterisk CLI type stop gracefully and it will do exactly what you described (stop accepting calls and shut down when all calls have completed). If you don't want to stop accepting calls, but still want to stop Asterisk when there are no active calls, you can use stop when convenient. The same qualifiers (gracefully and when convenient) can be applied to the restart command. Cheers, AR On Dec 10, 2007 7:29 AM, Jeng Yu [EMAIL PROTECTED] wrote: My Gurus! I'm still playing with asterisk in the lab here. There is a feature that I need in a production asterisk system. I was wondering if it already exists in asterisk. When we want to shutdown a production asterisk system, we would like the shutdown to happen after there are no more calls being processed. In other words, a shutdown command that does the following: - block asterisk from receiving/answering all new connection requests - monitor existing call connections it is currently handling - when all calls/connections have ended, then effect the shutdown and stop the asterisk process. Is there a way to do this in asterisk now, and how? This would be the ultimate graceful shutdown; perfect for routine system maintenance tasks on production servers handling continuous traffic. Thanks, Jeng ___ --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] time on polycom 501
Hi Jerry, Here's what's in my SNTP tag: tcpIpApp.sntp.resyncPeriod=3600 tcpIpApp.sntp.address=192.168.15.50 tcpIpApp.sntp.gmtOffset=-18000 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0 I had the same issue as you. The issue was the dayOfWeek.lastInMonth. For some reason I had set mine to 1. Digium has a KB article stating that it should be 0. Cheers, AR On Nov 8, 2007 10:46 AM, Jerry Geis [EMAIL PROTECTED] wrote: I have a polycom 501 phone that is 1 hour off now. Before last sunday (time change) the time was fine. ?xml version=1.0 standalone=yes? PHONE_CONFIG OVERRIDES _.0x20._log.level.change.sip=0 tcpIpApp.sntp.daylightSavings.stop.date=4 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.gmtOffset=-18000 tcpIpApp.sntp.address=time.apple.com reg.1.ringType=4 lcl.cpt=0/ /PHONE_CONFIG I also have in dhcpd.conf: option ntp-servers 17.254.0.27; How can I get my polycom phones back to the correct time? Thanks, Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Red5
Hi Dean, The BlindSide project is using Red5. I'm not affiliated with them at all, but I think the project looks great. It's intended to be an open source web conferencing/webcasting platform. See: http://code.google.com/p/blindside Cheers, AR On 10/26/07, Dean Collins [EMAIL PROTECTED] wrote: Are there any asterisk users/developers who have been working with or trialing installations of Red5? It's an open source version of Adobe Flash Media Server - ( http://osflash.org/red5) If so please email/call to discuss. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux limits
On 9/18/07, Wai Wu [EMAIL PROTECTED] wrote: Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for asterisk1/700 Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in Linux itself. Thnx Hi Wai, I had this issue once (different software, unrelated to asterisk), and I used this guide to increase file handles: http://confluence.atlassian.com/display/DOC/Fix+'Too+many+open+files'+error+on+Linux+by+increasing+filehandles Cheers, AR -- Alex Robar [EMAIL PROTECTED] ___ 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] Partitioning DSL input
pfSense works very well for this. You can use it to setup VLANs (one for your PCs, the other for your VoIP equipment), and it has a traffic shaping/queuing mechanism for prioritizing VoIP. AR On 9/10/07, C. Savinovich [EMAIL PROTECTED] wrote: Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). Many Thanks C. Savinovich ___ 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 -- Alex Robar [EMAIL PROTECTED] ___ 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] State of the Union: Vonage? Skype?
Hi Jay, Skype can be used successfully with the ChanSkype module on supported platforms (Fedora Core 3, 4 or 5 or Ubuntu 6.04). It's $19USD for a single personal license, and tends to work quite well. It's not the easiest item to setup (the OS needs a window manager running on it, and each Skype channel require it's own user with it's own desktop session running for that user), but once you get it going I've rarely found it to fail. I used it with the unlimited outgoing calling to North America from Skype, and it's saved me quite a bit. AR On 8/15/07, Jay R. Ashworth [EMAIL PROTECTED] wrote: I've looked around a bit, and I'm still not sure I quite know what the state of the union is with regard to configuring SkypeIn/Out and Vonage services as trunk-side appearances on an Asterisk PBX? Any good clear pointers? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FreePBX
Hi Linga, You will likely get a much better response by posting to the FreePBX list (here: http://sourceforge.net/mail/?group_id=121515 ) or the FreePBX forums (here: http://www.freepbx.org/forums/ ). FreePBX is an entirely different animal on top of Asterisk and this group mainly focuses on vanilla asterisk questions. Cheers, AR On 8/13/07, R.Linga Reddy [EMAIL PROTECTED] wrote: Hi All, I am trying to install Asterisk with FreePBX while running install_amp following error is coming can any one help in this regards Thanks in advance.. Linga Reddy Connecting to database..OK Connecting to Asterisk manager interface..OK DB Error: no such tableGenerating AMP configs..OK Restarting Flash Operator Panel..OK ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] les.net losing DID's
One of the things that we've done is get a standard PSTN line in place that rings down to the VoIP lines. In smaller shops there's a single copper line, in larger shops they might have a T1/PRI. It's obviously more expensive than pure VoIP lines, but the stability of the number is solid; You know that number isn't going away. If you have to change the number that your inbound rings down to because your VoIP DID just disappeared, then so be it. Pay the fee to your telco and make the change if that happens. But at least you know that one number you have that you've published everywhere isn't going away anytime soon. We originally found our incumbent very resistant to this type of ring strategy (they didn't want to let the call roll over to a number that wasn't theirs), so we moved to using CLECs. Recently we've found that the incumbent has allowed us to do this on certain line types too... So much the better from a stability perspective. AR On 8/8/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Mail list [EMAIL PROTECTED]: In general how painful has this sort of thing been to people so far ? I am pretty hesitant to put any sort of number like that on letterhead, website etc., when there might be doubt about having it long term when its provided by a small company. It seems you do voip to save money but to have long term stability you have to get a number from a large company and the savings disappear, or the terms are restrictive to use, or some other negative, and then you end up doing nothing. Yes they are co-operating to port DID to another provider and they have given time till august 23 so DID will continue to work till then but they are not providing any substitute DID though ( i dont expect that ) but atleast they should partially refund amount for remaining days ( i dont expect that either :P ) . On 09/08/2007, Stephen Bosch [EMAIL PROTECTED] wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] les.net losing DID's
Jon, No, not at all - Sorry, that's not what I meant. Indeed, a local extension would be quite prohibitively expensive. What we tend to do with people who require out-of-area calling ability is grab a toll free DID from a bit of a bigger or more stable provider. Here in Ontario, Canada, we've had great success with Unlimitel for providing toll free DIDs. AR On 8/8/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Alex Robar [EMAIL PROTECTED]: surely you wouldn't do this where you are getting voip numbers so you can have local numbers in other areas. Having an analog or other rung through like that would be impossible in most cases and hugely expensive where actually possible. One of the things that we've done is get a standard PSTN line in place that rings down to the VoIP lines. In smaller shops there's a single copper line, in larger shops they might have a T1/PRI. It's obviously more expensive than pure VoIP lines, but the stability of the number is solid; You know that number isn't going away. If you have to change the number that your inbound rings down to because your VoIP DID just disappeared, then so be it. Pay the fee to your telco and make the change if that happens. But at least you know that one number you have that you've published everywhere isn't going away anytime soon. We originally found our incumbent very resistant to this type of ring strategy (they didn't want to let the call roll over to a number that wasn't theirs), so we moved to using CLECs. Recently we've found that the incumbent has allowed us to do this on certain line types too... So much the better from a stability perspective. AR On 8/8/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Mail list [EMAIL PROTECTED]: In general how painful has this sort of thing been to people so far ? I am pretty hesitant to put any sort of number like that on letterhead, website etc., when there might be doubt about having it long term when its provided by a small company. It seems you do voip to save money but to have long term stability you have to get a number from a large company and the savings disappear, or the terms are restrictive to use, or some other negative, and then you end up doing nothing. Yes they are co-operating to port DID to another provider and they have given time till august 23 so DID will continue to work till then but they are not providing any substitute DID though ( i dont expect that ) but atleast they should partially refund amount for remaining days ( i dont expect that either :P ) . On 09/08/2007, Stephen Bosch [EMAIL PROTECTED] wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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 -- Alex Robar [EMAIL PROTECTED] Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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 -- Alex Robar [EMAIL
Re: [asterisk-users] Hardware that can ring my phone?
Lynn, What you need is an ATA (analog telephone adapter). The ATA is a SIP or IAX extension on your Asterisk box, and your standard phone plugs into it. Asterisk sends the call to the SIP extension (the ATA), and the ATA rings your phone. On the flip side, your phone dials normally and the ATA digitizes the data and sends it via SIP to Asterisk for routing. Check out Digium's IAXy or the GrandStream Budgetone/HandyTone. AR On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote: Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Welcome to the asterisk-users mailing list (Digest mode)
Please start new threads for new messages (don't reply and just wipe out the body). The headers still exist so you wind up with screwy threading in the list archives (ditto for those of us who have e-mail software that supports threading). AR On 7/31/07, Richard Brady [EMAIL PROTECTED] wrote: Hi folks When connecting two SIP users, is there any way to stop Asterisk from sending SIP 183 Session Progress messages, either globally or per-peer? Call from UA1 to Asterisk (UA2) to UA3 UA3 sends RTP before SIP OK to Asterisk (UA2) Asterisk (UA2) detects early audio from UA3 and sends 183 Session Progress with SDP to UA1. Instead I would like it to just send on the early audio, is this possible? Thanks in advance, Richard ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Welcome to the asterisk-users mailing list (Digest mode)
I meant to send that just to you, not the list - My apologies. I wasn't trying to be the public list cop. AR On 7/31/07, Richard Brady [EMAIL PROTECTED] wrote: Hi Alex Apologies for that, I noticed this immediately after I sent the email and resent it with fresh headers and a descriptive subject line. I will be more careful in future. Regards, Richard -- Richard Brady T: +44 (0)7771 623 348 E: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] In Vancouver is it a local to call from 778 to 604, and vice versa?
Check here: http://www.localcallingguide.com/ AR On 7/18/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I've got a 778 DID for vancouver, but don't know if it will be a local call if dialed 604 and vice versa. What are the different area codes in Vancouver and why its easier to get 778 DID than 604? -- Zeeshan A Zakaria ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDI behind NAT?
Hi Andreas,
In dundi.conf, look for the line of yours that is similar to this:
e164 = dundi-e164-canonical,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER}
Change ${IPADDR} to your external IP address or hostname, like so:
e164 = dundi-e164-canonical,0,IAX2,dundi:[EMAIL PROTECTED]
/${NUMBER}
Cheers,
AR
On 7/10/07, Andreas Anderson [EMAIL PROTECTED] wrote:
Hi,
i'm having asterisk with sip working fine, including dundi lookups. The
only
problem i'm having is that the dundi answer allways contains my internal,
private ip. Is there any way to set the targeting ip that is sent out in
the
dundi answer (to my public ip or any other where i want to receive the
call)?
Regards,
Andreas.
_
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Re: [asterisk-users] Google acquires Grand Central
GrandCentral isn't about hiding your number, it's about reachability. Grand Central gives you a single number that rings your home, office, cell, etc... And provides a single voicemail box for all of those numbers. As Asterisk users, these features do not seem very ground breaking to us, as most of us have got this setup for ourselves already. But for someone with no telephony experience or equipment, it's a great product to have. AR On 7/9/07, Wai Wu [EMAIL PROTECTED] wrote: I don't see the point of the service provided by GrandCentral. Party A calls party B through GrandCentral. Party B know party A's number and calls party A back, now party A can call party B directly, and party A has party B's directly number. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Saturday, July 07, 2007 6:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google acquires Grand Central On 4 Jul 2007, at 17:57, Stephen Bosch wrote: Jaswinder Singh wrote: Think about voicesense which will sense what you are talking and pop in a *relevant* voice ad to spice up conversation :P . If this happens I am going back to tin cans and string. Hmm, time to get that IAX encryption working along wit ZRTP Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstPligg
Give the new site a break. I think it's a good idea. Sure there are lots of news sites for VoIP, but many of them are poorly designed, and I can't recall any that are very good at letting the users provide the news content. I agree that the name could be better, but after having just tried it out, I really like AstPligg. AR On 6/25/07, Mark Phillips [EMAIL PROTECTED] wrote: Great! Another one. With such a catchy name too! On Tue, 2007-06-26 at 01:42 +0200, lenz wrote: Hello list, AstPligg is a new Digg-like website devoted to * and VoIP news. At the moment, it's in beta stage and very basic - no fancy custom templates. It allows posting new stories, comments on stories, RSS feeds and tags. Still, it can be very useful, as the number of * sites and blogs grows every day, and keeping track of what is hot in the * world is increasingly difficult. Yes, I know, it's not much; but at least it's there and can be used immediately. You can find it at http://oinko.net/astpligg I'm looking forward to your comments (and stories) to make it a useful tool for the * community! l. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstPligg
I don't think anything is _wrong_ with VoIP-Info at all, I just think the sites serve different purposes. This is all just personal preference, but to me VoIP-Info does not work that well as a social news site, as all stories/headlines, good or bad, have equal weight. With the Pligg system, the stories that are better are usually voted up so they have higher exposure. VoIP-Info is a great site for sharing Asterisk recipes and HowTos, but I'm not a fan of it as a news site. AR On 6/26/07, Jon Weisman [EMAIL PROTECTED] wrote: Whats wrong w/ voip-info.org? Jon Weisman | Sales Engineer International Bell Communications www.ibell.net - Original Message - *From:* Alex Robar [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Tuesday, June 26, 2007 8:38 AM *Subject:* Re: [asterisk-users] AstPligg Give the new site a break. I think it's a good idea. Sure there are lots of news sites for VoIP, but many of them are poorly designed, and I can't recall any that are very good at letting the users provide the news content. I agree that the name could be better, but after having just tried it out, I really like AstPligg. AR On 6/25/07, Mark Phillips [EMAIL PROTECTED] wrote: Great! Another one. With such a catchy name too! -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP-UDP SIP proxy?
SIP Express Router (SER - http://www.iptel.org/ser/) is fairly common solution for this problem. AR On Wed, 6 Jun 2007 19:14 +0300, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, One of our faculties have Microsoft's LCS and would like to connect it to our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS talks SIP over TCP with TLS. Anyone can recommend a gateway between these two protocols? Thanks! __Yehavi: ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
Hi Olivier, I would guess that most people aren't running any type of GUI on their Asterisk box. Running an X server plus some type of window manager adds a lot of overhead that's completely unnecessary for a server. I just SSH into the server and use VI to edit the files - The server doesn't run any type of GUI, there's no reason for it to. Alex On 5/16/07, Olivier [EMAIL PROTECTED] wrote: Do you mean nobody has ever done this before (as I thought before asking this question to the list) ? So which tool KDE users are using for this ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox problems
It has nothing to do with the GUI. Trixbox compiles Zaptel for you and provides them as RPMs for installation. Removing the RPMs and all the configs they leave lying around and compiling from source can be a complicated process, and the Trixbox forums/mailing lists will be better able to help the OP in this case. AR On 5/15/07, Diego Iastrubni [EMAIL PROTECTED] wrote: On Tuesday 15 May 2007 19:11, Dave Cotton wrote: On Tue, 2007-05-15 at 17:45 +0200, Marco Vescovi wrote: Hello, I'm writing because we have problems with an asterisk installation (Trixbox ver. 1.2.3). We have a customer which is receiving a lot o telephony traffic (more or less 1 call/2 min.); we are using a TDM400 board, with 3 PSTN lines configured and we have two big issues: - Calls are dropped during conversation (I have a busycount=8 from the initial value that was 4) - Sometimes when the user dials out, he hears the ringing tone but the line is already answered and the called party hears his voice while he's still hearing the ringing tone. How can I investigate those 2 problems in order to find what's happening ? Contact the Trixbox mailing lists? Why is that? You think some fancy-shmancy GUI will fix this? The problem is obviously in the zaptel area. But hey... this is asterisk-users... /me is in a fighting mode today ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some help with a very simple Queestion..
Hi Gavin, You don't need queues to ring two phones, you can simply use the dial command: Dial(SIP/1SIP/10001) -- Would dial SIP extensions 1 and 10001. Now if you want the ability to have multiple people waiting on the line for those two extensions, that's when you need to look at the option of queues. Cheers, AR On 5/11/07, Gavin Spurgeon [EMAIL PROTECTED] wrote: Hi List, Just a simple question for the list this time.. I need to setup 2 Phones than can Both ring when an incoming call is made to a certain number... I have done this 3,000,000s times with CCM and have no problems with it, But it is the 1st time I have needed to do this with Asterisk. I think it can be done using Queues/Agents but I'm just unsure how do it.. The setup in question is a very small 5 Phones System based on SME 7.1 SAIL (Asterisk web interface) I have a small Sandbox setup here with me to test the test before I need to go set it up on the live system. My test phone is a Grandstream GXP 2000 but I will be using SPA-941's in the Live Environment Any help with this simple question would be great. Best Regards Gavin Spurgeon Systems Administrator Leigh City Technology College [EMAIL PROTECTED] http://www.leighctc.kent.sch.uk Tel: 01322 620501 Fax: 01322 620599 IS HelpDesk : Ext 541 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
Hi Ronaldo, Yes, you can use DUNDi for this. DUNDi simply advertises routes that a given server can terminate to its peers. As a very simple example, if ServerA houses extensions 500 through 599 and ServerB houses extensions 600 through 699, ServerA would advertise that it can terminate 5XX, and ServerB would advertise that it can terminate 6XX. When any peer in your DUNDi cloud requests how to terminate extension 502, ServerA will return a route to itself that will allow that call to be made. There's a nice article on the Texas AUG site about setting up DUNDi with dynamic extensions ( http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20with%20a%20Cluster%20of%20Asterisk%20Servers.pdf). Cheers, Alex Robar On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headset for Polycom
Hi Mike, Yes, they use a standard headset jack. In our implementations so far we've just had the customers continue to use their existing headsets. We take one of them from the customer ahead of time and test it out... So long as it works well, we replace their phones and keep the headsets. I can't say that we've found ones that work better than others. I'm sure that there are some really cheap ones that wouldn't work as well, but I've found that the customer has already invested a bit into the headsets since their employees will be wearing them all day long. The headsets are of good quality, and all seem to work about the same. Cheers, Alex Robar On 5/4/07, Mike [EMAIL PROTECTED] wrote: Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Wireless bridge for Polycoms
Hi Mike, How close together are these phones? If you have a few clusters of them, you can use the Linksys WRT54G devices to act as wireless bridges (with some open source firmware - I use DD-WRT). Each device will give you 4 ports to plug into. It's not a particularly cost effective solution to provide one WRT54G per phone, but if they're clustered you could centralize one bridge and plug 4 phones into it. Alex On 4/27/07, Mike [EMAIL PROTECTED] wrote: Hi, I'm stuck doing an install with Polycoms at a small office with no RJ-45. They went wireless 100%, poor them. I insist on using Polycom unless it's impossible because that's what I am standardized on for many reasons. What's the best way/device to turn a wired Polycom 501 (or any Polycom for that matter) into a WiFi phone? Mike ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk slows down when unplugging internet cable with VoIP lines
Hi Giorgio, LOTS of people have had this issue, check the list archives for some other responses. You are quite correct, Asterisk DNS is synchronous, so when your internet connection goes down, it causes some problems for the rest of the system. There are a couple of ways of handling this: 1) Don't use DNS. Use IP's in your Asterisk conf files instead of hostnames. (Make sure there are no hostnames ANYWHERE in the conf files). 2) Use an internal DNS server/cache. Setup a DNS server that caches queries somewhere on your network (or on the Asterisk box itself, if you're unable to put it somewhere else on the network). Set the Asterisk box to use your internal DNS server. This way, even if the internet goes down, Zap calls and internal calls will still be routed, as Asterisk will still be able perform DNS lookups. Both of these options have the downside that if your provider changes their IP, nothing works and you'll have to either change all your conf files (option 1) or clear your DNS cache and force a lookup again (option 2). However, most providers won't change the IP on your, and I would hope that if they did, they would notify their customers ahead of time. Cheers, Alex Robar On 4/26/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, I have an Asterisk 1.2.9.1 on a Debian Sarge distro connected to a VoIP provider via internet. I noticed Asterisk gets slow and behaves in strange manner if I unplug my internet cable from the PBX: for example I get incoming calls after seconds or I get no audio during calls. I thought it was something connected to DNS resolution so I put VoIP provider addresses inside /etc/hosts but still have slow problems. I made some tests adding registrations to providers inside sip.conf keeping my PBX disconnected from internet: after a sip reload the CLI simply stay freezed waiting for something. Trying to sip reload gives a message Asterisk is still waiting to perform the last reload. A real mess! I read on internet, inside dns.h file reference, Asterisk is using synchronous dns functions...infact a note explains that: Asterisk DNS is synchronus at this time. This means that if your DNS does not work properly, Asterisk might not start properly or a channel may lock How can it be? If this should be true this would be a big problem with VoIP lines since losing internet connections is not so uncommon (if so, why nobody else got this trouble?) Is it possible to bypass this behaviour or should I avoid VoIP lines?? TIA Giorgio Incantalupo ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Voice sound level
Hi Erik, Asterisk by itself does not have the ability to alter the media stream with regards to volume. Your zap config will allow you to set rxgain and txgain for the Austrian ISDN, but that might not suit your purposes for all other calls that use this trunk. I believe it was mentioned here before that this might be an interesting application for Justin Tunney's VoiceChanger application ( http://www.lobstertech.com/code/voicechanger/ ). It's a replacement for the dial command that alters the pitch of the media stream in real time. I imagine that with some work, a VolumeChanger application could be written based off of VoiceChanger. Cheers, Alex On 4/26/07, Erik Wartusch [EMAIL PROTECTED] wrote: Hi, Is there a possibility to control sound levels (higher / lower) in Asterisk (so the codecs). Somebody asked me to evaluate that but I didn`t found any documentation about. I have the opinion that for these (audio) things the end user client is the only part where I can tune around. Problem is for example a (Austria) ISDN -- Asterisk -- SIP / IP --- (Romania) Asterisk --- mobile forward. Then the sound level is very low.. i guess thats normal for this kind of different technologies and forwards but if anybody can tell me something better where i can adjust something Cheers, Erik ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing Voice from Male to Female
Hi Dovid, You can use the Asterisk Voice Changer application ( http://www.lobstertech.com/code/voicechanger/ ). The software allows you to change the pitch of your side of the audio stream in real time. I heard a demo of it at a Toronto Asterisk User Group meeting, and it does a pretty good job. It's relatively obvious that the voice has been altered... But it can give some fun effects none the less. Cheers, Alex Robar On 4/26/07, Dovid B [EMAIL PROTECTED] wrote: Hi List, I wanted to know if anyone knew of a way with asterisk to switch the voice of a caller from male to female or vice versa. Thanks. Dovid ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 33
David, It's not US format. He's away April 4th through April 11th. There was a big discussion about FB and his absence on this list a few days ago. Alex On 4/9/07, David Boyd [EMAIL PROTECTED] wrote: Could someone please remove this person from the list. It seems that the person is saying they will be away for (9) nine months, with their auto-reply set. dave On Mon, 2007-04-09 at 18:26 +0200, [EMAIL PROTECTED] wrote: Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] simplify
Hi Josu,
[miprimerejemplo]
exten = 2,1,Dial(SIP/${EXTEN},3-,Ttm)
exten = 2,2,Hangup
exten = 2,102,Voicemail(${EXTEN})
exten = 2,103,Hangup
... is all you need in that context. Asterisk will match any called number
that starts with a 2 and is 5 digits long. ${EXTEN} carries the value of the
dialed number.
Alex
On 4/2/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote:
hello friends,
is there any way to simplify that extensions.conf file?
[miprimerejemplo]
exten = 2,1,Dial(SIP/2,30,Ttm)
exten = 2,2,Hangup
exten = 2,102,Voicemail(2)
exten = 2,103,Hangup
exten = 20100,1,Dial(SIP/20100,30,Ttm)
exten = 20100,2,Hangup
exten = 20100,102,Voicemail(20100)
exten = 20100,103,Hangup
exten = 20200,1,Dial(SIP/20200,30,Ttm)
exten = 20200,2,Hangup
exten = 202000,102,Voicemail(20200)
exten = 20200,103,Hangup
exten = 20300,1,Dial(SIP/20300,30,Ttm)
exten = 20300,2,Hangup
exten = 203000,102,Voicemail(20300)
exten = 20300,103,Hangup
exten = 20400,1,Dial(SIP/20400,30,Ttm)
exten = 20400,2,Hangup
exten = 204000,102,Voicemail(20400)
exten = 20400,103,Hangup
thanks to all
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Re: [asterisk-users] error in FreePBX
What output does the CLI generate when you try to make a call? It will tell you what the system is doing, so it will usually give you a good indicator of what is causing the call to fail. Alex On 3/29/07, Carlos Jerónimo [EMAIL PROTECTED] wrote: Ive installed asterisk and freepbx. Through the interface ive configured 2 extensions, 6000 and 6001. My problem is that when i try to call from extension 6000 to 6001, i hear this msg Im-sorryan-error-has-occured and the call is terminated. As expected if i call to another number i get an error. i thought the problem might been related with the NAT but if checked and changed some NAT configuration parameters, it didnt worked aswell. As this ever happened to anyone before? Any hints are very appreciated. Thank you very much -- Carlos Jerónimo ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error in FreePBX
I believe that's Roger Workman's job... I'll go kick him and see that he activates you. Alex On 3/29/07, Carlos Jerónimo [EMAIL PROTECTED] wrote: Hi Steve, your sugestion is correct, but i registed 2 times in FreePbx foruns this week, and my login is inactive yet. In the mail i receive this msg: Welcome to FreePBX Forums Forums Please keep this email for your records. Your account information is as follows: Your account is currently inactive, the administrator of the board will need to activate it before you can log in. You will receive another email when this has occured. ** because this i post this here. Regards 2007/3/29, Steve Murphy [EMAIL PROTECTED]: On Thu, 2007-03-29 at 13:26 +0200, Remco Barendse wrote: On Thu, 29 Mar 2007, Carlos Jerónimo wrote: Ive installed asterisk and freepbx. Through the interface ive configured 2 extensions, 6000 and 6001. My problem is that when i try to call from extension 6000 to 6001, i hear this msg Im-sorryan-error-has-occured and the call is terminated. As expected if i call to another number i get an error. i thought the problem might been related with the NAT but if checked and changed some NAT configuration parameters, it didnt worked aswell. As this ever happened to anyone before? Any hints are very appreciated. Thank you very much I have the same problem, it seems to occur when an extension is busy here. All my extensions are on local lan with phones having ip addresses in a private range without NAT or anything so that is not the problem. Sounds like an error in the dial pan FreePBX generated. My suggestion: try a FreePBX mailing list first; the problem *is* more likely to be in their stuff. murf -- Steve Murphy Software Developer Digium ___ --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 -- Carlos Jerónimo ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FWD outgoing problem
Can you post the portion of extensions.conf where your Dial command is for FWD? From the output there it looks like you're trying to dial a FWD number from a Zap trunk. Alex On 3/21/07, Bogdan GONCIULEA [EMAIL PROTECTED] wrote: I have configured iax.conf and extensions.conf as instructed on FWD website (http://www.freeworlddialup.com/help/?p=knowledgebasec=18a=76 ) and I can successfully receive calls and make test calls to 612, 613, etc. The problem is that I can not make a call to another FWD user. Here is what asterisk says: -- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1, CALLERID(all)=xx) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1, IAX2/yy:[EMAIL PROTECTED]/xx|60|rhttp://IAX2/yy:[EMAIL PROTECTED]/xx%7C60%7Cr) in new stack -- Called yy:[EMAIL PROTECTED]/xx -- Call accepted by 192.246.69.186 (format ulaw) -- Format for call is ulaw -- IAX2/192.246.69.186:4569-1 is busy -- Hungup 'IAX2/192.246.69.186:4569-1' == Everyone is busy/congested at this time (1:1/0/0) -- Executing [EMAIL PROTECTED]:3] Congestion(Zap/1-1, ) in new stack == Spawn extension (default, 393xx, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' xx - FWD number I want to call yy - FWD number used by asterisk to register ppp - password for yy Thanks, Bogdan ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk on debian
Hi Josu, I've done it both ways, and they both generally work equally well (so long as the package maintainers are doing a decent job). As Victor mentioned though, the version you wish to install plays a factor in this. I found the Asterisk build in the repos to be a bit out dated. Also, it's always bothered me having to wait on another party to create a package so that I can fix a security vulnerability. I've just gone with the straight from source method for now, but that's all personal opinion on that matter really. The bottom line is that if you want the latest and greatest (in terms of both feature sets and security updates), build it yourself. Apt-get may be easier, but there's plenty of good guides to get you going with building from source. Alex On 3/20/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote: hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error, install freePbx
Hi Dima, You're better off following the Ubuntu guide written by the FreePBX developers: http://aussievoip.com.au/wiki/freePBX-Ubuntu Alex On 3/20/07, dima [EMAIL PROTECTED] wrote: perhaps you should try pear install DB However note that this mailing list has nothing to do with pear. Hi, i try install FreePbx by tuturial in http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Pasoview_comment_id=13443 but i have this error when i try install freepbx: #pear install db No releases available for package pear.php.net/db Cannot initialize 'db' , invalid or missing package files Package db is not valid install failed Why this error? help me, please. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx Incoming call's configuration
Hi Younss, You just need to setup Inbound Routes in FreePBX. The inbound routes allow you to route calls based upon caller ID or DID. Since you want to route based upon the number your caller dialed, you want to route based on DID. For your example: 1. Create a new inbound route. 2. In the DID field, enter the number you wish to route (555-4570). Keep in mind that this must match what your provider sends. Some providers send +1554570, some sent just 4570, and some send something in between. Check with your provider for their format. 3. At the bottom, select where you'd like that number to be routed to (I believe you need to select Core: Extension 202). Save the route, apply the settings (via clicking on the red bar), and that's it! Alex On 3/15/07, younss azzayani [EMAIL PROTECTED] wrote: Hi every body, I've set up a Trixbox Server with TE110P,all things seem to work fine(Thank You Malling lists irc Forums), but i need your help, i ve 30 numbre from 60 to 89, i need to specify for each sip extension a Zap number for example to call the sales service the caller must call 555-4570 and automaticly the caller will be redirected to the 202 ( sales service ) so nobody else can use this number ..70 im using freepbx, so can someone please help me :) Kind Regards ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DST changes for the US
Hi Peder, I think that CF was correct in his original post. From the Polycom SP IP admin guide: Attribuite: tcpIpApp.sntp.daylightSavings.start.date Values permitted: 1-31 Default: 1 Description: Day of the month to start DST. What the start.date=8 does is tell the phone to start DST on the first start.dayOfWeek it finds after the start.date. So in this case, we're telling it to start DST on the first Sunday (1) after the 8th of March (making it the second Sunday in March). Alex On 3/12/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ I'm pretty sure this is wrong: tcpIpApp.sntp.daylightSavings.start.date=8 Should be: tcpIpApp.sntp.daylightSavings.start.date=2 which indicates second week of month. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fix for TZ values updates for DST
Please start new threads for new questions. Alex On 3/12/07, Luis Claudio Santos [EMAIL PROTECTED] wrote: Somebody could help me with a call back implementation, please? I mean, I just want call to my Asterisk, hung up the phone, and wait it calls me back... Somebody ever did that for local or international calls? Thanks. LC. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mobility with asterisk
Hi Alvaro, There was a discussion about this a little while ago. Andrew Joakimsen had some good ideas with how to allow a wifi sip phone to roam between APs seemlessly. His post pretty much said the following: - Set all APs to the same channel and same SSID. - Make sure all APs are connected to the same LAN (no NAT on the AP). - Play with the settings on the phone with regards to roaming deltas and receive levels. Suggested settings: - RxLevel: -60 - PreRoaming: Enable - RxLevel: -75 - Try Over TxErrcnt: 15 - Try Over RxErrorcnt: 10 Playing with the pre-roaming settings will help you, but you may see a drop in battery life. Cheers, Alex On 3/7/07, Alvaro Pacho [EMAIL PROTECTED] wrote: Hello, I´m working testing every feature of asterisk in a lab. Now I am very interested in asterisk over network mobility environment. For example : when somebody is talking with his ip-phone ) and moving around a big enterprise, needing to change the ip-address (other AP) would it be possible in the minimum time to avoid loosing quality in the current call? I read this test http://lists.digium.com/pipermail/asterisk-dev/2006-December/025263.html but it´was written in December of 2006!! Were this ideas implemented? If you can help me with information about that please write me and I´ll test and give you my end result. Does anybody knows something about which is the best Cisco router to this mobility environment? Best regards, Pacho ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'
Hi Ken, Trixbox comes with the Flash Operator Panel. The FOP server is likely setup with incorrect authentication parameters, and hence is failing authentication everytime it attempts to use the Asterisk Manager API to update it's tracking of what's going on in your system. Check your op_server.cfg file (/var/www/html/panel/, I think). Look for the manager_user and manager_secret parameters, and make sure they match an entry in /etc/asterisk/manager.conf. Alex On 3/6/07, Ken Williams [EMAIL PROTECTED] wrote: Every few seconds I get the following message: * == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate * I'm trying to track down where it's coming from. I've used TCPDUMP NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked manager.conf (only lines are [general] enabled=yes). I've cleaned out /var/lib/asterisk. My full log shows the following every attempt: *[Mar 6 13:32:39] DEBUG[28578] manager.c: Manager received command 'Challenge' [Mar 6 13:32:39] DEBUG[28578] manager.c: Manager received command 'Login' [Mar 6 13:32:39] VERBOSE[28578] logger.c: == Parsing '/etc/asterisk/manager.conf': [Mar 6 13:32:29] VERBOSE[28567] logger.c: Found [Mar 6 13:32:40] VERBOSE[28578] logger.c: == Connect attempt from ' 127.0.0.1' unable to authenticate * I've updated from 1.2.13 to 1.4.1 and done everything I could to remove Trixbox from the picture. I thought for sure it was a module, but moving them all out of the picture didn't alleviate the problem. It seems as long as manager.conf exists I'm getting these messages. I've got 3 boxes setup with mostly identical setups (extensions.conf is different) and only one box is getting this message. From what I can tell from google searches it appears astbill and/or trixbox are likely to blame but I'm running out of places to look for these culprits. Any suggestinos would be greatly appreciated. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NetFilter (IPTables)
Hi, I've found this doc helpful in configuring my iptables: http://www.voip-info.org/wiki-Asterisk+firewall+rules Following those settings, my devices register and function properly. Alex On 2/27/07, -- [ UxBoD ] -- [EMAIL PROTECTED] wrote: I have this running on my Asterisk server, and have opened up ports UDP/5060 and UDP/1-2 but for some reason when I try and connect too my SIP extension it does not work. Are these the correct ports ? -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // SIP Phone: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Something wrong with the list?
I saw the same thing, but got a huge flood of messages today. A Gmail issue perhaps? Alex On 2/6/07, C F [EMAIL PROTECTED] wrote: Since Monday I didn't see much traffic. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nobody there, continuing...
Hi all, Running Asterisk 1.2.12 (a bit out dated, but it was fully operational until a few days ago), I'm seeing the following message in my logs, repeated literally millions of times: channel.c: Nobody there, continuing… We've started to see some odd behavior (incoming callers can hear us, we can't hear them, we can't dial out, etc). I read that this error might possibly be related to not setting rtptimeout, but I've set this and the issue persists. The symptoms seem very familiar to the types of issues we see when the internet goes down (call routing seems to get all screwy), but the connection appears to be fully operational when the symptoms appear. A reboot fixes the issues for about 3/4 of a day, but then they start happening again. Does anybody out there have any clue as to the meaning of the nobody there message is? Thanks, Alex Robar -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1 phone 2 voicemail accounts
Hi Chris, We have a customer who we set this up for on Polycom 501's. We set the first two lines buttons to be their own extension, and the last one to be the general delivery mailbox. If either account has a message, the MWI lights up. For transparency, you can have all the buttons say the same thing (ie. Ext. 223) so everything looks the same. Alex On 1/18/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: What is the best way to have 1 phone check multiple voicemail accounts. I am using polycom 650 phones, and am wondering if mwi can work when checking multiple accounts. -Chris Sent from my BlackBerry(r) wireless handheld ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nufone
I second that, seems to be working fine from here (Toronto/Rogers fiber connection). Maybe a lagging DNS or routing issue with your ISP? Alex On 1/15/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: I can connect to http://www.nufone.net/ just fine. Wiley Siler wrote: Are these guys still around? I cannot get to _www.nufone.net_ file://www.nufone.net or nufone.com ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restrict International Calls
Hi Julian, The way I do things is to break my call types up into blocks, such as [emergency], [local], [longdistance], [international], [localextensions], etc. Then, I create contexts for my users which include those basic blocks. For example, a courtesy phone at reception (the [courtesy-phones] context) has emergency, local and localextensions included in it. Then when I setup an extension for a courtesy phone, I set context=courtesy-phones. Alex On 1/11/07, Julian Varanini [EMAIL PROTECTED] wrote: Hi, Does anyone have a good example of how to restrict International calls to only certain users? I have been messing around with contexts in the extensions.conf file with no success. Thanks Julian ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi ENCREJ
Hi Ramon, Please post your peer details from dundi.conf so we can see what your setup is. Also, have you tried regenerating your keys? I wound up generating my keys twice, they just didn't work the first time, I'm not sure why. Alex On 1/10/07, Ramon Schönborn [EMAIL PROTECTED] wrote: hi list, i have the same problem as mentioned here: http://forums.digium.com/viewtopic.php?t=2678view=nextsid=bd94cefd823b23156c5748843febb3ab my asterisk version is 1.2.12.1 any ideas? ___ Der frühe Vogel fängt den Wurm. Hier gelangen Sie zum neuen Yahoo! Mail: http://mail.yahoo.de ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)
I use pfSense, which is based upon m0n0wall. It provides a lot more features than a stock m0n0wall, and I haven't had any problems with it. The RRD graphs it provides are really great informational tools, and there's a built in QoS wizard that even has Asterisk as a built-in option to prioritize. Alex On 1/6/07, Robbie Hughes [EMAIL PROTECTED] wrote: As I posted yesterday, Use m0n0wall from m0n0.ch on an old pc or a little router box for the best results. I use draytek 2910 routers and they work fine. On 6/1/07 19:00, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Re: Best inexpensive home office router for VoIP (QoS with maybe PoE) ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk (FreePBX) and queues
The problem is that you've setup 2003 as a static user. In FreePBX, static users are ALWAYS in the queue, no matter what. Take this guy out of the queue as a static agent, and have him login and logout as he needs. (login via ##* and logout via ##**, where ## is the number you've given your queue in FreePBX). Alex On 1/5/07, Felipe Neuwald [EMAIL PROTECTED] wrote: Hi folks, I'm using a fewestcall queue here, and I'm having the follow problem: I have 3 static agents in my default queue: 2001 2002 2003 User 2001 and 2002 are logged in, but 2003 are logged out. When someone call to my default queue, the queue try to ring 2003 (that isn't logged). There is some way to the queue only ring logged users? Here is my show queue: zeus*CLI show queue 100 100 has 0 calls (max unlimited) in 'fewestcalls' strategy (4s holdtime), W:0, C:3, A:3, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED] /n (Unknown) has taken no calls yet Local/[EMAIL PROTECTED]/n (Unknown) has taken 1 calls (last was 346 secs ago) Local/[EMAIL PROTECTED]/n (Unknown) has taken 2 calls (last was 195 secs ago) No Callers Thank you, Felipe. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best inexpensive home office router for VoIP(QoS with maybe PoE)
Hi Mike, The Linksys WRT54G can do QoS, and I've found it to be a great little router... I install the DD-WRT open source firmware on mine for additional features, but the stock firmware works well also. Alex On 1/5/07, Mike [EMAIL PROTECTED] wrote: You're quite right, I typed before thinking. Upload is the problem anyways, since it usually (in homes) uses much more limited bandwidth than downloading does. No answer to my question though: How do you people handle QoS without relying on the phones to do that? I'd like a box that can be purchased and installed easily (Linksys type of product) Mike -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Olivier *Sent:* Thursday, January 04, 2007 15:56 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Best inexpensive home office router for VoIP(QoS with maybe PoE) Having QoS on your router is valuable to prevent some large download from buggering your calls though. Isn't QoS only useful to prevent large uploads, as download rely on ISP router prioritizing Voice over Data ? Cheers ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] have a phone number from stanaphone and a working trixbox, how do I connect them?
I used these directions to get Stanaphone working on my FreePBX box: http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+6#614Stanaphone Alex On 1/3/07, blackwater dev [EMAIL PROTECTED] wrote: I have a phone number for traditional phone lines through stana phone and a working trixbox server. What do I need to do to connect the two so when someone calls the number from a normal phone, they get my server? Thanks! ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] over 200 queues, anyone?
Not necessarily... The same agents could very well be providing support for multiple companies. You wouldn't want an announcement from company A in company B's queues. Alex On 1/3/07, Joe Dennick [EMAIL PROTECTED] wrote: Yeah, get a Business Process specialist to analyze the client's environment and develop a better solution. 200 queues with only 100 agents sounds pretty ludicrous to me! On Wed, 2007-01-03 at 14:22 -0600, lenz wrote: Hello list, one of our clients is going to be deploying a system with over 200 differently composed queues and 100 agents. We are going to do a full test of the viability of this solution before deployment, but I was wondering if anyone has experience of such a setup and if there are any obvious problems or no-nos. Any suggestion welcomed, l. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] connecting asterisk (trixbox) to traditional phone lines?
I agree with this... The cheapest way is to do this without anymore hardware. Grab a pay-as-you-go VoIP provider (VoIPJet, Unlmitel, Gizmo Project, etc.) and setup a trunk. They'll give you a number callable from the PSTN, and that's all you need. The setup you have already can handle a voip trunk with no additional hardware. Ales On 1/2/07, Todd H [EMAIL PROTECTED] wrote: To go nice and cheaply, you could just get a free number from IPKALL.com or Stanaphone.com.. And do it all over IP... -t- On Jan 2, 2007, at 10:15 PM, blackwater dev wrote: Ok, I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. Thanks! ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WIFI SIP- The Best phone
I agree with regards to the standby time. The Dlink DPH 540 has comparable talk time, but 30 hours of standby time. I sometimes go for 18 hours or so before my phone can see a charger again... Alex On 12/30/06, Noah Miller [EMAIL PROTECTED] wrote: HOWEVER- The Zultys WIP 2 is an INCREDIBLE WIFI B/G SIP PHONE- IT IS EXCELLENT IN ALL RESPECTS. Thanks for the tip! I hadn't seen these advertised before, and I've been searching for some time for a Wifi SIP phone that can handle multiple line appearances. One Question: Really only 12-13 hours on the standby time? That seems pretty short in comparison with all the other wifi phones. - Noah ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trixbox web-administration
Hi Kurt, You'll most likely get a better answer for this question from the Trixbox forums at Trixbox.org. Trixbox is a pretty specialized distribution of Asterisk, and this list is generally for plain vanilla asterisk-related questions. Cheers, Alex On 12/29/06, Kurt Kuo [EMAIL PROTECTED] wrote: Hi list, trixbox web-administration can be reached by host ip. since I am trying trixbox on the machine where I host my website as well, can I move trixbox main page to xxx.xxx.xxx.xxx/asterisk? which file I should move and should I modify the file? Thanks. Kurt _ Get live scores and news about your team: Add the Live.com Football Page www.live.com/?addtemplate=footballicid=T001MSN30A0701 ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
Of course everyone is allowed to use VoIP... Asterisk is open! I think Dovid's point was more that this guy's website says he buys and sells precious metals and other random items, his postings on this list indicate that he installs PBXes and resells VoIP services, and then his private e-mails say that he's a PI. The PI thing sounds just like him trying to get those who attacked him to back off. Alex On 12/28/06, Kevin Walsh [EMAIL PROTECTED] wrote: Dovid B [EMAIL PROTECTED] wrote: A PI that does asterisk on the side ?? WTF ?? Do you have a list of business types that are not allowed to use VoIP? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Wifi SIP phones - LinkSys WIP330
Hi Wayne, I was a very lucky guy this christmas, and received a D-Link DPH-540. Despite the very first gen feel of the phone, I have been very impressed so far. You are correct in thinking that it can act as an extension external to your network. So long as the place you're in has a decent router, it shouldn't be a problem. I have tested the phone within my local network, as well as on three other wifi networks that my friends gave me the WEP keys for, and I was able to register fine, as well make and receive calls without issue. On one network, I needed to turn the registration refresh down to 90 seconds (down from one hour) to keep the NAT hole open (but I have to do that with my Polycom 501 at the office too). I set the phone to use G729 (to lower bandwidth usage), and I've found the quality to be great. Depending on where I was, there was a slight delay, but that's typical of any IP phone outside the local net if the router is QoSing VoIP or the net connection isn't up to snuff. The only negative things I have to say about the phone are: 1) You can only store 6 network profiles. I can think of 5 off the top of my head that I visit frequently. If the 6th is left unused for open APs, what happens when I find a sixth wifi enabled venue that I visit? Hopefully this is an artificial limit that will be upped with a firmware upgrade. 2) The refresh rate is _terrible_. It's not really an issue since you're generally not looking at the screen except for dialing, but it would be nice to see some type of fluid refresh. 3) Data entry is rough. There are only two input modes: text or numeric. The text mode defaults to uppercase characters, and if you want to enter a lowercase character, you have to cycle through all the uppercase characters on a key before you reach the lowercase ones. For example, a lowercase a takes four taps of the 2 key. WEP keys are case-insensitive, so that doesn't matter, but phone book entries are a nightmare. The only saving grace for this is that you can access the phone via a web interface and edit your phone book from there. I've found that I get a number from someone, type their name quickly in uppercase and then fix it later via PC when I'm connected at home. Cheers, Alex On 12/28/06, Wayne [EMAIL PROTECTED] wrote: Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as long as the relevant ports were open on the firewall) it would be possible to still be an extension to * if you could access the internet from, say, a wifi hot spot that was not a part of the lan? Thanks Wayne . ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The Good, Bad and Scam VoIP Providers
As if we needed more proof that Bochter was a screw-ball... He's now accused me of being the owner of TRXTel. Not that we needed proof he wasn't actually a PI, but in case anyone had any doubts, read the thread. Alex -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Dec 28, 2006 7:41 PM Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers To: Alex Robar [EMAIL PROTECTED] There are small minded then there is you Bent Fuck you Your spoof email address is blocked Get a life and stop your scams by hiding.. use a real email address... You are a waste of my time GOOD BYE :-P Best regards, Al Bochter Bochter Serviceshttp://www.BochterServices.com/?t=Email Alex Robar wrote: If you actually wanted to give the information to people, you would have just posted it instead of ranting like a lunatic. Your real problem is that you need attention. Stop being a diva and deal with stuff like this on your own. The bottom line is that if you actually had a case, you would have just proceeded with it and dealt with this privately like any normal, decent person would have done. My gut tells me you have jack shit in terms of evidence, and you were just fired as a customer by Brent for pulling shit like this... Something I would certainly agree with him on if that's what he did. I'll bet this never moves forward and we'll never hear anything about any action you've taken. In fact, I'll bet we'll see the inverse - That TRXTel has sued you for libel for attempting to defame them in public. And FYI, I actually did answer your question, you just didn't read my response... Something quite common in your responses, it seem. Alex On 12/28/06, Al Bochter [EMAIL PROTECTED] wrote: Alex But if you READ the posts. I replied to all OFF THE LIST So that is YOUR POINT... They posted my replys That were off the list to the list I blocked the other two jackasses on the server to stop there pointless messages. They can't send any messages to any users at any domains on my servers. The same as we are talking OFF THE LIST // The way you insulted the owners of TRXTel, not to mention the half a dozen other list members who defended them, was very childish. What you need to do is check into the PERSON (*Thats one owner*) that is around 28 years I have a list of 32 others that were scammed by bent Ask me for the links on textel no one as asked for the links.. The point is I am not going to waste any more of my time on the ones like you that don't what the information on the truth. *By the way you never answered my question Do you want to be scammed and lose your money???* New question?? What is unlimited use So your replys are pointless Best regards, Al Bochter Bochter Serviceshttp://www.BochterServices.com/?t=Email Alex Robar wrote: The POINT that you keep whining and complaining about so much, is that you're trying to bully and scare people into ceasing their posts that reflect negatively on you. The original points of your post are not what anyone is focusing on anymore - YOU moved the points away from that by insulting people. Everyone else who is off the point is simply responding to you. The issue here is not that anyone LIKES to be scammed... But that you've insulted valuable, respected members of the Asterisk community simply because of a bad experience you had. To post a complaint is one thing, to rip into someone the way you did is quite another. The way you insulted the owners of TRXTel, not to mention the half a dozen other list members who defended them, was very childish. Alex Robar On 12/28/06, Al Bochter [EMAIL PROTECTED] wrote: Alex This is off the list. The point is that I don't like scammers. The ones that tried to attacked are some of the scammers that I am dealing with. Do you like to get scammed out of your money? And what is the point of I am a PI or not. Thats not the point of my message or the subject So if you like to get scammed then there is no point to a reply to this message. Only if you want some links to the sites where you will lose your money... ;-) Hope you have great day! Best regards, Al Bochter Bochter Serviceshttp://www.BochterServices.com/?t=Email Alex Robar wrote: Of course everyone is allowed to use VoIP... Asterisk is open! I think Dovid's point was more that this guy's website says he buys and sells precious metals and other random items, his postings on this list indicate that he installs PBXes and resells VoIP services, and then his private e-mails say that he's a PI. The PI thing sounds just like him trying to get those who attacked him to back off. Alex On 12/28/06, Kevin Walsh [EMAIL PROTECTED] wrote: Dovid B [EMAIL PROTECTED] wrote: A PI that does asterisk on the side ?? WTF ?? Do you have a list of business types that are not allowed to use VoIP
Re: [asterisk-users] Searching the list
Hi Mark, I don't think there's a built in search (someone please correct me if I'm mistaken here), but Google can filter results for you: site:http://lists.digium.com/pipermail/asterisk-users/ searchterm Alex On 12/27/06, Mark Greene [EMAIL PROTECTED] wrote: Hey guys. I am new to the list and would like to know how to search it so that I do not post any questions that have already been answered (like this one) - Mark ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Insert 1+areacode for VOIP calls
Hi Phil,
Using your example:
exten = _NXXNXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/1${EXTEN})
... Would match NXX-NXX- and pop a one in place of what you dialed.
Alex
On 12/21/06, Phil Finkler [EMAIL PROTECTED] wrote:
Greetings,
Currently my asterisk box is using Voicepulse. It works fine with the
exception that people need to enter the 1+area code for local calls. I'd
like to get around this if possible. The following is what I have in my
extensions.conf..
exten = _1NXXNXX,1,Set(CALLERID(num)=6162997590)
exten = _1NXXNXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN})
exten = _1NXXNXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|500)
exten = _1NXXNXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/${EXTEN})
Is there a way I can create a _NXX extension and insert 1 and areacode
when dialing?
Any help appreciated,
Phil
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Re: [asterisk-users] Insert 1+areacode for VOIP calls
Phil,
Yeah, I just realized that I didn't answer your question. Time Bandit did
though, look at his solution!
Alex
On 12/21/06, Alex Robar [EMAIL PROTECTED] wrote:
Hi Phil,
Using your example:
exten = _NXXNXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/1${EXTEN})
... Would match NXX-NXX- and pop a one in place of what you dialed.
Alex
On 12/21/06, Phil Finkler [EMAIL PROTECTED] wrote:
Greetings,
Currently my asterisk box is using Voicepulse. It works fine with the
exception that people need to enter the 1+area code for local calls. I'd
like to get around this if possible. The following is what I have in my
extensions.conf..
exten = _1NXXNXX,1,Set(CALLERID(num)=6162997590)
exten = _1NXXNXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN})
exten =
_1NXXNXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|500)
exten =
_1NXXNXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/${EXTEN})
Is there a way I can create a _NXX extension and insert 1 and
areacode when dialing?
Any help appreciated,
Phil
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Re: [asterisk-users] Need Wholesale Termination
Hi Shady, You'll have better luck posting this to the -biz list. This list is for non-commercial discussion only. Alex On 12/20/06, Shady [EMAIL PROTECTED] wrote: Looking for a good termination provider for US/Canada Please contact offlist. Shady ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX connection to FWD not working
Hi Timothy, Mine seems to be working OK as of a few minutes ago: unlimited*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 727044 216.58.41.183:4569 60 Registered Do you have any other IAX trunks? Are they working for you? Alex On 12/20/06, Timothy Parez [EMAIL PROTECTED] wrote: Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 814179 Unregistered 60 Timeout 192.246.69.186:4569 805208 Unregistered 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX connection to FWD not working
You mean that you can't call other FWD users? Alex On 12/20/06, Timothy Parez [EMAIL PROTECTED] wrote: However I can call 613 and it works I can be called and it works but when I call any other number I get call ended right away :p Alex Robar wrote: Hi Timothy, Mine seems to be working OK as of a few minutes ago: unlimited*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 http://192.246.69.186:4569 727044 216.58.41.183:4569 http://216.58.41.183:4569 60 Registered Do you have any other IAX trunks? Are they working for you? Alex On 12/20/06, *Timothy Parez* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 http://192.246.69.186:4569 814179 Unregistered 60 Timeout 192.246.69.186:4569 http://192.246.69.186:4569 805208 Unregistered 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk to asterisk - to zap
DUNDi can do this for you. Advertise the routes you can terminate on Box A. When you place a call on Box B, have it check your DUNDi cloud, and Box A will provide the route and terminate the call via zap for you. Alex On 12/18/06, Pryakhin Dimitry [EMAIL PROTECTED] wrote: Hello that might would be an easy question for someone, but im in doubt Is there any possibility to pass a call from one asterisk to another and then to ZAP channel. For instance I have A asterisk with numbering 45670 B asterisk with numbering 45680 second asterisk has TE110P card with single PRI port connected to Siemens EWSD. When I originate call from asterisk B I reach the world thru ZAP, when I call from asterisk A I reach numbering of asterisk B but cant get to the PSTN network. ASTERISK---ASTERISK-ZAP-PSTN Should I have OpenSER for that and terminate my call on CISCO AS5350 or something? Thanks ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] for all Asterisk Users
Hi Uugan, Because of the numerous additions and changes that Trixbox makes to the system, you're better off posting this question to the forums on Trixbox.org. As I recall, there is a forum dedicated to H.323 there. Cheers, Alex On 12/6/06, Uuganbayar.B [EMAIL PROTECTED] wrote: I have installed Asterisk from TRIXBOX.1.2.3 Please help me, How to I configure H.323 TRUNK between Trixbox and AvayaIPPBX. Uugan ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odd issue with IP tables
Hi Curt, I would try to find out what it's doing that's slowing your system down so much. Try turning up logging (or examining the logs) to see what's going on there, or what it's waiting for. Off the top of my head, I'm wondering if you've blocked everything else, or have you allowed through the standard fare? Ie. Have you allowed existing and related connections? Also, where is iptables sitting? Is it on the local Asterisk box, or is it on a firewall/router box in front of the Asterisk box? Alex On 11/18/06, Curt Shaffer [EMAIL PROTECTED] wrote: I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 1-2. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some suggestions? Thanks Curt ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx changes dont reflect in asterisk
Hi Pedro, Did you press the red bar at the top of the page? Until you do this, the config files are not written out. Alex On 11/17/06, Pedro Silva [EMAIL PROTECTED] wrote: Hello, From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx changes dont reflect in asterisk
I think you guys are all misunderstanding the problem here. Unless I'm misunderstanding, Pedro's issue is that when he makes changes in FreePBX, those changes are not written out to the config files. Alex On 11/17/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: You can't do any modifications in extensions_additional.conf and sip_additional.conf. Same is true for extensions.conf and sip.conf, and other original trixbox files. As soon as you press the red bar, they are returned to their original state. For modifications, you create your own files or use sip_customs.conf and extensions_custom.conf. Please don't mix trixbox with asterisk just because its based on asterisk. Its a completely customized solution of various software packages configured to make it work according to its own requirements. For help, post on trixbox.org forums. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2000 -- What's the Catch?
Doug, Just a note on this subject: I have a Snom 320 at home, and it's got a nice orange MWI that's pretty visible (especially if the apartment is dark). At the office I have a Polycom 501. It's got a great red light right at the top of the phone in the middle. It's very visible unless the phone isn't facing you at all. Alex On 11/15/06, Doug Crompton [EMAIL PROTECTED] wrote: Well I have a Grandstream 200 in a home application and so far I have been happy with it. My biggest complaint is that 99% of these IP phones are black!! One of the reasons I bought the 200 was because it has a bright red, see across the room, message waiting indicator. I have not seen that spec'ed on other phones. That doe not meant they don't have it, it is just not spec'd. I imagine the multiline LCD's have it on the screen, but you would not see that unless you specifically walked over and looked. I would be interested if any other phones have message waiting indicators as visible as the GS 200. Doug On Wed, 15 Nov 2006, Tom Vile wrote: They brake easy. Speaker phone is not very good. Overall sound not good compared to a Snom, Polycom or Cisco phone. Drop registrations with Asterisk randomly. Power supplies die. Had 4 out of 10 go bad within a year. LCD backlight died on 2 that I deployed. We only do the Snom 320 or 360's now and are just as easy to configure and have alot of great options as well. On 11/15/06, Jeronimo Romero [EMAIL PROTECTED] wrote: We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF intercom right out of the box. They can also be centrally managed and provisioned. They also sound great and work in a very intuitive way. We don't have real life experience deploying this phone so I'm just going to ask: Is there a catch? Why the huge price difference? These phones seem to do everything a busy corporate office would need. Is there a big qualitative difference between this phone and Polycom501/601?? Is there a major problem with this phone not disclosed by the manufacturer or vendors. Some feedback from people who have deployed them would be great. Thanks In advance. JR ___ --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 -- Tom Vile Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why dont my messages get through
They do get through. Messages you send to the list won't get sent back to you, because you sent them. On 11/7/06, Christian [EMAIL PROTECTED] wrote:Hi,My messages to the list don't get through. This must be the tenth message i am trying to send! Please ignore this test message.___--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-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft will enter VoIP market in earnest next year, says Ballmer
Unified messaging would be nice. Not just having my VM's e-mailed to me, but to be able to manage them from with Outlook (or any other mail client for that matter) would be nice. I picture it sort of like an IMAP mailbox, and the mail client just has some kind of functionality to recognize that the message is a VM and not a mail message (so it could display length, date/time received, CID, and provide a play button). Just my two cents.AlexOn 11/7/06, Dean Collins [EMAIL PROTECTED] wrote: http://www.siliconvalley.com/mld/siliconvalley/business/international/asia/15944981.htm There's not much in the article so only click through if super interested but I'm curious and looking for people's opinions. What application integration would you like to see between MS (either Office or other aspects of the vista/xp OS) and Asterisk. Apart from dial from outlook and number pop I'm kind of curious what other functionality there is to be developed (I'd also like to see drop and drag from outlook into conference calls. What would you like to see in asterisk, if we get some solid responses we'll see about organizing some bounties to get it developed. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why dont my messages get through
I _was_ sure until mention it just now... I certainly don't get a copy of any messages I sent to the list, whether I send from my personal or office accounts. Maybe the way my mail clients are handling it? If so, my apologies to Christian. AlexOn 11/7/06, Nick Hoffman [EMAIL PROTECTED] wrote: On 11/7/06, Christian [EMAIL PROTECTED] wrote: Hi, My messages to the list don't get through. This must be the tenth message i am trying to send! Please ignore this test message.On Wed November 8 2006 13:08, Alex Robar [EMAIL PROTECTED] wrote: They do get through. Messages you send to the list won't get sent back to you, because you sent them.Hi Alex. Are you sure about that? I receive a copy of every email I send tothe list.-- NickE: [EMAIL PROTECTED] P: +61 7 5591 3588F: +61 7 5591 6588If you receive this email by mistake, please notify us and do not make anyuse of the email.We do not waive any privilege, confidentiality orcopyright associated with it. -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] In bound SIP context issue
I would think that if the call isn't using the information you've setup under username1, then the call probably isn't coming into your system using username1. Try to verify which username the call is being sent to. AlexOn 11/3/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi All, I am trying to configure asterisk to receive an inbound SIP connection and send it to a specified context. Instead of sending the call the specified context, asterisk is using the context default from [general]? Any thoughts? I am sure that it is something simple I am missing. To recap, it is sending calls to the context default, not thecontext... [general]context=defaultsrvlookup=yes [username1]type=peerusername=username1secret=test1234host=dynamicdtmfmode=rfc2833context=thecontextnat=nodeny= 0.0.0.0/0.0.0.0permit=XXX.XXX.XXX.XXX /255.255.255.255permit=XXX.XXX.XXX.XXX/255.255.255.255permit=XXX.XXX.XXX.XXX/255.255.255.255permit=XXX.XXX.XXX.XXX/255.255.255.0permit=XXX.XXX.XXX.XXX/255.255.255.255 Thanks ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to clear trixbox configuration
You don't need to reinstall everything, only FreePBX. Upon installation, FreePBX will create any missing .conf files. Your best bet is to backup everything under /etc/asterisk, and then delete your sip*.conf, iax*.conf and extensions*.conf (where the * indicates all included files, such as sip_additional.conf and sip_custom.conf). Then grab the tarball of FreePBX and run the install script. You should have the default FreePBX conf files now.AlexOn 11/2/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:Only an expert sitting beside you can help you now. Otherwise you'll have to reinstall everything. Also this is Asterisk mailing list, not Trixbox. Trixbox forums are www.trixbox.org . ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax eater
You could hack together some kind of solution for accepting the fax digitally (turning it into a PDF or some other type of file) using SpanDSP and app_rxfax, and then just have a script that runs every hour or so, deleting the generated images. I've never done this, but you should be able to receive the fax digitally without issue. From there, it's a simple cleanup script. Cheers,AlexOn 11/2/06, James Harper [EMAIL PROTECTED] wrote: We have a 100 number indial range and every so often get fax calls onour voice numbers (our fax number isn't in the 100 number range). If you just hang up the sending fax will often try a few times before finallygiving up.Our outgoing fax is connected to the PBX (not asterisk), and we can do ablind transfer to that which will print it out, but right now the fax is printing a misdialled fax and it's up to about 3 meters long and stillgoing.I have an asterisk server plumbed into the PBX via an ISDN trunk, so I'mthinking that if I could map an extension to that which would just 'eat' any fax we transfer to it, it would save some paper. Any fax coming inon the 100 number range isn't something we want anyway.Anyone done this before?ThanksJames___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages
Alok, Two things: 1) You said you installed AMP. AMP has ceased development a while ago, but is survived by the FreePBX project. If you actually installed AMP and not FreePBX, I would suggest you get FreePBX running first. A lot of effort went into improving FreePBX from AMP. 2) You typically won't find much help for the GUIs from this list because the GUIs have their own mailing lists and forums. Try posting your question to FreePBX.org. You're more likely to get a response there. Alex On 10/31/06, Alok Mohapatra [EMAIL PROTECTED] wrote: Hi All, I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 2.2. Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) . Thanks and Regards Alok Mohapatra ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium vs. Sangoma
Everyone has their own opinions one way or another with regards to voice boards... Mark has every right not to like a competitor's product (though I'd say that asking someone simply wearing a competitor's logo to leave is a bit over the top). But it doesn't matter one way other the other. People will use which products they want to; Those battles have been fought on this list before. AlexOn 10/23/06, Brian Roy [EMAIL PROTECTED] wrote: 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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with Dialplan Rules Please!
If the order is giving you problems, create two separate outbound routes, one for local calls and one for long distance. Make sure the local route is before the LD route, and it should work for you. Both outbound routes can use the same trunk without issue. AlexOn 10/17/06, Chris Ramsey [EMAIL PROTECTED] wrote: This was posted at The Asterisk Blog Forums Click here for the original post. I need someone to explain how the dialplan rules work? I'm having a hard time getting it. I know that to dial out you need a 9 and to ignore that 9 once your out... requires a switch of sorts that tells asterisk to ignore the first digit on the left. In freePBX it's this: 9|NXX For Long distance it is 9|1NXXNXX Here is my problem using Free PBX: I want to be able to dial long distance and local at will while using free PBX to set it up. Right now we have 1 line for testing purposes and soon to be expanded into 2. When the rules are arranged like this in FreePBX 9|1NXXNXX 9|NXX the long distance portion works but the local one does not. When its arranged like this 9|NXX 9|1NXXNXX They both work! But the above is only done when it's hard coded into the configuration file (additional_extensions.conf) and free PBX always puts it in this order... wether I like it or not. 9|1NXXNXX 9|NXX And causes problems in the configuration file when and I change the settings. This isn't going to help me much! Im just a tad bit confused. A Little help here?-- www.AsteriskBlog.comYour home for easy to learn Asterisk stuff. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stopping putgoing calls after working hours
If the phone is not registered, how will you make outgoing emergency calls?AlexOn 10/17/06, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote: Dear Rich, It seems that my question is very general I apologize for that, but I am glad to see others like yourself pointing me in different directions, it seems all around the world we have problems with the cleaning folks. What I have in mind is to make the phone user lock his phone when he is leaving with a special code and relock it back when he comes to work (and as for emergency calls there are attendants who work at night who will be able to make an emergency call whenever needed at the spot), now there is nothing that seems to be able to do that directly, I have played around with the gotoiftime and also the time based dial plan include sent in mails before that. But while working I thought of another approach why not create a php web interface that each user logs in with a special username and password and gives him access to lock his phone, and what php does is actually change the secret password to something else than the configured on the phone, this should make the phone unable to authenticate thus not being able to make a call, and unlocking it returns the password to it's right form, I have already found the tables that I need to play around so I will restart making the php. I will update the list back with my final result. Do you guys think I could send a mail to the dev site to see if they can add this feature to asterisk. Thx MAG Rich Adamson wrote: I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk?? You need to better describe your objectives. If you really mean stop all calls (including emergency calls), that's easy. If you mean stop all calls that cleaning folks initiate (usually not employees), that just requires some extensions.conf changes to force the user to enter an access code before a call can be placed. (Just don't advertise that access code anyone that you don't want making calls. If your talking about a fairly major security issue (such as your users call forwarding their phones to the brother-in-law after normal hours, you'll probably need to disable call forwarding on the phone itself. If your talking about primarily managing expenses, use the CDR detail to generate a personalized report for each employee show this calls make between 5pm and 7am, and forward that report to each employee (and cc: the manager). That's usually enough to significantly cut those calls. If you don't have a policy relative to use of company assets (phones PC's) for personal use, you might put one together and reference that policy in the morning CDR detail report. (I'm sure at lease some of those calls are likely legitimate calls, so cutting all calls is not likely a workable solution. ___ --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 --ThxMAG ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P incoming route for DID
Thanks Eric, I didn't know that. The general answer I've always seen regarding analog lines was that they supported CID only, and never sent DID. Good to know... Thanks,Alex On 10/12/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Alex Robar wrote: Analog routes (ie. copper telco lines) do not have DID information on them. Only digital lines (PRI, often VoIP DID) have this information sent alongside the call.Analog lines in the USA can support DID, but only using things like EM Wink which the Digium cards do not support.___--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 -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unauthenticated calls
The way that I've done it is to set the context= line under [general] in sip.conf to a context that just gives the congestion command and hangs up the call, something like this:exten = s,1,Answerexten = s,n,Wait(2) exten = s,n,Congestionexten = s,n,HangupI suppose you could really just use Hangup instead, but this seems to work for me.AlexOn 10/12/06, Mark Quitoriano [EMAIL PROTECTED] wrote: Hi list,i noticed from the cli my asterisk box is accepting unauthenticated calls how can i prevent this?CLI:-- Accepting UNAUTHENTICATED call from 192.168.0.2: requested format = gsm, requested prefs = (), actual format = ulaw, host prefs = (g729|ulaw|alaw), priority = mine ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you like TrixBox?
That's not really a fair criticism... Any kind of documentation about FreePBX will tell you that you need to put hand-coded dialplans into xxx_custom.conf, or the GUI will overwrite the changes.Alex On 10/13/06, Klaverstyn, David C [EMAIL PROTECTED] wrote: I don't mind Trixbox. The early version did have bugs but 1.2.2 seems to be pretty good. I do prefer not to use Trixbox and use plain Asterisk as you get more control. I have modified conf files in Trixbox, just to have Trixbox overwrite my changes. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Chris Ramsey Sent: Friday, 13 October 2006 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How do you like TrixBox? So I'm sure many of you are using or have tried to use TrixBox. Thus far, I'm in love with it. I haven't had a single snag. Then again, I don't need to get into anything overly nitty gritty with my Asterisk box. What are your views? -- Want a free copy of TrixBox Made Easy? Read the contest rules here. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P incoming route for DID
Analog routes (ie. copper telco lines) do not have DID information on them. Only digital lines (PRI, often VoIP DID) have this information sent alongside the call. What you need to do is change the context for each port you want redirected in zapata.conf, like so:signalling=fxs_kscontext=from-pstngroup=0channel = 1signalling=fxs_ks context=redirect-context group=0 channel = 2Cheers,AlexOn 10/12/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I am an asterisk newbie. I have successfully installed asterisk on Freebsd. The problem I am having is when I try to route based upon incoming DID. CALLERID(dnid) nor CDR(dst) have a number in them. Please help. Thanks ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [EMAIL PROTECTED] problems
Within FreePBX, under the tools menu, there is an Asterisk CLI module. Select that one and type sip show peers when one of the phones isn't working. Paste the output that you get back here.Alex On 10/9/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, Alex...thank you for your response How do you do that, at the Portal or using a dos command? Thanks again. Ed ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [EMAIL PROTECTED] problems
FreePBX is the successor to the Asterisk Management Portal, a part of [EMAIL PROTECTED] I don't know what version that is, so you might not even have the module I'm talking about. From the command shell, type asterisk -r, which will drop you to another prompt. From there, type sip show peers and paste the output. AlexOn 10/9/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Alex...I do not have FreePBX. What I have is this: http://70.89.124.237/ Ed ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [EMAIL PROTECTED] problems
Ed,Do the phones lose their registration? If you run sip show peers when the phones are not working, do they show as being registered or not?AlexOn 10/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thanks for your response. No, there;s no firewall and they are all correctly connected to the LAN. They work just fine, and then, one or two days later and out of the blue, they start having problems. Ed ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] swap CID with DID
Michael,You should be able to just do this: Set(CALLERID(num)=${DNID})... Though the VoIP-Info page is very vague about the DNID variable. You might try it out though.Best of luck!Alex
On 10/6/06, Michael Sampson [EMAIL PROTECTED] wrote:
Does anyone have a way to send the DID in place of the CID number. Iwant pop a web page with the DID in the URL but all the software I haveseen only supports putting the CID info in the URL. If I could swap the
two I could just use the programs as is. The two programs I have lookedat so far are SNAP and HUDlite. They both pop based on CID. SNAP worksvery well for that.Unless anyone knows of any software that can connect to asterisk, in any
method, and pop a web page when a call comes in and pass the DID intothe URL.--Michael SampsonInformation Systems ManagerCustomer Contact Services[EMAIL PROTECTED]
952-936-4000___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar
[EMAIL PROTECTED]
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