Re: [asterisk-users] Changing labels on Phones
On Sun, 15 Nov 2009, Leif Madsen wrote: However, changing the label is probably not really the right way to go about this. For example, I have created an Asterisk system for a call centre that uses hot desking with Polycom phones, and those phones then use the built in web browser with an auto-refresh rate that contacts a website (internal) that runs a PHP script that returns the currently logged in person. What model Polycom phones are you using? Any hints on the XML conf you used to get the broswer to start up? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI and paging
Hello, I have an AGI (in C) on 1.4.26.3 that puts a caller on hold, does a few things, then blind transfers the call (with EXEC Dial...) to a parking space. This is working fine. Now I want to add an overhead page AFTER the transfer has happened, basically announcing that there is a caller waiting. Trouble is the channel is gone, and my EXEC Page... is returning -1. I had to trap SIGALRM to keep the channel teardown from killing the AGI program in the first place (though I tried DeadAGI first - it still gets a SIGALRM when the Dial is complete). It seems that I should be able to do the EXEC Page as a new channel... why is it not allowing it? Also open to any other implementation ideas. The AGI has an AMI connection open and I suppose I could try something via AMI for the page? Thanks for any clues! j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
Next question will be How can I keep my server from crashing? :) (add more RAM... which may have been a good answer for question 1...) j On Tue, 24 Nov 2009, Alex Balashov wrote: Disable swap space. swapoff -a Jerry Geis wrote: Is there a way to keep asterisk in RAM and tell linux not to swap it out (ever). There are times when delays are noticed and I presume its due to linux swapping out the program. As if I call right back in then everything responds right away. Wait awhile and the same thing might occur. How can I keep asterisk always in RAM? I use CentOS 5. 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 Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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] keep asterisk in RAM
On Tue, 24 Nov 2009, Richard Kenner wrote: On a closely related note, has anyone built a normal (not embedded) system on SSD? I've been running Asterisk on a 20GB SSD drive for a while now. What mft/model? I was recently quoted a 4GB Compact Flash drive as part of a small system we plan to run asterisk on. Loosely tieing this to the recent thread on swap configuration, assuming a small number of SIP phones and no PSTN hardware, we were planning on 1GB of RAM to avoid swapping to this CF device. I know that CF cards have a limited number of writes before frying. If we keep it from using swap am I really only concerned about voicemail and logs? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk trunk CURL hangs in the dialplan
On Tue, 24 Nov 2009, Eric Chamberlain wrote: On Nov 24, 2009, at 6:17 AM, Tilghman Lesher wrote: Sounds like your local DNS resolver isn't answering queries promptly. Thanks, I'll look into it. Our CURL function only calls one hostname over and over. Would setting CURLOPT dnstimeout be of use in this situation? Put that host in /etc/hosts if it is static and under your control... j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI and Music on hold
Hi, Happy Thanksgiving to those of us in the USA... Been trying to debug an AGI (in C) on 1.4.26.2. I blind transfer a call to this snippet of dialplan: exten = 00,1,DeadAGI(pq.agi,50) pq.agi then plays a prompt (which I hear just fine): [Nov 26 02:42:47] VERBOSE[28721] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/pq.agi [Nov 26 02:42:47] VERBOSE[28721] logger.c: -- Playing 'you-are-caller-num' (escape_digits=) (sample_offset 0) [Nov 26 02:42:48] VERBOSE[28721] logger.c: -- IAX2/w2bdialplan-239 Playing 'digits/2' (language 'en') It then tries to put the caller on hold with the AGI command SET MUSIC ON. The AGI logs to syslog, and I see this: Nov 26 02:42:49 hades pq.agi[28725]: JAGI.c:161: AGI command(timeout=1000) : SET MUSIC ON Nov 26 02:42:49 hades pq.agi[28725]: JAGI.c:191: AGI response: 200 result=0 And I see this in /var/log/asterisk/full: [Nov 26 02:42:49] VERBOSE[28721] logger.c: -- Started music on hold, class 'default', on IAX2/w2bdialplan-239 [Nov 26 02:42:51] WARNING[28273] channel.c: Exceptionally long voice queue length queuing to IAX2/w2bdialplan-239 [Nov 26 02:42:51] WARNING[28268] channel.c: Exceptionally long voice queue length queuing to IAX2/w2bdialplan-239 [Nov 26 02:42:51] WARNING[28267] channel.c: Exceptionally long voice queue length queuing to IAX2/w2bdialplan-239 The Exceptionally long voice queue... continues as long as the hold music should be playing, but I hear nothing. 30 seconds later the AGI turns the music off: Nov 26 02:43:19 hades pq.agi[28725]: JAGI.c:161: AGI command(timeout=1000) : SET MUSIC OFF Nov 26 02:43:19 hades pq.agi[28725]: JAGI.c:191: AGI response: 200 result=0 Nov 26 02:43:19 hades pq.agi[28725]: JAGI.c:161: AGI command(timeout=1) : STREAM FILE pq_thanks-for-holding Nov 26 02:43:29 hades pq.agi[28725]: JAGI.c:191: AGI response: 200 result=0 endpos=80160 Nov 26 02:43:29 hades pq.agi[28725]: JAGI.c:161: AGI command(timeout=1) : STREAM FILE pq_you-are-currently-caller-number Nov 26 02:43:33 hades pq.agi[28725]: JAGI.c:191: AGI response: 200 result=0 endpos=29440 Nov 26 02:43:33 hades pq.agi[28725]: JAGI.c:161: AGI command(timeout=300) : SAY NUMBER 2 Nov 26 02:43:33 hades pq.agi[28725]: JAGI.c:191: AGI response: 200 result=0 Nov 26 02:43:33 hades pq.agi[28725]: JAGI.c:161: AGI command(timeout=1000) : SET MUSIC ON Nov 26 02:43:33 hades pq.agi[28725]: JAGI.c:191: AGI response: 200 result=0 I hear the prompts that show above, and it attempts to turn the music back on (with a 200 result=0 response from asterisk), but again I hear no hold music, and again the Exceptionally long voice queue... messages start streaming in /var/log/asterisk/full. I tried applying the patch from issue number 16268 with no effect. I read the comments for issue number 15609, which is still open, and don't really know if it is relevant, since I am not experiencing a crash - just no hold music! If I change the AGI command, in the exact same place, to do this instead: EXEC MusicOnHold default I hear the hold music just fine, but of course this command doesn't return, and is not useful within my AGI! I'm pretty stumped and don't know what to try next. Any suggestions? Thanks, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1800 DID Provider - Suggestion
Try IPComms. j On Fri, 27 Nov 2009, Marco Cordeiro wrote: Hello All, Do you guys suggest any 1800 DID Provider in the US ? I'm having a hard time to find one. Thanks, Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] network config
Slightly OT? A client has two offices in the Virgin Islands that MUST maintain data connectivity, and there are no available leased line options to run a P2P link between them. To date, broadband Internet connections at both offices have been used as the link, with a VPN tunnel, and phones in one location use the tunnel (Sonicwall) to talk with asterisk at the other location. Although this functions well, it only takes an (unfortunately frequent) hiccup to lose calls and/or severely impact quality. The client has decided to get a second Internet connection at both sites, and use the Sonicwall or any other possible firewall to manage the tunnel over both links, such that the phones won't know what link is being traversed, or (hopefully) that a link has gone down. So the first question is - has anyone attempted anything similar and made it work? Do you lose an in progress call when the tunnel switches from one link to the other? And finally - is there a device that will manage the tunnel such that a high water mark of latency will also cause the tunnel to switch to the other link, rather than actual packet loss? Thanks for any tips, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network config
Hi David, On Tue, 8 Dec 2009, David Gibbons wrote: snip A client has two offices in the Virgin Islands that MUST maintain data connectivity, and there are no available leased line options to run a P2P link between them. snip Is there line of sight? I've been wanting to do a long-shot wifi link and my company would give it a shot if you want :). Sadly no, because cruise ships park (dock?) directly in front of one the locations, which is directly between them. Worse high intensity radar blasts seem to give any kind of wireless signal we have attempted lots of trouble. If it weren't for the ships, this would work well I think, but as its happens the ships are the source of the client's revenue! snip And finally - is there a device that will manage the tunnel such that a high water mark of latency will also cause the tunnel to switch to the other link, rather than actual packet loss? See above. Fail-over routers have to wait some criteria are met in order to fail over (ping latency, ping loss, etc). This means that the connection you're using as the 'default' WILL go 'down' BEFORE it switches to the other one, regardless of the criteria used. Hmm, an excellent point. I suppose some amount of tweaking might cause the switch to happen before asterisk or the endpoint decides that the call is lost? Are these SIP timers that we might play with? Some amount of silent interruption might be tolerated during a switch, but a lost call is hard to accept. Another plan would be to set up two routers at the site with two separate VPN tunnels across the two different links, both tunnels being always on. You could then use a SIP proxy or iptables magic to choose which tunnel was the best at any given time. Hmm, another good thought. Now its getting complicated :) I would go for the wifi. Maybe because I want to do a long-shot link. Also because I want to go to the virgin islands :). Heh. Come on down! Water is fine... j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA FXO
On Fri, 11 Dec 2009, Joseph wrote: On 12/11/09 14:05, Jonathan Thurman wrote: On Fri, Dec 11, 2009 at 12:44 PM, Connor Spiess cspi...@idea-ma.com wrote: Joseph You could also check out the Audio Codes gateways if the Grandstream doesn't work out for you. They make FXO/FXS gateways. They were reliable boxes for us but this was to a non-asterisk PBX over MGCP. I mention them cause I know they make a SIP based one. We use AudioCodes MP-114 2FXS/2FXO and they have been rock solid. I have a bunch used for faxing connected back to Asterisk over SIP. I will say that I have had a LOT of issues with faxing on the larger GrandStream GXW-4024s and had to replace them. I put a AudioCodes MP-124 in and have had no complaints since. -Jonathan Thank for suggestion. Well, it is not that cheap but the problem with their equipment is luck support and decent manual. Whatever I google about AudioCodecs everybody seems to be straggling with the setup; I don't think this should be that hard to write a decent instructions if they want to sell their product. Maybe they have a good product but without support it will not mean much. eg.: http://www.trixbox.org/forums/trixbox-forums/trunks/trixbox-2-2-and-audio-codes-mp-114-fxo-setup Well now that you have shot down just about every decent piece of hardware that has been suggested, you are probably left with designing your own! I totally disagree with your comments on Audiocodes... excellent product. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote: Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows (ULAW) (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody suggest what I might be overlooking? In the sip.conf entry for your peer you need to specify the codec negotiation order. Though you put g.729 first on the phone, asterisk probably has ulaw first, and is taking precedence. In the sip.conf entry put this: disallow=all allow=g729 j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote: O.K., interestingly enough when I call our extensions from my mobile phone it still seems to be using ULAW, but when they dial out it seems to be using G.729 now. Is there something in Dahdi that I need to configure so that inbound calls (from the PRI on a Digium TE205) use G.729 to get to the phones too? A Dahdi channel over a PRI will always be ulaw - that is the encoding on the PRI (at least in the US). If your phones are using G.729 then transcoding will be taking place within asterisk for the bridge between the channels. My guess is you are looking at the PRI channel. There should be another channel for the phone. That should always be G.729 now. Cheers, j Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of j...@jeff.net Sent: Tuesday, December 15, 2009 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a restart when convenient pending. Is that sufficient or do I have to actually reboot the server for the change to take effect? Just do a sip reload at the asterisk CLI prompt and you will be good to go. It won't cutoff any calls in progress. Then reboot your phone. Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID on Indian PSTN is not working.
On Wed, 6 Jan 2010, Arun Sasidhar wrote: Hi, I dont know the type of caller ID. What you mean by this?. I am from India. I don't know more about this. * Thanks, Arun S* Hi Arun, Just for fun I read over the bug id you quoted below, and it seems there are a number of settings you may need to try to get your particular situation working. Have you done this? You may need to open another issue on Mantis if not. It seems India is not very consistent with its CID methods. It also seems from the issue quoted that there are hardware dependancies. If you are using hardware and a provider that is quoted by the issue as resolved, you should be fine, as the changes were comitted to 1.4 a long time ago (assuming you set your cid options correctly). j On Wed, Jan 6, 2010 at 4:40 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote: Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow. Is this patch will resolve my issue? Kindly help me to fix this issue. What type of caller ID is used in that line? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Arun S System Administrator. Cabot Solutions www.cabotsolutions.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] Beginners Guide to setting up a Call Centre
On Tue, 12 Jan 2010, Danny Nicholas wrote: Since you are small, trixbox would probably be the ideal flavor of Asterisk for you. It is a downloadable ISO that installs Scientific Linux and Asterisk and sets you up to manage everything with a GUI interface from a browser. Once you outgrow that, you can either expand it, go for Commercial Asterisk or join the fun world of Open Source Asterisk where we work on releases and/or SVN branches. I agree that FreePBX would be the ideal flavor for him, but I am a recent convert to Elastix. Much tighter GUI, more included stuff (like hylafax and iaxmodem), and just overall a better stab at the whole integration. After two horrid experiences with Trixbox Pro and my impression of Elastix over Trixbox CE I will never install another Fonality product. j -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Childs Sent: Tuesday, January 12, 2010 4:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Beginners Guide to setting up a Call Centre This is currently still at a proof of concept stage. After being mis-sold a Alcatel phone system, that does None of the things we asked for (Ok it takes calls but that's about it) We are looking at alternatives to try and bring some of the features we previously had on our old Analogue STS phone system. Looking at all the docs I can find Asterisks looks like it should be able to do the job and a whole lot more. This is for a small call centre so ideally we want all the features of an average call centre, ACD, Call Recording, Queue's etc etc. Any pointers on how to get started would be most helpful. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
On Tue, 12 Jan 2010, Richard Kenner wrote: And, I'd be in the camp that would advocate getting your hands dirty and learn to program without the GUI. You'll learn a lot and then if you'd want to move to a GUI and something breaks, you'll have some idea on what and how to fix it. Knowing now what I do, I find a GUI to restrictive. I agree. I originally felt I wanted the GUI approach too, but then when I looked into things in more detail and understood that you really can't BOTH use the GUI approach and edit files explicitly, I decided that the GUI did nothing for me except add a additional level of complexity and that I'd be MUCH better off just doing things directly. That is so not true. FreePBX has hooks in a million places to do custom dialplan stuff - I do it all the time. I also link in custom AGI/AMI applications, custom provisioning, custom LCR, and am even working with one customer that has mastered making FreePBX multi-tenant. If you want to get your hands dirty there is plenty of dirt underneath FreePBX. On the other hand, if you want a simple setup that is easily managed, the GUI is fantastic and saves a LOT of time. And if you are a PHP programmer you can easily modify the operation of any part of it. Your comments both come from having taken a short look at FreePBX and dismissed it without investigating how powerful it can be. Now as far as Switchvox goes, now THAT is a restrictive platform. You cannot ssh into the box for starters. Every extension requires a license. There is no support for dual homing the box (my default installation configuration - one port on public!). Another horrid experience. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
On Tue, 12 Jan 2010, Richard Kenner wrote: Your comments both come from having taken a short look at FreePBX and dismissed it without investigating how powerful it can be. Yes, but the discussion is about COMPLEXITY, not power! I thought the discussion was about how an IT guy with no previous asterisk experience could get up and running the fastest. By FAR that answer is to use one of the pre-packaged installations such as TrixBox or Elastix. Sure, there are hooks where you can do anything you want, but if you were to set up identical configurations via FreePBX and by writing a dialplan (and other config files) from scratch, the latter will be the least complex. By whose estimation? To even get that far with asterisk requires a lot of reading and experience. It took me several weeks to get my first installation answering the phone in 2003, before there were any serious GUIs available. My first intallation of aster...@home, however, was answering the phone in about 2 hours. What that means is that if your goal is to learn the least about Asterisk that you can get away with, but that you expect to need to tweak the dialplan, doing so is going to have a lower learning curve if you JUST use Asterisk: using FreePBX just means that you have to learn BOTH systems and that you'll be modifying a more complex configuration than if you did it yourself. The thing is the OP probably won't need to tweak the dialplan to do what he needs to do. My take is this - if you want to get started with Asterisk and you have NO experience, a pre-built package like Asterisk NOW, PIF, Trixbox, or Elastix is the quickest and cleanest way to get setup and running. After having some experience with it and finding the things that may require some custom dialplan work (getting harder and harder to find given the most recent releases of FreePBX and the things possible from the GUI), you can then learn the internals of dialplan coding and work that out over time. For someone starting from scratch, learning to setup Asterisk properly and coding your first diaplan - even using the samples - is difficult and non-intuitive. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxmodem / hylafax receive problem
On Thu, 14 Jan 2010, Doug Lytle wrote: Kingsley Tart wrote: Hi, I'm trying to receive faxes using hylafax / iaxmodem but I just can't get it to work. We're using Sangoma E1 cards and have calls coming in Without seeing your config files for iaxmodem and hylafax and also seeing a dialplan snippet on how you're launching calls from Asterisk to hylafax, it's not going to be easy to help you. Doug Actually it is fairly clear that his dialplan is correctly routing the calls to iaxmodem, and that iaxmodem is simply not completing the training. I would say that the fax machine you are testing with is either on a horribly noisy POTS line, is on a VoIP line and you don't realize it, or is one of the fax machines that is just incompatible with hylafax class 1 support (have run into a few of those). The suggestion to try from other fax machines is a good one. I suppose that is one piece you could tell us, though. Are you forcing class 1 support? Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi issues
Hello,
My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port
modular card and a single FXS module.
Got the Rhino card installed and the machine sees it:
r...@pbx:/etc/dahdi# dmesg | grep rcbfx
[ 71.985309] rcbfx :04:00.0: PCI INT A - GSI 21 (level, low) - IRQ
21
[ 71.985440] rcbfx 1: Rhino PCI BAR0 5010 IOMem mapped at
c90008d7c000
[ 71.985504] rcbfx 1: Waiting for response from card .
[ 71.986276] rcbfx 1: Firmware Version 2.1
[ 71.986288] rcbfx :04:00.0: firmware: requesting rcbfx.fw
[ 72.047192] rcbfx 1: firmware rcbfx.fw not available from userspace
[ 72.047202] rcbfx 1: Hardware version 11
[ 72.047233] rcbfx 1: G168 07 08 DSP Loader file size = 170 App file
size = 48414
[ 72.350080] rcbfx 1: G168 DSP Ping DSP Version 106
[ 72.510185] rcbfx 1: G168 DSP Active and Servicing 2 Channels - 3
[ 72.510681] rcbfx 1: Starting DMA
[ 72.530147] rcbfx 1: Spotted a Rhino: Rhino RCB8FXX (1 modules)
r...@pbx:/etc/dahdi#
Dahdi also sees it:
r...@pbx:/etc/dahdi# lsdahdi
### Span 1: Rhino RCB8FXX/1 Rhino RCB8FXX/1 (MASTER)
1 FXSFXOKS (SWEC: MG2)
2 FXSFXOKS (SWEC: MG2)
3 EMPTY
4 EMPTY
5 EMPTY
6 EMPTY
7 EMPTY
8 EMPTY
r...@pbx:/etc/dahdi# dahdi_cfg -vvv
DAHDI Tools Version - 2.2.0
DAHDI Version: 2.2.1-rc2
Echo Canceller(s): MG2
Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
2 channels to configure.
Setting echocan for channel 1 to mg2
Setting echocan for channel 2 to mg2
r...@pbx:/etc/dahdi#
I am running FreePBX, so it created /etc/asterisk/zapata_additonal.conf,
so I linked /etc/asterisk/chan_dahdi.conf to it:
r...@pbx:/etc/asterisk# ls -ltr {chan_dahdi,zapata_additional}.conf
-rw-rw-r-- 1 asterisk asterisk 678 2010-01-13 21:27 zapata_additional.conf
lrwxrwxrwx 1 asterisk asterisk 22 2010-01-13 23:11 chan_dahdi.conf -
zapata_additional.conf
r...@pbx:/etc/asterisk#
r...@pbx:/etc/asterisk# cat chan_dahdi.conf
;;
; Do NOT edit this file as it is auto-generated by FreePBX. All
modifications to ;
; this file must be done via the web gui. There are alternative files to
make;
; custom modifications, details at: http://freepbx.org/configuration_files
;
;;
;
;;[20001]
signalling=fxo_ks
pickupgroup=
mailbox=20...@device
immediate=no
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callgroup=
callerid=device 20001
busydetect=no
busycount=7
accountcode=
channel=1
;;[20002]
signalling=fxo_ks
pickupgroup=
mailbox=20...@device
immediate=no
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callgroup=
callerid=device 20002
busydetect=no
busycount=7
accountcode=
channel=2
r...@pbx:/etc/asterisk#
BUT asterisk doesn't seem to see it:
r...@pbx:/etc/asterisk# asterisk -rx 'dahdi show channels'
Chan Extension Context Language MOH Interpret
r...@pbx:/etc/asterisk#
Any clues? I am sure I am missing something stupid.
Thanks,
j
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Re: [asterisk-users] Dahdi issues
On Thu, 14 Jan 2010, Danny Nicholas wrote:
I'm on 1.4.26.2 and have to have DAHDI entries in user.conf for Asterisk to
see the DAHDI line (dahdi_genconf users).
Hmm, that would seem to coincide with the entries I already have in
chan_dahdi.conf. I did it anyway, but unfornuately there is no change.
Still no channels to show in asterisk. Interesting that it just randomly
decided to create extensions 4000 and 4001 for my two channels :)
j
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Thursday, January 14, 2010 9:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dahdi issues
Hello,
My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port
modular card and a single FXS module.
Got the Rhino card installed and the machine sees it:
r...@pbx:/etc/dahdi# dmesg | grep rcbfx
[ 71.985309] rcbfx :04:00.0: PCI INT A - GSI 21 (level, low) - IRQ
21
[ 71.985440] rcbfx 1: Rhino PCI BAR0 5010 IOMem mapped at
c90008d7c000
[ 71.985504] rcbfx 1: Waiting for response from card .
[ 71.986276] rcbfx 1: Firmware Version 2.1
[ 71.986288] rcbfx :04:00.0: firmware: requesting rcbfx.fw
[ 72.047192] rcbfx 1: firmware rcbfx.fw not available from userspace
[ 72.047202] rcbfx 1: Hardware version 11
[ 72.047233] rcbfx 1: G168 07 08 DSP Loader file size = 170 App file
size = 48414
[ 72.350080] rcbfx 1: G168 DSP Ping DSP Version 106
[ 72.510185] rcbfx 1: G168 DSP Active and Servicing 2 Channels - 3
[ 72.510681] rcbfx 1: Starting DMA
[ 72.530147] rcbfx 1: Spotted a Rhino: Rhino RCB8FXX (1 modules)
r...@pbx:/etc/dahdi#
Dahdi also sees it:
r...@pbx:/etc/dahdi# lsdahdi
### Span 1: Rhino RCB8FXX/1 Rhino RCB8FXX/1 (MASTER)
1 FXSFXOKS (SWEC: MG2)
2 FXSFXOKS (SWEC: MG2)
3 EMPTY
4 EMPTY
5 EMPTY
6 EMPTY
7 EMPTY
8 EMPTY
r...@pbx:/etc/dahdi# dahdi_cfg -vvv
DAHDI Tools Version - 2.2.0
DAHDI Version: 2.2.1-rc2
Echo Canceller(s): MG2
Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
2 channels to configure.
Setting echocan for channel 1 to mg2
Setting echocan for channel 2 to mg2
r...@pbx:/etc/dahdi#
I am running FreePBX, so it created /etc/asterisk/zapata_additonal.conf,
so I linked /etc/asterisk/chan_dahdi.conf to it:
r...@pbx:/etc/asterisk# ls -ltr {chan_dahdi,zapata_additional}.conf
-rw-rw-r-- 1 asterisk asterisk 678 2010-01-13 21:27 zapata_additional.conf
lrwxrwxrwx 1 asterisk asterisk 22 2010-01-13 23:11 chan_dahdi.conf -
zapata_additional.conf
r...@pbx:/etc/asterisk#
r...@pbx:/etc/asterisk# cat chan_dahdi.conf
;---
-;
; Do NOT edit this file as it is auto-generated by FreePBX. All
modifications to ;
; this file must be done via the web gui. There are alternative files to
make;
; custom modifications, details at: http://freepbx.org/configuration_files
;
;---
-;
;
;;[20001]
signalling=fxo_ks
pickupgroup=
mailbox=20...@device
immediate=no
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callgroup=
callerid=device 20001
busydetect=no
busycount=7
accountcode=
channel=1
;;[20002]
signalling=fxo_ks
pickupgroup=
mailbox=20...@device
immediate=no
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callgroup=
callerid=device 20002
busydetect=no
busycount=7
accountcode=
channel=2
r...@pbx:/etc/asterisk#
BUT asterisk doesn't seem to see it:
r...@pbx:/etc/asterisk# asterisk -rx 'dahdi show channels'
Chan Extension Context Language MOH Interpret
r...@pbx:/etc/asterisk#
Any clues? I am sure I am missing something stupid.
Thanks,
j
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[asterisk-users] Dahdi and FreePBX
Perhaps this more belongs on the FreePBX list, but for the archives, this is what I did to make it work: chan_dahdi wants to read /etc/asterisk/chan_dahdi.conf FreePBX, at least how I installed from source, seems to think I am still running Zaptel. It created zapata_additional.conf when I added two ZAP channels. For some unknown reason it did NOT create zapata.conf, although the sample was still in /etc/asterisk. I blindly linked chan_dahdi.conf to zapata_additional.conf, but that failed (and had I looked in /var/log/asterisk/full before posting earlier I would have seen why), because zapata_additional.conf has no '[channels]' context identifier, as it is really just meant to be included by zapata.conf. So the solution, barring trying to figure out why FreePBX is still using Zaptel filenames, is to create a chan_dahdi.conf that looks like this: [channels] language=en #include zapata_additional.conf Or a simple soft link to zapata.conf, containing the above, would also have worked. Then a quick restart of asterisk and all was well. No need for a users.conf. Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GXV3140 and Xlite video
On Thu, 14 Jan 2010, Julian Lyndon-Smith wrote: Has anyone managed to get these two phones to make a video call to each other ? If so, care to share how the hell you managed ? the GXV is at the latest firmware, and xlite the latest download Asterisk 1.4 trunk TIA Julian I've done it. I don't recall there being much special about it. Have to make sure you have videosupport turned on in sip.conf, and that each peer has h264 in its codec list. Can you make two GXV3140s talk to each other with video? j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
On Fri, 15 Jan 2010, randall wrote: Sure. My point was just that IF you only got one connection in the wall, its cheaper to get a switch than getting a phone with dual 1Gbit ports. Leif OK, point taken. but i have 6xisdn2 and already 2x24 gigabit switches (will need to replace one with a PoE version ) these connections include both desktops and current phones. i was just hoping to cut back the amount of cabling with 50%, and when i found out that most phones with 10/100/1000 connection cost about 250,- euro's a piece instead of 90,- for a decent version with 10/100 it was a real bummer, it would mean about doubling my budget. I'm not sure you get it - he is saying you can eliminate the extra cable run to the desk, and place a small 5 port gigabit switch under the desk and drive both your PC and the phone from it. Total cost per desk - 90 + 17 euros. Significantly less than 250 euros for a dual gigabit port phone. No change to your switching infrastructure in your machine room. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
On Fri, 15 Jan 2010, Hans Witvliet wrote: If you connect your pc with GB-lan card to an dual-ported ip-phone, you and up with an 100Mbps lan connection to your pc. Only way to avoid that, is to insert a cheap second lan-card in your pc, and connect your phone to the second lan, so your pc will act as an switch, instead of your phone... I'm curious - how have you managed to connect a second LAN card and have it bridge your (presumably onboard) ethernet? Does Windows have such capability? But I guess the OP was running XUbuntu, and though relatively complicated I guess you could get it to do that. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual Asterisk Installation
On Thu, 21 Jan 2010, Gergo Csibra wrote: Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote: Forget about virtualization! ... Virtualisation is nice for test-setups, but thats it. for any real job it's a major pain in the ass and makes stuff bork beyond imagination. Well. Why do you use computer? There're slide-rule. You can calculate anything with that... Pretty crappy analogy. Just because you *can* do something doesn't mean it is production ready. But then the OP said it wasn't all that important, so I would say go Xen and tell us how it works out. I think you will only have trouble with conferencing, and maybe not even then if the machine is beefy enough and unloaded. Monitoring servers are usually pretty unloaded. I'm playing a lot with OpenVZ, but you won't have access to your PSTN hardware... at least I haven't been able to make that part work. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mitel integration
Hi, A potential client (hotel) has a Property Management System that talks the Mitel protocol to their current Mitel PBX in order to receive CDRs (which end up being rated by the PMS system and charged back to guests). Does anyone know of any (free or otherwise) docs on this protocol, or better still have experience interfacing asterisk in a hotel situation like this? The PMS developers claim that the Mitel spec is proprietary, and that they cannot give it to me, and are basically unwilling to try and develop a method with us to integrate directly. Funny enough they also claim that just about every traditional PBX emulates this protocol for integration with PMS systems, so they say that if I can manage to do the same I will instantly integrate with MANY PMS systems. Sounds good to me, but without the spec I'm stuck in a catch 22! Thanks, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mitel integration
On Wed, 27 Jan 2010, Mark Wiater wrote: the mitel 3300 sends SMDR on TCP 1752. It spews software and hardware logs in the same manner, different ports. This particular model (need to get the model number) has a serial connection. I'm all for putting a serial sniffer between them (if they let me!), but was really hoping someone had already done this and could give me a headstart. I'll investigate the ethernet options, though, as that would make more sense anyway! If the PMS will talk over ethernet I'll try to pretend to be a 3300. Cheers, j On 1/27/2010 11:00 AM, Steve Howes said: On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote: Sounds good to me, but without the spec I'm stuck in a catch 22! tcpdump? (assuming IP). Bet its fairly simple plain text or something. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec conversion
On Tue, 2 Feb 2010, Steve Edwards wrote: On Tue, 2 Feb 2010, wassim darwich wrote: Thanks for?your reply,ill give?you my situation, iam using my asterisk box as a switch ,so my client is sending me ulaw and my voip provider?only accept g723 ,So what i have to do is to receive?g711?codec and convert them to g723 at?asterisk ,i tried this before but i saw the cpu?usage is overloaded when doing conversion ,Iam using asterisk 1.4.22 ,So what do you advice me. Get your client to switch to g723 or your provider to switch to ulaw. If that is not possible, get more CPU resources: 1) Eliminate any unrelated processes (X, Apache, MySQL, Tetris). Make sure Asterisk is running with elevated priority. 2) If your other processes (AGIs?) are written in scripting languages (Perl, PHP), re-code them in compiled languages (C). 3) Use more powerful processors (faster clock, more cores, more processors). 4) Split the load across multiple hosts. This has the added advantage of not putting all your eggs in one basket -- you can take a host out of service for maintenance or upgrades. 5) If you are swapping, more RAM may help. Don't forget the fancy Digium codec translator card thingy! j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk video support and IPTV
Has anyone played with the idea of Asterisk as an H.264 multicast tool? I am wondering what the possibility would be to have some kind of machine with a capture card call asterisk over SIP and have asterisk make another hundred calls to subscribers. Then any H.264 compatible device (Android? Set top boxes? Plugin to MythTV?) would be able to receive a video/audio stream. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goes down?
On Fri, 5 Feb 2010, Nikhil Nair wrote: Hi again, OK, I've now installed a local caching nameserver, but don't see any change at all. IN detail, what I did: - Installed Debian packages resolvconf and dnsmasq (resolvconf just takes care of dynamic nameserver allocations in /etc/resolv.conf). - After looking at the docs, edited /etc/network/interfaces to add a dns-nameservers line in the entry for eth1. Then reconfigured resolvconf. - Checked /etc/resolv.conf: now showing 127.0.0.1 as the only nameserver. - Tested name resolution in general: working fine. - Turned ADSL router off and tried to make local and Zap calls: no luck. - Rebooted machine and tried again: still no luck. Again, the logs indicate that Asterisk thinks the SIP phones are unreachable. Was there anything special I needed to do with the setup of dnsmasq, or its interface with Asterisk? If not, I'm stuck again. Thoughts? Nikhil. Hi, I am stepping out on a limb here, since I have never run dnsmasq, but I don't think it is an actual caching server. I think it just relays queries to upstream servers, which in your case are still unreachable, and will still cause asterisk to timeout waiting for a reply. You need a true local DNS server that can answer for your asterisk box and any named phones. A caching server should do also, assuming that your link is up long enough to serve and cache a few local queries before it goes down - pretty much how most of my systems run. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goes down?
On Fri, 5 Feb 2010, Vinícius Fontes wrote: I solved similar issues by setting srvlookup=no, having bind running locally and just the line nameserver 127.0.0.1 on /etc/resolv.conf. Your local bind is what solved the problem. The srvlookup=no didn't actually help IMO. j Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Randy R randulo2...@gmail.com escreveu: 2010/2/5 Vinícius Fontes vinic...@canall.com.br: Have you tried to set srvlookup=no on your sip.conf? I think that just stops SRV lookups, not regular DNS. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] large scale paging
On Fri, 5 Feb 2010, Mark Willis wrote: Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. How did you get that number? Even with ulaw @ 64Kbps you theoretically get 32Mbps. If you used G.729 you would cut that down to 4 or 5Mbps. Totally oversimplified, but that seems a lot more doable. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? I can't say I have ever pushed that hard, but that doesn't sound like it would be difficult to handle. There are plenty claiming they have 400 simultaneous two way conversations going on a single box. j Any suggestions or war stories are appreciated. Mark Willis Cartama Consulting LLC 210 698 5097 -- _ -- 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] Dial script
On Sat, 6 Feb 2010, Thomas Perron wrote: karl, does it make you feel good ? wow. pathetic. On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife karlf...@gmail.com wrote: Try this: #rm -rf / I second that opinion. Tell us first WHY you want to dial 1 numbers in sequence. Without any reason, you must be assumed to be a call spammer, and you are looking for help in the wrong place. j - Original Message - From: Thomas Perron thomas.per...@gmail.com To: asterisk-users@lists.digium.com Sent: Friday, February 05, 2010 8:54 PM Subject: [asterisk-users] Dial script Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] test
fail. On Tue, 9 Feb 2010, aster...@opensourcesolution.in wrote: test -- _ -- 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] IP Phone recommendation
On Wed, 10 Feb 2010, Tim Nelson wrote: - Gordon Henderson gordon+aster...@drogon.net wrote: If not using PoE I'd suggest getting a few extra PSUs though - that's one area I have had a few issues with - but maybe it's just been the UK ones. Gordon The same can be said for the US versions. My experience has been it's not a case of 'if' the PSU will fail, but 'when'. In a past (less intelligent) life, I deployed a fair number of the GXP2020s and GXP2000s. There are not very many of them left that haven't completely died(the phone itself), and of those left, they've all had power supplies replaced. I cannot speak for the quality of the later devices from Grandstream. After being burned, it's a bit hard to look at them again when there are so many other quality devices available (think Polycom, Aastra, etc). --Tim I haven't used any standard Grandstream IP phones, but I am *trying* to stabalize the new video phones they have come up with. I have several GXV3000 and GXV3140s. I got through central provisioning using their java based tool and for the most part these phones work, but have very odd bugs. If left to itself for more than a few days the 3140 simply stops answering calls. The 3000 has very odd DTMF issues - like doubling every digit pressed. This is all fine and I know they are new products, but what is frustrating is Grandstream's lack of support. The forums are next to useless, and the firmware releases are always coming very soon. Then there are my horrid experiences with their FXO gateways. Echo, bad audio in general, needing a reboot every few days, etc. Again, support is non existant. So regardless of the quality of the latest phones, the company itself leaves a lot to be desired IMO. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] video voicemail
Playing around with the Grandstream GXV3140. I'm interested in having the video voicemail clips emailed in a format that might be opened by Windows Media Player or even Quicktime. Have been googling around a lot and have tried various bits of OSS to read the resulting .h264 file that asterisk is saving, but having absolutely no luck. A video nut I know took a look at the file and said it had no header, and was actually convinced there was no video in it. Anyone else trying to do this? Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] video voicemail
On Mon, 15 Feb 2010, Tilghman Lesher wrote: On Monday 15 February 2010 14:09:38 Olle E. Johansson wrote: 15 feb 2010 kl. 20.31 skrev Jeff LaCoursiere: Playing around with the Grandstream GXV3140. I'm interested in having the video voicemail clips emailed in a format that might be opened by Windows Media Player or even Quicktime. Have been googling around a lot and have tried various bits of OSS to read the resulting .h264 file that asterisk is saving, but having absolutely no luck. A video nut I know took a look at the file and said it had no header, and was actually convinced there was no video in it. Anyone else trying to do this? Asterisk is not saving a proper h.264 file, it's saving the raw RTP media. I think that ffmpeg had a module that could handle this at some point in time. Because of patents for H.264, we can't convert the media to anything useful. IIRC, the actual format of the file is: 1-bit: full-frame marker 15-bits, unsigned: length of the RTP packet, in bytes RTP-data 1-bit: full-frame marker 15-bits, unsigned: length of the RTP packet, in bytes RTP-data (etc.) The format was designed to be easily convertable back into an RTP stream, because as the format does not include audio data, it was believed that it would never be useful outside of Asterisk's own usage. Am I naive in assuming that I could extract the RTP data given the format above into something that is inherently h.264 encoded? Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] time/date over POTS?
I had a customer ask me about time/date information being sent to his analog (attached to a Linksys SPA2102) answering machine. I didn't know that POTS could carry this information. Is this something Asterisk could send over SIP? Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time/date over POTS?
On Thu, 4 Mar 2010, Dave Fullerton wrote: Jeff LaCoursiere wrote: I had a customer ask me about time/date information being sent to his analog (attached to a Linksys SPA2102) answering machine. I didn't know that POTS could carry this information. Is this something Asterisk could send over SIP? Cheers, j Time and date info on a POTS line is part of the caller ID stream. It is up to the analog endpoint sending the caller ID stream to know the current time to send. Anything that works with SIP should also have NTP capabilities and should be getting its time using that. -Dave Aha. Sadly I know that the incoming calls from our PSTN provider (over RBS T1) do NOT carry caller ID, so what we are passing on via SIP to the Linksys box must also be missing the time info. Is there any way to add that to the outgoing call to the Linksys box? Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 30 mins GSM file
On Thu, 4 Mar 2010, Steve Howes wrote: On 4 Mar 2010, at 23:11, Steve Edwards wrote: On Thu, 4 Mar 2010, Steve Edwards wrote: On Fri, 5 Mar 2010, David @ULC wrote: I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way to create it ? Record yourself thinking of the solution for 1/2 of an hour. Use sox to concatenate 6.9 copies of John Cage's 4'33 Get permission first.. S Considering that was a 1950's era composition, perhaps the copyright has already expired? j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Switchvox SOHO 4.5 is Here
On Fri, 12 Mar 2010, Angelito Manansala wrote: If you are having trouble reading this email, read the online versionhttp://now.eloqua.com/es.asp?s=491e=78675elq=55426a8b6c714f5bb6f2bf4b5d37bf55 . http://app.en25.com/e/er.aspx?s=491lid=215elq=55426a8b6c714f5bb6f2bf4b5d37bf55 Dear Lito, *The information in this email is given to you in advance to make you aware of an impending product release announcement. You are obliged, under the terms of your NDA with Digium, to keep this information confidential until the Switchvox SOHO 4.5 release is announced on March 30, 2010.* So am I missing something or did you just blatantly disregard the above warning to honor your NDA? j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 peer question
What does the (T) mean? Am playing around with running an IAX trunk over an OpenVPN session and see this only on this peer. demopbx/sunfone 10.222.0.6 (D) 255.255.255.255 4569 (T) OK (26 ms) Same thing on the other side: sunfone/demopbx 10.222.0.1 (S) 255.255.255.255 4569 (T) OK (31 ms) Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 peer question
On Sat, 13 Mar 2010, Jeff LaCoursiere wrote: What does the (T) mean? Am playing around with running an IAX trunk over an OpenVPN session and see this only on this peer. demopbx/sunfone 10.222.0.6 (D) 255.255.255.255 4569 (T) OK (26 ms) Same thing on the other side: sunfone/demopbx 10.222.0.1 (S) 255.255.255.255 4569 (T) OK (31 ms) Doh! I think it means that I have included trunk=yes in the peer config, which I have only done on this peer :):) Sorry for the noise. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Android Phones ;-)
On Mon, 15 Mar 2010, Ishfaq Malik wrote: Conrad Wood wrote: FWIW, just received an android-based phone and after installing sipdroid found that it works very well with asterisk ;). It automatically dials numbers through asterisk if available and otherwise through the gsm network. Contacts integrate well too. No ties to any telco or to google, just a happy user ;) Conrad I did the same last week and agree totally, a nice little softphone, well integrated with the rest of the phone and took about 1 min to configure without looking at any instructions. -- Ishfaq Malik Software Developer PackNet Ltd So which Android phone, and is it using the GSM interface for the SIP traffic, or only if you are on wifi? Does anyone have a wimax android phone yet? j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
On Mon, 15 Mar 2010, David Backeberg wrote: and also to do LCR and Quality based routing of International calls? I don't know what that means. Least Cost Routing. Asterisk doesn't have anything built in for this. We do it with an in-house AGI. Others have done similar things that you might be able to buy. Try on asterisk-biz. The question I have is - why the Cisco? Assuming you have SIP or H.323 capable phones, just dump the Cisco and use the asterisk box for the whole shebang. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
On Mon, 15 Mar 2010, Mohit Saxena wrote: We are a mobile operator so has to work with the PSTN side E1s from the Mobile switch. This is the reason for using Cisco Media gateways. I know you may be stuck with them, but you could just as easily plug in a Digium/Sangoma/Rhino T1/E1 card (or Xorcom channel bank?) into your asterisk box and you would be able to accomplish the same thing, but in IMO a much more asterisk-friendly way. Can't help you with the Cisco config... you will need to post a lot more details about your asterisk config if you want help on that side. j Kindly help Br, Mohit C. Saxena I Data/ISP Department Starcomms Plc. 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709 email:moh...@starcomms.com www.starcomms.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Monday, March 15, 2010 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways On Mon, 15 Mar 2010, David Backeberg wrote: and also to do LCR and Quality based routing of International calls? I don't know what that means. Least Cost Routing. Asterisk doesn't have anything built in for this. We do it with an in-house AGI. Others have done similar things that you might be able to buy. Try on asterisk-biz. The question I have is - why the Cisco? Assuming you have SIP or H.323 capable phones, just dump the Cisco and use the asterisk box for the whole shebang. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time counting while playback
On Tue, 16 Mar 2010, Pham Quy wrote:
Hi all,
This question has been asked for days, I think that would be more
comprehensible if i post it in a new thread.
What i want to do is something like karaoke. when users call to
asterisk, a music song is played while caller sings. Their voice
will be recorded and mixed with the music. To do that i used
MixMonitor() and Playback() applications.
I also want to enable users to select a part of song to be recorded
(monitored) for example: Users press '*' to start recording. For
stopping record, there are two ways: (1) he press '#'to stop recording
OR it will be stopped (stop MixMonitor) AUTOMATICALLY after 60 seconds.
How can I count down 60s? MixMonitor app doesnt have any time out
argument.
I detect '#' using Read() app as following
[ivr-test]
exten = test,1,Answer()
exten = test,n,Wait(2)
exten = test,n(prompt),Read(digit,hello-world,1,,3,2)
exten = test,n,NoOp(Input digit - $[${digit}])
exten = test,n,GotoIf($[${digit} = 1]?one,1)
exten = test,n,GotoIf($[${digit} = #]?sharp,1)
exten = test,n,GotoIf($[${digit} = ]?nokey,1)
exten = test,n,Goto(prompt)
exten = test,n,Hangup()
exten = one,1,NoOp(1 pressed)
exten = one,n,Hangup()
exten = sharp,1,NoOp(You press # )
exten = sharp,n,HangUp()
exten = nokey,1,NoOp(No key pressed)
exten = nokey,n,Hangup()
---
But it couldnt read #, key '#' have recognized as NoKey
ps: sorry for my english
Quyps
I think you would be more successful and have more control if you wrote it
as an AGI. Then you could set a timer that would interrupt the process
and you could do what you like from there (hangup?). I think you are
asking too much of the dialplan.
j
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Re: [asterisk-users] 8Port Junghanns BRI card under Dahdi
On Sat, 20 Mar 2010, Loic Didelot wrote: Hi, I try to get an 8 Port Junghanns BRI card working under dahdi. The card works with zaptel but I have no success under dahdi. I load the module with modprobe wcb4xxp. I dont get any errors but I dont see the spans in /proc/dahdi. The output from dmesg remains empty. I use the following dahdi version: URL: http://svn.digium.com/svn/dahdi/linux/trunk Repository Root: http://svn.digium.com/svn/dahdi Repository UUID: a0bf4364-ded3-4de4-8d8a-66a801d63aff Revision: 8353 Node Kind: directory Schedule: normal Last Changed Author: tzafrir Last Changed Rev: 8347 Last Changed Date: 2010-03-18 13:38:53 +0100 (Thu, 18 Mar 2010) Any idea is welcome. Did you run dahdi_genconf? Post your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf . Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8
I finally got it at Calories Consumed. Geesh. Good one! :) j On Thu, 1 Apr 2010, Olle E. Johansson wrote: FOR IMMEDIATE RELEASE Puerto Escondido, Mexico, April 1st, 2010: Digium launches Asterisk VCC (TM) - a new virtual communication platform for enterprises, the public sector and the home. === Asterisk 1.8 will contain a stunning new technology for all Asterisk users world- wide - virtual communication clouds or VCC (TM). With this technology, call handling will never be the same. In one move, the Asterisk development team leaves the old world of PBX call switching behind and moves the enterprise telephony server to the cloud. By combining IPv6, the 3G cell network and cloud services with existing Asterisk technologies like Dundi and IAX2, Digium moves into the era of cloud computing. The launch includes end-user applications powered by cloud services - moving Digium technology to the palm of your hand. - Our new platform is built for the new organization in the workplace, the family and the community - a truly virtual multimedia communication network for the Internet age. By moving our focus away from the traditional PBX, we succeeded in changing the Digium solution from a server centric view to a service centric view. says Sokkie Stevens, product manager for the new platform. The first step was to transform Digium into a virtual service provider. Digium is one of the first companies to get an IPv6 assignment on a global service provider level. After signing peering agreements with major carriers world- wide, the next step was to apply the successful Dundi protocol on top of IPv6. -Dundi and IPv6 was a match made in heaven, says Mick Spenser, the CTO for Digium, Dundi had a successful peering and discovery infrastructure that is now even stronger with IPv6 multicast and secured by using IPsec. VCC will be a binary module distributed with Asterisk 1.8. It will connect to the Digium VCCnet over native IPv6, IPv6 over IPv4 tunnels and directly over layer 2 technologies like Ethernet. All VCC clients will get a native IPv6 address assigned. Enterprises may purchase a full IPv6 network range in the VCCnet to get full access. VCCnet is a network service managed by Digium worldwide. VCCnet will enable automatic follow-me functionality. When you turn on your VCC-enabled smartphone, the VCCnet client will automatically report your location (from 3G cells or GPS) back to the Asterisk service. Your status will be automatically updated as you move between networks, from WiFi in the office to 3G on the road. One person can have multiple VCC clients - one supporting video, another old-fashioned audio and a third HD audio and video. The new IAX3 protocol used in VCCnet will automatically negiotiate media capabilities and select the right client for the right call, depending on privacy settings and personal preferences. For VoxSwitch customers, VCCnet will mean that every user can monitor the movement of coworkers in realtime. By using the new APIs, additional data like credit card transactions, fuel consumption in the car, mileage in the air and calories eaten can be reported with a 3D graphical display using HTML5. As an additional service in the VCCnet cloud, Digium will offer extended capacity for your telecommunications platform. When you need more capacity for video calls, 5+1 hd voice conferences and other coming services, including 3D multimedia conferencing, your existing PBX will be virtually extended by using resources available (and unused) in the cloud. For the system manager, it will look like all these services are produced locally, just like before. VCC includes clients for all popular platforms, including the soon to be released Apple iPAD. Many people was asking us for the Digium Phone, but it felt very wrong to implement an old-fashioned device on top of a modern communication network says Mike Spenser. The client will be a natural part of the personal computing infrastructure that already exists out there. It will be the personal communication exchange, the Facebook of the multimedia realtime communications world. Digium will rename the recently launched Asterisk marketplace to VCCstore and use that infrastructure for distribution of the VCCblocks - applets that enhance your virtual communication cloud. 3rd party developers may apply for development kits and distribution agreements. Digium is currently negotiating the rights to distribute audio books and radio shows for the new culture-on-hold service while not using the VCCclient for two- or multiparty communication. While testing, the most popular VCCblock was the TimeShiftBlock that includes the former voicemail service, now enhanced with virtual timeshifting for realtime calls between timezones. The TimeShiftBlock includes ten popular synthetic voices, including the
[asterisk-users] realtime jitter/latency measurements
Howdy, Can anyone point me to links or discussions about realtime jitter measurement? I read a long thread from 2007 (Douglas Garstang) that didn't end with any conclusions. I want to do the same thing he was trying to do - allow realtime jitter measurements to help control call routing with multiple upstream ITSPs. Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
On Thu, 8 Apr 2010, bruce bruce wrote: I am not sure if unplugging line from card would work as it's still in a hunt and calls will keep coming through that number and won't fall over to next line unless there is a BUSY on the line. There is no timeout; it's a hunt on BUSY. Plus, I don't have site access for two days :-) Nope - unplugging a line that is in a hunt will result in Ring-No-Answer. Ditto for previous advice to destroy the zap channel or to leave it out of the zaptel configuration. You need to busy out that line. You can only do this onsite as far as I know. Or maybe run a script that continually takes that channel offhook and dials something benign... For calls out I give them a funny workaround of using another set to call out and not get audio and then use another phone to call so that a different channel is used. They are happy. Since, I been nagging to them to move to PRI because rain keeps brining their lines down all the time. I can't check zaptel disable of the line now as it nears 9:00 A.M. operation time. I will try that later in the day. I am amazed there is not much control to the lines in situations like this. I totally agree. A busy out application would be a wonderful addition :) I complained about this a few years back... in the meantime, when I need to do such a thing, I busy it out by shorting it at the block. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] jitterbuffer
What is the consensus on using the 1.4 jitterbuffer? Do most people enable it? We have a PSTN server that has our RBS T1 trunks in a central location, then have clients that connect via SIP to us for access to those trunks. Most of them are just fine, but lately we have a handful that are having latency and jitter issues. I am hesitant to just turn on the jitter buffer in zapata.conf on the PSTN server for fear of impacting the clients that are just fine. Should I be? Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer
On Thu, 8 Apr 2010, Tim Nelson wrote: - Jeff LaCoursiere j...@jeff.net wrote: What is the consensus on using the 1.4 jitterbuffer? Do most people enable it? We have a PSTN server that has our RBS T1 trunks in a central location, then have clients that connect via SIP to us for access to those trunks. Most of them are just fine, but lately we have a handful that are having latency and jitter issues. I am hesitant to just turn on the jitter buffer in zapata.conf on the PSTN server for fear of impacting the clients that are just fine. Should I be? I'm using the 1.4 jitterbuffer extensively as many of my customers have poor connectivity (lossy wireless, satellite, etc). It functions well, albeit keep in mind you'll likely need to do some fine tuning to get it just right. I guess that is part of my question - it would seem to me that tuning is basically sizing the buffer, correct? And that the tuning would be different from client to client, as their latency/jitter needs will be different. How did you handle that aspect? Did you just keep playing until you found something that was a best fit for all clients? I kind of understand that the dejitter must happen on the way out as the data gets placed onto a zap channel, and that the other direction should be dejittered at the customer's phone or adapter. In our case this is mainly Polycom IP 501s. I suppose some amount of tuning there will help what our client hears. But the phones are on a 100mb LAN. So would it be worthwhile to force a jitterbuffer on chan_sip on the asterisk server sitting at the client's location? Sorry for trhe vague questions. I think this would be a great topic for someone's BLOG - I haven't found too much in the way of advice via Google this morning. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I record the conversations in a conference call?
On Fri, 16 Apr 2010, Carlos Chavez wrote: On Fri, 2010-04-16 at 08:38 -0700, Luki wrote: Please note: A Zaptel timer must be present for conferencing to work!, but if the user does not have ZAP/DAHDI hardware, he can use ZAP/DAHDI DUMMY Actually, my understanding is that this is incorrect. The conference must contain ZAP/DAHDI callers. A dummy won't do. The reason is that the ZAP/DAHDI driver mixes the audio in the driver and when this is not available it falls back to mixing within MeetMe. But in such case, you can neither record the conference nor run an AGI in the background. See: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe. This is from 2004, maybe it changed by now. Luki I use Meetme all the time in servers that just use dahdi_dummy. I would say that in the past no one would recommend having more than 10 users in a conference is you did not have a hardware clock but that has changed. With newer kernels and Asterisk versions I have been able to get over 50 people in a single Meetme room without any glitches. I also run all SIP conferences with dahdi_dummy and have recorded them. As old as 1.4.22.1 seems to work fine. Using options riM and setting MEETME_RECORDINGFILE beforehand. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Testing a sip call through Asterisk?
On Fri, 16 Apr 2010, Nathan Clemons wrote: I'm looking to find a test tool that will register with our Asterisk (Trixbox) server here at work and place an outgoing call via our main SIP trunk (BroadVoice) to confirm that things are working. I've looked around but I can't seem to find any tools that will do what I'm looking for. I can't just monitor the status of the trunk inside Asterisk, as this is the normal status: [snip] just add qualify=yes to your context and it will monitor the RT latency. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
On Thu, 29 Apr 2010, David Backeberg wrote: I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an alternative to the PAP2T, and I'm seeing prices hovering between $25 and $30. I'm considering getting one of these Grandstream ATAs onsite to play with before I make my final decision. What do people think about both products? Bonus points for if people have bulk deployed these, either with TFTP and configs pushed from a server, or some other good idea. It seems that the PAP2T does support TFTP and an XML-based config for deployments... PAP2T - excellent Handytone - crap Pretty much every large scale TSP has standardized on the PAP2T or 2102. There is a reason the Handytone is priced so low... j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: NAT in SPA922
On Thu, 6 May 2010, Sebastian Milioto wrote: It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Sebastian Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: NAT in SPA922
On Thu, 6 May 2010, Sebastian Milioto wrote: I see the following in SPA922 System tab (new firmware) VLAN Settings Enable VLAN:yesnoEnable CDP:yesno VLAN ID:PC Port VLAN Highest Priority:01234567No Limit Enable PC Port VLAN Tagging:yesnoPC Port VLAN ID: VLAN ID:1 for all Phones, and VLAN 2, 3, 4, 5..,24 for each PC. This should work, right? Sebastian Then you will have to do some work on the gateway and layout all your IP ranges. One for the phones and presumably your asterisk server, then one range for each PC. Your gateway will end up with 25 networks. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digits and Vestec
I'm pretty sure you want it to say naught to make the british happy, for zero anyway... j On Tue, 11 May 2010, David Backeberg wrote: Make it say 'zed'. It will make the British happy, and cause a different kind of confusion for the Americans. On Tue, May 11, 2010 at 4:09 PM, Richard Kenner ken...@gnat.com wrote: This one works on my box (Vestec on 1.4.30 on OpenSuse) Hmm... Not for me. $Digit = (ONE:1 | TWO:2 | THREE:3 | FOUR:4 | FIVE:5 | SIX:6 | SEVEN:7 | EIGHT:8 | NINE:9 | (OH|ZERO):0); This is basically the first thing I tried. At least for my voice, this gets whole lot of spurious 0's. I just tried exactly that (I had the zero case first) and still no-go. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?
On Thu, 20 May 2010, Gordon Henderson wrote: On Thu, 20 May 2010, SIP wrote: Even IF you could get a keyboard with lights you could individually turn on and off (never seen one), http://www.artlebedev.com/everything/optimus/ Bit expensive though... Gordon Heh. A $2400 keyboard. That's crazy. Cool though. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ring splash
Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a call forward option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them try out our dialtone service before porting the number to us and ditching the POTS lines. So we perform some test calls and they all go through fine, and everyone is happy, BUT everytime a call comes through it ALSO causes the POTS line to ring, and a ghost call rings all the phones in the office (the desired result of an inbound call from POTS). When they answer it they get fast busy because it isn't actually a real call. I spoke to the telco this morning about it and they said oh yeah - that is a ring splash that lets the customer know that a call was forwarded. They said this was a feature of their DMS-100, it has worked that way for twenty years, and they can't turn it off. So to the question - can the TDM410P somehow tell the difference between a ring splash and an actual inbound call? I think in the meantime I will send inbound POTS calls to an auto attendant that will eventually hang up, but would love a more elegant solution ;) Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over virtualized VPN
I have several Atom based boxes running OpenVPN and processing up to six simultaneous calls over it with no issues. I am quite sure it could do more. Load is still at .2 :) j On Wed, 26 May 2010, Andrew Hakman wrote: I use openvpn for VOIP traffic all the time. It's not a commercial application, and only one simultaneous call usually on each vpn link, but I even have a VPN client on a Linksys WRT-54g wireless router with 1 phone behind it - it works flawlessly, so it does not take a lot of CPU to run a vpn connection. Andrew 2010/5/26 Motiejus Jakštys desired@gmail.com: Hi List, Our company has several small distributed offices we would like to inter-connect with bridged VPN a single subnet (last example in http://www.shorewall.net/OPENVPN.html). We have SIP phones in every office (up to 5) so we can use SIP without any NATing and securely. Max theoretical simultaneous calls possible ~30, but we have ~5-10 @ regular basis. OpenVPN server would be in the same datacenter like Asterisk PBX (in one physical subnet). Asterisk and OpenVPN are virtualized XEN guests. I wonder about overheads, system loads and other possible gotchas in this setup. Is there anything I should (re-)consider before implementing this? Anyone had difficulties running VoIP or VPN traffic over (virtualized if it makes any difference) VPN? We use mainly g729 and speex, and very little g711. Regards Motiejus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring splash
On Wed, 26 May 2010, Brent Davidson wrote: Just set the POTS lines to answer after a second ring rather than after the first. Problem solved. Now that sounds like a good plan. But a quick look through the options in zapata.conf don't show any kind of option for waiting before pickup. Something that *did* look promising is distinctive ring detection. Has anyone used this ability to detect different ring styles? Presumably with a lot of trial and error I might be able to detect a ring splash from a real ring. ALternatively if someone knows how to actually make the card wait X rings or seconds before answering, that would be great. I'm coming up zero on searches. Its already set to wait for callerid, so I am a bit confused why it is picking up on a splash... seems it should wait for that second ring anyway. Cheers, j On 5/26/2010 11:36 AM, Jeff LaCoursiere wrote: Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a call forward option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them try out our dialtone service before porting the number to us and ditching the POTS lines. So we perform some test calls and they all go through fine, and everyone is happy, BUT everytime a call comes through it ALSO causes the POTS line to ring, and a ghost call rings all the phones in the office (the desired result of an inbound call from POTS). When they answer it they get fast busy because it isn't actually a real call. I spoke to the telco this morning about it and they said oh yeah - that is a ring splash that lets the customer know that a call was forwarded. They said this was a feature of their DMS-100, it has worked that way for twenty years, and they can't turn it off. So to the question - can the TDM410P somehow tell the difference between a ring splash and an actual inbound call? I think in the meantime I will send inbound POTS calls to an auto attendant that will eventually hang up, but would love a more elegant solution ;) Cheers, j -- Brent Davidson Texas Country Title Company 112 W 2nd / P.O. Box 663 Cameron, TX 76520 254-605-0140 ex. 21 br...@texascountrytitle.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to have Asterisk respond from the IP address used for registration
On Thu, 27 May 2010, Mike wrote: Hi, I have a test server with 2 NICs, each with it own IP address. Let`s say 192.168.1.2 and 192.168.1.3. I would like some phones to register by using 192.168.1.2 and some by using 192.168.1.3 as the address. Since the default IP is 192.168.1.2, that is the only working address. Every phone connecting to 192.168.1.3 fails to register, presumably because Asterisk answers back from 192.168.1.2 and the phone doesn't recognize this as the correct SIP server. I am using 1.4.31. Is there any way to have Asterisk answer from the IP address used instead of using the default one? I think you should take a step back and ask yourself why you are trying to do this in the first place. Presumably you have both of these NIC's plugged into the same logical LAN or you will have even more difficulties with routing later. What problem are you actually trying to solve? j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)
On Tue, 1 Jun 2010, Mike wrote: Thanks Joe, They are on different segments. Those two NICs share nothing but the server. But more to the point, it doesn't explain why a simple routing rule matching the destination by IP address works wonderfully, but not one where I match a fwmark that has been set (apparently correctly according to my logging) with iptables. Mike Is this the same thread about having multiple ISP's, and you have external phones hitting the asterisk server on one or the other, and you want the replies to come back on the same segment they came in on? I think IP mangling is making it way too complicated. I suggested you front each segment with a NAT router. Unless you are expecting very heavy traffic volumes, even a cheapo $50 router from Officemax should suffice. Create two internal subnets - one for each interface. Set each router in DMZ mode, so it will send all inbound traffic to the asterisk server on the appropriate interface. The asterisk server will then think that the connection is coming from a locally attached phone, and it will respond out the correct NIC, using the correct IP. The NAT router will send it back out the right Internet connection using the appropriate public IP. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tuning software echo cancellation
We have been distributing asterisk servers for several years now, and early on decided that hardware echo can was the way to go. Our first few boxes without it had horrid echo problems, and attempts at tuning in 2006 didn't make any difference. We installed a new server yesterday at a client's location with a Rhino 4 port FXO card (HW EC included), and when an inbound call was answered the oddest shrieking sound was heard by the caller, and the internal SIP phone heard nothing at all. On a call with Rhino support they disabled the echo cancellation module and all was well, though of course we have a horrible echo problem now. We are going through an RMA process with Rhino, which is fine (kudos for them to cross ship - really good support team there). But the client is of course chomping at the bit to get the system live. We are totally out of touch on the subject of software echo cancellation in asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand that when Dahdi detects no HWEC, it enables SWEC by default. Is there anything I can do to tweak the settings to try and make this liveable for the client until we get the card? The server is in the Caribbean, so it may actually be a bit before the card arrives. We would love to get them running before then, but it is so bad right now that we cannot. Thanks for any links to info... Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tuning software echo cancellation
On Thu, 10 Jun 2010, Gordon Henderson wrote: On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: We are totally out of touch on the subject of software echo cancellation in asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand that when Dahdi detects no HWEC, it enables SWEC by default. Is there anything I can do to tweak the settings to try and make this liveable for the client until we get the card? The server is in the Caribbean, so it may actually be a bit before the card arrives. We would love to get them running before then, but it is so bad right now that we cannot. I've been using OSLEC and TDM400 type cards for a while now (openvox). It just works Gordon Isn't OSLEC on by default? Or is this something I must turn on specifically? If it is on it isn't doing much in our case :) Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tuning software echo cancellation
On Thu, 10 Jun 2010, Gordon Henderson wrote: On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: On Thu, 10 Jun 2010, Gordon Henderson wrote: On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: We are totally out of touch on the subject of software echo cancellation in asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand that when Dahdi detects no HWEC, it enables SWEC by default. Is there anything I can do to tweak the settings to try and make this liveable for the client until we get the card? The server is in the Caribbean, so it may actually be a bit before the card arrives. We would love to get them running before then, but it is so bad right now that we cannot. I've been using OSLEC and TDM400 type cards for a while now (openvox). It just works Isn't OSLEC on by default? Or is this something I must turn on specifically? If it is on it isn't doing much in our case :) I compile up stuff from scratch, so a lot might depend on your distribution.. You need the module dahdi_echocan_oslec loaded, and in /etc/dahdi/system.conf, I have: echocanceller=oslec,1-4 Ahh. I see that the MG2 canceller is installed by default, and I see by Google that it is not very much liked. SVN'ing the latest OSLEC now. Thanks for the advice! Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dual Atom mobo - call capacity
I don't know how it calculates it, but FreePBX shows a bar for total calls that looks like it maxes out at six. We haven't hit that on any installs of this device yet, but that seems pretty low for sure. I know with four calls in progress, all VoIP, transcoding G711u to G.729, the load of the machine is still around .3 . j On Thu, 10 Jun 2010, Michelle Dupuis wrote: I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? MD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small PC to build and run Asterisk
On Wed, 16 Jun 2010, Randy R wrote: On Wed, Jun 16, 2010 at 3:46 AM, Michael Graves mgra...@mstvp.com wrote: Some distro's, like Askozia and Astlinux, have been specifically engineered around running from flash media. This basic form of operation has been well proven in projects like monowall and pfsense. I think you hit the essence of the argument for using these embedded systems, Michael. And we both know from experience and through knowing the people involved that they're both excellent choices! I am now considering using an about-to-be-retired Mac Mini. I'm pretty sure it can be done. How well it might work is another story. I'm pretty much giving up on Skype for Asterisk (and Skype for SIP) now that I realize that they'll be charging a monthly fee that is disproportionately high compared to my need to let Skype users call us. We'll know the pricing in Q4 of 2010, but it looks to be about $15/month for one user. $5 for the channel and $10 for Skype Manager. Maybe something for each name, too? I may have missed this part of the thread, but why giving up on SfA? I was just getting ready to start playing with that myself. Thanks, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small PC to build and run Asterisk
On Wed, 16 Jun 2010, Randy R wrote: On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere j...@sunfone.com wrote: pretty much giving up on Skype for Asterisk (and Skype for SIP) now that I realize that they'll be charging a monthly fee that is disproportionately high compared to my need to let Skype users call us. We'll know the pricing in Q4 of 2010, but it looks to be about $15/month for one user. $5 for the channel and $10 for Skype Manager. Maybe something for each name, too? I may have missed this part of the thread, but why giving up on SfA? I was just getting ready to start playing with that myself. Monthly fees as I mentioned above. In addition to the binary, youneed to pay for Skype Manager and each seat on that (name) - at least that is my understand of their page. Ack! I thought SfA was a one time charge, like their G.729 license. j-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma - how to show channels in use?
Hi, I have several 1.4.29 installations with Sangoma AFT101d cards. Normally we have been collecting the raw data and then graphing channel use for these customers with: asterisk -rx 'show channels' | cut -f1 -d' ' | grep Zap | sort -u | wc -l Then I recently noticed that there were some zombie calls in this list that were not actually active anymore. They go away if I restart asterisk, but in the meantime channel use appears artificially inflated. I am wondering if there is a better method, perhaps with Sangoma CLI tools, to show which channels are ACTUALLY in use? I played around with wanpipemon but that doesn't really give channel specific info. Any clues? I posted on the Sangoma forums also... Thanks! j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma - how to show channels in use?
On Tue, 22 Jun 2010, Danny Nicholas wrote: Since you are already grepping, just add a grep -e zombie (you should probably go ahead and do core show channels instead of show channels since this will bite you at some time in the future). True. Its an old script ;) But I used the zombie term adjectively - there is no zombie text in the output. I just know that a call is not still ringing hours after it was initially placed. Not sure how it is getting into that state... here is an example excerpt: Zap/5-1 18666902...@from-pst Ringing AppDial((Outgoing Line)) SIP/7157787-08331ec8 18666902...@resident RingDial(Zap/g0/18666902511) Zap/3-1 18666902...@from-pst Ringing AppDial((Outgoing Line)) SIP/7157787-08335df0 18666902...@resident RingDial(Zap/g0/18666902511) Zap/2-1 18666902...@from-pst Ringing AppDial((Outgoing Line)) SIP/7157787-b6d28360 18666902...@resident RingDial(Zap/g0/18666902511) It kind of looks like this one SIP endpoint tried to make the same call three times in a row without success, and all of the calls show as still active, though I know they are not (in fact they show as still ringing). So are channels 2, 3, and 5 actually still busy from the telco's perspective because asterisk is keeping them open? That would suck. A lot. I did get a reply from Sangoma, who basically said that their driver doesn't know about the individual channels - that is totally handled by asterisk. So it seems there is no way other than what I am already doing to judge the channels in use? Thanks, j -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, June 22, 2010 11:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sangoma - how to show channels in use? Hi, I have several 1.4.29 installations with Sangoma AFT101d cards. Normally we have been collecting the raw data and then graphing channel use for these customers with: asterisk -rx 'show channels' | cut -f1 -d' ' | grep Zap | sort -u | wc -l Then I recently noticed that there were some zombie calls in this list that were not actually active anymore. They go away if I restart asterisk, but in the meantime channel use appears artificially inflated. I am wondering if there is a better method, perhaps with Sangoma CLI tools, to show which channels are ACTUALLY in use? I played around with wanpipemon but that doesn't really give channel specific info. Any clues? I posted on the Sangoma forums also... Thanks! j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] one for your filters
Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place four thousand calls to what appears to be a toll number in Zimbabwe last night. Filter 82.150.165.5. A more overriding problem for me is how do we know what *destinations* to filter so this idea of war dialing a toll number is something we can cutoff before it gets to our upstream provider? Is there some collected list of toll prefixes that I can filter on? Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one for your filters
On Wed, 23 Jun 2010, Gordon Henderson wrote: On Wed, 23 Jun 2010, Jeff LaCoursiere wrote: Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place four thousand calls to what appears to be a toll number in Zimbabwe last night. Filter 82.150.165.5. A more overriding problem for me is how do we know what *destinations* to filter so this idea of war dialing a toll number is something we can cutoff before it gets to our upstream provider? Is there some collected list of toll prefixes that I can filter on? How did they guess the SIP username and password? That's what I'm more concerend about... Gordon I'm still trying to figure that out. Our SIP usernames are seven digit phone numbers, so not really difficult to guess, but the passwords are 7 char alpha-numeric strings, auto generated. We don't at present restrict people to their addresses, as some are dynamic. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one for your filters
On Wed, 23 Jun 2010, Tarek Sawah wrote: you can start by simply telling us what is the purpose of your server.. and does it have long distance of overseas?? do you use Numeric usernames? simple passwords? passwords the same as your username? this way you can offer more info so we can help you.a quick answer will be.. opening a few and blocking ALL is easier.. as you can have upto 400 prefix to block .. unless you call world wide.. then you will have to block the countries you don't call .. another option.. make your usernames more complex.. letters and numbers.. an additional option is to use fail2ban with Asterisk support.. it will block the IP after the number of attempts you set in the configs. a client of mine wanted simple usernames and passwords to be setup using the keypad on the ipphones.. two months ago they had the same problem you faced.. 400$ to Zimbabway .. and later on 1200$ to Zimbabway.. their provider have a limit of 30 minutes per call .. so the caller had to redial.. unless it's automated.still you can provide us with more info.Regards -- Tarek Sawah Well we run local dial tone service in the US Virgin Islands. So our customers are connecting with ATA's, various models of Polycom phones, and SIP trunks from a custom PBX we sell to hotels and businesses. They connect from dynamic addresses most of the time, so we cannot apply any IP based filters to their accounts, though we may be able to restrict them to certain IP blocks. I'd rather not, since the upkeep would be quite a hassle, and would remove their ability to take their ATAs traveling. Our SIP usernames are their seven digit phone numbers, which may have been a bad choice, but most of the brute force attacks we have witnessed are trying combinations of 3 digit extension numbers. I haven't seen anyone try a brute force attack with 7 digits. The passwords are seven char auto-generated alpha-numeric gibberish, and it seems rather unlikely to me that this account was broken by brute force trial and error. I'm still investigating other methods... like perhaps they broke into my server first and found the provisioning files. That would be bad. All of that aside - I know there are various things I can do to tighten up our SIP security. My question was more geared towards what do people do to keep their customers or employees from dialing toll numbers worldwide? I cannot restrict my customers to calling a set of countries. But I would feel justified in blocking toll numbers that I don't have a way of billing back. I just don't know where to start to build such a filter list. Surely other ITSPs have had to deal with this issue - fraud situations or not. The US is easy - all toll numbers start with 1-900 (I think :). Other countries are not so straightforward I understand. Has anyone else tackled this problem? Thanks, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one for your filters
On Wed, 23 Jun 2010, Steve Edwards wrote: On Wed, 23 Jun 2010, Jeff LaCoursiere wrote: Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place four thousand calls to what appears to be a toll number in Zimbabwe last night. Filter 82.150.165.5. Ouch. 82.0.0.0/8 is on my block list, available at: http://www.sedwards.com/class-a-block-list If you don't need to receive packets from far away places, it's a great start. Nice! I am now one of your grateful subscribers ;) Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call file question
I am sure this is simple, but have been struggling. I want to create a
call file that dials out a particular Dahdi channel to enable call
forwarding on a POTS line. I have this in extensions.conf:
[custom-callfwd]
exten = s,1,Answer
exten = s,n,Dial(DAHDI/4-1/*717157750)
exten = s,n,Verbose(${DIALSTATUS})
exten = s,n,Hangup
[custom-callfwdcanc]
exten = s,1,Answer
exten = s,n,Dial(DAHDI/4-1/*72)
exten = s,n,Verbose(${DIALSTATUS})
exten = s,n,Hangup
Using FreePBX I have setup custom destinations and custom
applications such that users can dial a code from their desks and enable
or disable forwarding via the above contexts. That works fine.
Now I whipped up a C program to create a call file to do the same thing
from the command line:
[snip]
fprintf(callfile, Channel: Local/*...@custom-callfwd/n\n);
fprintf(callfile, Application: Playback\n);
fprintf(callfile, Data: hello-world\n);
[snip]
When I run this it creates the call file and I see this in the console:
-- Attempting call on Local/*...@custom-callfwd/n for application
Playback(hello-world) (Retry 1)
And that is all... no call actually goes out on the Dahdi line.
I'm sure I am not properly creating the call file to do what I want. Any
suggestions?
Thanks,
j
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Re: [asterisk-users] call file question
On Wed, 30 Jun 2010, Steve Edwards wrote: Now I whipped up a C program to create a call file to do the same thing from the command line: [snip] fprintf(callfile, Channel: Local/*...@custom-callfwd/n\n); I don't see exten *71 in custom-callfwd. Doh! That was the problem. In FreePBX I made *71 the feature code to access that context, and it was still in my head when I made the callfile. Why are you using a local channel in your call file? That was the meat of the question, actually. I want to create a single leg with a callfile - just the outbound call. All other times I have used callfiles I was creating two legs and bridging them. Is there a better way to do what I am attempting? fprintf(callfile, Application: Playback\n); fprintf(callfile, Data: hello-world\n); [snip] When I run this it creates the call file and I see this in the console: -- Attempting call on Local/*...@custom-callfwd/n for application Playback(hello-world) (Retry 1) What does the call file look like before you mv it to the spool directory? Exactly the above fprintf lines... Thanks, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
On Tue, 6 Jul 2010, C.Savinovich wrote: I am writing to you privately... [snip] Doh! Need another cup of coffee? j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video IVR Asterisk ?
On Sat, 2010-07-17 at 00:08 +0530, Anita Hall wrote: Hi Is it possible to receive video calls using Asterisk and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of your valuable inputs. regards, Anita Hall, Simmortel. We use Grandstream video phones and have noticed that if we record our prompts with these phones, the video is saved with the audio. So we set our main IVR up this way, and without doing anything special (other than enabling video in sip.cfg), we have video IVR for those customers that call with video capable endpoints. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you use for Invoicing?
On Mon, 2010-08-02 at 14:26 -0400, bruce bruce wrote: Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. Its partially open source (you get the source to everything but the financial routines), and it runs on Unix rather than Windows, though you do have a web interface. Checkout BillMax: www.billmax.com They have some extensions that create PDF invoices in telecom style. Its pretty powerful otherwise for doing any kind of recurring billing. I wrote the initial version, but I am not associated with the company anymore. j Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you use for Invoicing?
On Tue, 2010-08-03 at 09:17 -0400, bruce bruce wrote: I agree but the mentioned software is not opensource. My conditions clearly included opensource. No, your prefer listed opensource. If you had said requirement I wouldn't have suggested it. j On Tue, Aug 3, 2010 at 12:35 AM, Nick Brown n...@ipera.com.au wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Tuesday, 3 August 2010 1:58 PM To: j...@sunfone.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What do you use for Invoicing? Maybe good but the first look brought me to a Pay version. Doesn't satisfy the opensource condition. thanks, Open Source software does not necessarily mean free software. Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opensource Speech recognition for Asterisk
On Sun, 22 Aug 2010, David Backeberg wrote: On Sat, Aug 21, 2010 at 10:49 PM, Duncan Turnbull dun...@e-simple.co.nz wrote: Voice recognition is a pain for people with accents and poor lines and when Everybody has an accent. Some people live in a place where the people they talk to sound like themselves, so they forget that fact. Of course, this is a huge problem if you, for example, want to have an English language voice recognition system that works across the continental United States. Even for people who speak 'correct' or 'common' English for their region, these systems aren't that great in my experience. The bigger of a vocabulary you have, the worse trouble you'll have, because these systems, again, in my experience, only know synonyms or alternate regional words for the same thing if they were programmed by somebody who thought of the synonyms / alternate words / alternate legitimate pronunciations. Anybody with an imagination can think of plenty examples, for example, from the United States: * soda / pop / soft drink / beverage / drink / Coke / other trademarked names Comes down to the designer - most of the systems I am used to using (like American Airlines system, which is quite good IMO) are focused on the basics - digits 0-9, yes/no, agent, etc. I don't think it is overly difficult to make this work even with varying accents, though UK folks used to saying double naught might have issues :) j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opensource Speech recognition for Asterisk
On Sun, 22 Aug 2010, Jason Aarons (US) wrote: I'm not aware of an open source speech product. Some great examples where speech recognition works well are 1-800-USA-RAIL, Microsoft/Cisco corporate directory 425-882-8080 you can say the employees name and be connected and those works great, 1-800-Goog-411 also works well. Windows 7 Speech Recognition, Dragon Natually Speaking work pretty good. Vonage does a good enough job of sending my home voicemails to my email in Speech to Text, I use this app daily, rarely having to listen to actual voicemails. What Speech-Text doesn't convey is anger/happiness, etc. Great story from a friend in a large unnamed corporation - an upper level mgr named Jack Smith got a call from a very angry customer. He did his best to help him and in the end asked how he got transferred directly. The man said the system asked me who I wanted to speak to and I said 'JACK ASS' and I got you! j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, HylaFax and Cardiff
On Mon, 23 Aug 2010, Don Kelly wrote: I’m looking for a way to use our implementation of HylaFax on Asterisk with Cardiff (an old installation of Cardiff document stuff). Is someone doing that? If no one has direct experience, is there a HylaFax client that emulates WinFax print-to-fax? Lots! http://www.hylafax.org/content/Desktop_Client_Software j-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
On Fri, 2010-09-10 at 23:07 -0400, bruce bruce wrote: Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. I think this may be because the peer is not be recognized as a peer. If you know the IP of the source of the call (the provider) try sip set debug ip X.X.X.X. Then you will probably see the rejection. Not that that will help you much :) You need to find out why it is being rejected. Either you changed the peer parameters or they did... j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
On Sat, 2010-09-11 at 03:53 -0400, Zeeshan Zakaria wrote: This is not elastix or FreePBX forum and asking non-asterisk related questions here is misusing this mailing list. Allow anonymous sip is not an asterisk feature. Look in the code in extensions.conf what it is programmed to do and you'll figure out why it is happening. Or maybe post the code and ask why such a behaviour, which'll be better way to ask this elastix related question here. If you know what this part of dialplan does, rest is easy to figure out. Zeeshan A Zakaria Heh - listen to you - top posting, bad english, and self appointed list police. His problem certainly seemed asterisk related to me, and has NOTHING to do with code in extensions.conf. He even posted CLI commands he is attempting to use to find his problem. I applaud him for taking the initiative to try working it out on his own, and see no problem at all with his question. I hope we can help him fix it. j -- www.ilovetovoip.com On 2010-09-10 11:17 PM, bruce bruce bruceb...@gmail.com wrote: Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see the ACK, REQUEST and ACCEPT of sip debug just fine. This is Elastix with Asterisk 1.4.33.1 Any thoughts? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
-- www.ilovetovoip.com On 2010-09-11 7:22 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere j...@sunfone.com wrote: Sending to 123.123.12... Either you changed the peer parameters or they did... If he is not receiving any response, it is most likely a routing issue. -- [un top posting] On Sat, 2010-09-11 at 19:30 -0400, Zeeshan Zakaria wrote: Actually it is a very easy to understand and fix issue, but looking at the code taking care of anonymous sip calls is the key. Those who post third party GUI related issues should at least post the underlying asterisk config or code here, so the asterisk part of the problem can be fixed. Zeeshan A Zakaria Its not that he isn't receiving a response - its that his peer debug statement isn't getting tripped because the peer hasn't authenticated. That's why I suggested he debug by IP rather than peer. Then what he will see is the SIP auth attempts and asterisk rejecting them, but in my experience not much is of value in seeing those packets - it doesn't point to *why* the connection is being rejected. The routing must be ok since allowing guest sip connections (the result of setting accept anonymous in FreePBX) allows the calls to come in fine. His problem is the peer authenticating. This of course has nothing to do with extensions.conf, as the dialplan is not involved. It is a SIP authentication problem, purely. There is no relevant code to post, and if you had ever looked into FreePBX's relevant code you would realize that it is actually fairly complex, and you would indeed have a difficult time debugging the flow. It *might* help if he posted his peer entry, but without seeing the other side that may not help much either. As Paul suggested first off, he should be in touch with his provider, whose tech support should be able to help him sort it out. I ran into a strange one EXACTLY like this just last week. We have a residential dial-tone customer with a Linksys SPA2102 (our standard device for this service). He had someone come out and replace his home router, and when he did he stopped authenticating. He has a fixed IP, so I enabled the debugging as I have mentioned twice now (by IP) and saw the attempts and rejections. After much hair pulling I *disabled* nat in his peer entry and it suddenly connected fine. This is bizarre, as our standard peer configuration works for 100% of the rest of our customers, who all connect from behind their home nat gateways of all kinds. I still don't know why that fixed it. Sorry you took it so harshly Zeeshan, but the only posts that stick out to me from you are the ones where you are bashing people for posting questions. I don't recall any off the top of my head where you are actually helping. Yup, I consider that policing, and it isn't needed. Like someone else suggested, if you don't want to read it, delete it. And no, I am not going to bother to read back through archives to see if that is the truth. Its my impression of your posts, thats all. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
[snip] customers, who all connect from behind their home nat gateways of all kinds. I still don't know why that fixed it. Sorry you took it so harshly Zeeshan, but the only posts that stick out to me from you are the ones where you are bashing people for posting questions. I don't recall any off the top of my head where you are actually helping. Yup, I consider that policing, and it isn't needed. Like someone else suggested, if you don't want to read it, delete it. And no, I am not going to bother to read back through archives to see if that is the truth. Its my impression of your posts, thats all. j [un top posting again] On Sat, 11 Sep 2010, Zeeshan Zakaria wrote: Poster is having problem when he disallows anonymous sip peers. Do you know at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet seen the dialplan for FreePBX. Umm, no, poster is having problems that are only SOLVED by allowing guest sip connections (since you want to stick with asterisk terms, not FreePBX, right?). That is because when he doesn't allow guest connections his inbound calls are getting rejected, as they are not matching any of his defined peers. I'm not guessing here - these are facts based on his observations. Your bizarre assumptions that he (or I) need to better understand FreePBX's dialplan code are guesses. j-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 800 Origination/Termination - International
On Tue, 14 Sep 2010, Joe Freeman wrote: Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can. Thanks- Joe I use IPComms to do US/Canada/US Virgin Islands. 2.5c/min + a flat rate per channel (I think $15?). j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 800 Origination/Termination - International
On Wed, 15 Sep 2010, Kyle Kienapfel wrote: On Wed, Sep 15, 2010 at 6:04 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Tue, 14 Sep 2010, Joe Freeman wrote: Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can. Thanks- Joe I use IPComms to do US/Canada/US Virgin Islands. 2.5c/min + a flat rate per channel (I think $15?). j Is that the same rate for calls from US and canada? I ask as these two, good incoming rate for calls from the states, but 7 cents a minute for calls from canada: http://flowroute.com/services/inbound/ http://vitelity.net/index.php?p=retailserv voip.ms has two options for tollfree's $0.99 a month and is mentioned on their website $1.49 a month and 3.2 cents a minute incoming from USA or canada, its only listed in the account manager Yes, same rate all around, which is why I settled on them. Very competitive - especially for the Caribbean. Thanks, j-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] random hangups on RBS T1
Hi, I have an asterisk 1.4.35 server with a Digium TE410P (1st gen) four port T1 card. Only one RBS T1 plugged into it right now. I have been getting complaints about random hangups. Endpoints are all remote, but I track very closely the latency (by graphing the output of sip show peers) which normally shows me when a peer is having connectivity issues. Several that I have investigated this morning have no such issues (latency less than 20ms and steady). I have several servers with Sangoma A104d cards, and the Sangoma driver has a debug mode that lets me see the RBS bit transitions. I have used this in the past to prove that the T1 provider is actually triggering the hangup from their side. Does any such debug mode exist for the Digium cards? I would like to dig into this, because if I can prove the carrier is at fault I will have hard data to bring to a PUC meeting next month :) Any suggestions? Thanks, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] random hangups on RBS T1
On Tue, 21 Sep 2010, Shaun Ruffell wrote: On 09/21/2010 10:48 AM, Jeff LaCoursiere wrote: I have several servers with Sangoma A104d cards, and the Sangoma driver has a debug mode that lets me see the RBS bit transitions. I have used this in the past to prove that the T1 provider is actually triggering the hangup from their side. Does any such debug mode exist for the Digium cards? I would like to dig into this, because if I can prove the carrier is at fault I will have hard data to bring to a PUC meeting next month :) Any suggestions? Jeff, you can monitor the state of the RBS bits via 'dahdi_tool'. But in case you need a running log (and I haven't tested this) I added a patch to https://issues.asterisk.org/view.php?id=18025 if you want to try that out. Cheers, Shaun Fantastic! I will give the patch a shot. Thanks, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for a better ATA
On Fri, 8 Oct 2010, Bryant Zimmerman wrote: I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF talkoff and T38 Any advise before I start testing with these brands would be apperciated. Any better option you may know of. Thanks for any input Bryant I'm curious which of the above ills you attribute to the Linksys (assuming an SPA2102? The PAP2T does have the T38 problem I believe). Its basically the defacto standard for all the giant ITSPs. Perhaps your problem is one that could be rectified in some way. I have also tried Grandstream and Audiocodes (still use the MP-124s in certain situations) and have found that the SPA2102s work the best for us... Cheers, j-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fraud advice
Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was taken advantage of over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern destinations have been placed, which ended up being sent over our customer's PSTN trunk, and of course there was no warning until the bill came today. Unfortunately the bill only covered the first few days of this fiasco, and was only $700. I am afraid the one that is on the way will be tens of thousands. ONE CALL on the bill that just arrived was $200 (80 minutes to Sierra Leone). I'm sure this started out as a single scan. It must have been posted, because I have at least ten IP addresses now that were placing calls via the same peer. They are from all over the world. So what is the accepted procedure? I'm in the US Virgin Islands, so do I go to the FBI? Police? Is their some telecom fraud body to report such things to? Does any one ever get any relief from such events? I'm basically sick to my stomach right now. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
On Thu, 21 Oct 2010, Steve Howes wrote: Hi, Given the recent increase in SIP brute force attacks, I've had a little idea. The standard scripts that block after X attempts work well to prevent you actually being compromised, but once you've been 'found' then the attempts seem to keep coming for quite some time. Older versions of sipvicious don't appear to stop once you start sending un-reachables (or straight drops). Now this isn't a problem for Asterisk, but it does add up in (noticeable) bandwidth costs - and for people running on lower bandwidth connections. The tool to crash sipvicious can help this, but very few attackers seem to obey it.. The only way I can see to alleviate this, is to blacklist hows *before* they attack. This means you wont ever be targeted past an initial scan. Is there any interest in a 'shared' blacklist (similar to spam blacklists, but obviously implemented in a way that is more usable with Asterisk/iptables)?. Clearly it raises issues about false positives etc, but requiring reports from more than X hosts should alleviate this. There's all the usual de-listing / false-listing worries as with any blacklist, but the SMTP world has solutions we could learn from. Leaving a 'honeypot' running on a single IP address has revealed a few hundred addresses in less than a month. I am fairly certain these are all 'bad' as this host isn't used for anything else. There is obviously a wealth of data (and attacks) out there that would be good to share. Anyone have any thoughts? S -- I'll subscribe, that is for sure. What is the best way to dist the blacklist? iptables include file? Or something more integrated to asterisk... just thinking off the top of my head that a module that vetted inbound connections against an external list would be a very cool thing. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Billing Question???
[snipped very confusing top and bottom posting mix] On Thu, 21 Oct 2010, Sherwood McGowan wrote: Dhaval, You're right, I forgot one thing. The frozen table's id column should not be an autoincrement, it should be set by the insert statement, using the original method I decsribed for creating a unique integer from the callerid number and the current EPOCH. That way, you can be sure that multiple concurrent calls that have frozen funds will only retrieve the record they created. (Oh and, once you thaw the frozen funds, delete the appropriate record in the frozen table) I'm not sure why you think this will only work for a single call at a time. Each time a call occurs that is related to an account will cause more money to be frozen from that account, thereby causing future calls to have less available balance and therefore less time for a call limit. This works for ANY number of concurrent calls on an account, and every one of those calls freezes funds based on the rate at which THAT call's amount to freeze was calculated against. EACH call determines IT'S rate, which is then used to determine the amount to freeze from the account ON THAT CALL. Additionally, since the rate is specific to each call, the limiting of the length of THAT call, your issue of limiting is also a non-issue. I also have worked on the logic for this scenario, and I gave up. Our calling card system now locks a balance and forces the account to one simultaneous call at a time. We report the maximum length of a call to the customer just before the ringing starts, and as someone else stated - to cut it off prematurely is very confusing to the customer (and one of the number one complaints against calling cards - if you sell in Florida it could actually get you in serious trouble). The problem with each call freezing a portion of the balance is that no one call has access to the whole balance, and that was determined (in our case) to be unacceptable, and is definitely unacceptable to the calling card customer. But I don't think we are talking about calling cards. I am guessing that Dhaval is trying to create a termination company, and has customers that maintain a balance with him that want to be able to place multiple simultaneous calls. A common problem. We often end up with negative balances with our upstreams for this very reason - we may be near the bottom of our balance and several calls in progress terminate and bring us below zero. I am sure this is what he is trying to avoid, as the industry is full of people that will simply walk away from a negative balance. Dhaval - your wish, I think, is to manage exactly in real time to decrease the balance as the calls progress. In that way all calls in progress would be cutoff simultaneously as the balance hit zero. That kind of scenario would be very complicated with asterisk. Some external program would have to keep track of the balance and the calls currently in progress, and cut them off at the appropriate time. I would be very interested if anyone has attempted this. I envision something that EVERY SECOND deducts from a balance for every call in progress, at the current rate for each call. Not impossible for sure... Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
On Thu, 21 Oct 2010, Andrew Latham wrote: Always start here... http://www.spamhaus.org/drop/ If the AS is stolen, you can block the network and never have to worry about it... ~ Andrew lathama Latham lath...@gmail.com I guess you are assuming that spam networks should be included in the blacklist by default? I'm not sure that is a good assumption. Some of my customer netblocks have ended up on spam lists unknowingly (by leaving open SMTP servers for example), and if that had affected their ability to place phone calls also it would have been disastrous. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
On Thu, 21 Oct 2010, Steve Howes wrote: On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote: I'll subscribe, that is for sure. What is the best way to dist the blacklist? iptables include file? Or something more integrated to asterisk... just thinking off the top of my head that a module that vetted inbound connections against an external list would be a very cool thing. I was thinking some sort of script to pull via HTTP to update whatever you wanted (output as iptables etc). I know its not an instant 'lookup', but an hour delay between updates is nothing. Also means whoever is running the server isn't getting hammered by everyone ;) Realtime lookups from Asterisk would be quite a load (and would introduce latency). S -- I agree in principle - some cron job pulling the list by http would certainly be simple. But just to continue my thoughts to the brick wall, I don't see a lookup adding latency to the call other than what should be a very brief addition to the time taken for a call to be accepted. Once accepted you would just continue to accept the packets. How about something DNS based? Load could potentially be distributed that way if a number of people agreed to participate. I'll mull this over a bit more. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
On Sat, 30 Oct 2010, Joel Maslak wrote: For me, monitoring outbound call volume makes a lot more sense. I would love to see an easy to use, out of the box method to alert me if more than x number of erlangs* are exceeded within a five minute, sixty minute, and one day time period. For me, I would want alerting on more than 10 erlangs over five minutes, 8 over an hour, and 2 over a day. Exceeding these would likely indicate fraud for my installation. Smaller sites would use smaller numbers, larger ones would use bigger ones. This only tells you after it is way too late that you now have upstream bills to wrangle with your carriers about, or (like in my case) that your balance is now depeleted, if it trips anything at all. In my very recent case only FIVE calls, all placed at the same time, caused charges of over US$8K as they stayed connected for over two days. This would not have tripped any erlang threshold, and you don't even know that it is affecting your balance until the calls cease. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
