Re: [asterisk-users] openvz
On Fri, Sep 03, 2010 at 03:11:39PM +0200, mattias wrote: Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi I don't expect any problem. Absolutely right: 1.6.x works fine with OpenVZ and Virtuozzo out of the box as long as you don't need any hardware interfaces. You don't need any kernel sources, though these are available. You basically just install exactly as you would on any other system. There is no problem with timing (e.g. MOH) - at least none that I've ever come across and at least not for 1.6.x which is what I've used under Virtuozzo for some time now. If you want to install Digium g.729 licenses you need to do some small configuration changes but these are easy to do and full instructions are on the OpenVZ Wiki somewhere. Faris. -- _ -- 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: UK PPP certification -- what is it?
They mean PhonePayPlus (formerly ICSTIS). www.phonepayplus.org.uk I am not aware of them certifying particular phone systems. Rather, they impose certain requirements and obligations on the service provider depending on the nature of the service being provided and the number range it is provided on. But maybe more stringent regulations and phone system certification does apply to certain types of service which I've never had to deal with - adult stuff, for example - so I'd give them a call if you can't find the info on their website. Faris. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: 13 August 2010 1:14 AM To: Asterisk Users Mailing List Subject: [asterisk-users] OT: UK PPP certification -- what is it? A client asked me to come with a system that will pass certification with PPP in the UK. Google is not being helpful :( It has something to do with recording calls in case the PPP requests a copy. Supposedly their rules were relaxed on August 1st if that helps. Any clues (links?) will be appreciated. -- Thanks in advance, -- _ -- 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] 1.6.2.10 sounds Makefile error?
I don't have such a centos 4.8 system handy to test with. What version of 'make' do you have? make --version rpm -q make In any case, please submit a report to http://issues.asterisk.org/ Thanks Tzafrir. GNU Make 3.80 Make-3.80-7.EL4 I'll submit a bug report. I just can't figure out why it would only be me seeing this. I'm mystified. Faris. -- _ -- 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] 1.6.2.10 sounds Makefile error?
I'm having some mysterious problems installing 1.6.2.10 on Centos 4.8 (totally up to date). I can't see anything on Google or the list regarding this issue, which I find a bit odd considering 1.6.2.10 was released a few days ago. I'm therefore assuming there's something weird about my setup, even though there shouldn't be! I had no problems with 1.6.2.7 or any other release. It is just 1.6.2.10 that's causing the problem. I've tried using an svn checkout and downloading asterisk-1.6.2.10.tar.gz and asterisk-1.6.2-current.tar.gz but the same thing happens. Basically, after the usual ./configure and make, when I make install I get the following (this is from the SVN attempt but other than the paths all is the same): #make install CFLAGS= -I/usr/include/libxml2 -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -march=i686 build_tools/mkpkgconfig /usr/lib/pkgconfig; mkdir -p /var/lib/asterisk/static-http for x in static-http/*; do \ /usr/bin/install -c -m 644 $x /var/lib/asterisk/static-http ; \ done if [ -d doc/tex/asterisk ] ; then \ mkdir -p /var/lib/asterisk/static-http/docs ; \ for n in doc/tex/asterisk/* ; do \ /usr/bin/install -c -m 644 $n /var/lib/asterisk/static-http/docs ; \ done \ fi mkdir -p /var/lib/asterisk/images for x in images/*.jpg; do \ /usr/bin/install -c -m 644 $x /var/lib/asterisk/images ; \ done mkdir -p /var/lib/asterisk/agi-bin make -C sounds install make[1]: Entering directory `/root/asterisk-svn/asterisk-1.6.2/sounds' Makefile:144: *** missing separator. Stop. make[1]: Leaving directory `/root/asterisk-svn/asterisk-1.6.2/sounds' make: *** [datafiles] Error 2 Looking at the sounds directory, I have Makefile and sounds.xml Running make install in that directory gives me the same Makefile:144 *** missing separator. Stop When I copy across the sounds/Makefile from my 1.6.2.7 source directory to the 1.6.2.10 source directory, all is well again and I can make install with no errors. I did a diff on the two Makefiles and there are what appear to be several differences, but I can't put my finger on any obvious errors. Any ideas? Faris Raouf -- _ -- 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] Virtual Asterisk Installation
We have been successfully using Asterisk (1.6.0.x) in a heavily loaded Virtuozzo (= commercial OpenVZ) environment for over a year. I'm sure we aren't the only ones to do so. We had some terrible problems with random one-way audio a few minutes into some calls to start with, which I was worried were to do with the virtualisation/timing. But after much hair pulling and investigation it turned out to be down to some serious firmware bugs in the routers we were using, combined (if I recall correctly) with an IAX bug-ette. At any rate we've had no problems at all since these things were corrected. This is all without a timing source too, but then we don't use conferencing. We never handle more than 4 simultaneous calls though, and everything is IAX/SIP based so there's no hardware interfacing issues for us to worry about either. There's actually a commercial Asterisk-based product, 4PSA VoipNow (www.4psa.com), that specifically supports Virtuozzo and VMWare and also Amazon EC2 (!!). Indeed, they even provide VMWare images, Viruozzo Templates and an Amazon EC2 AMI for ease of installation in these environments. There's a free version too with a 10 extension limit. I should point out that although I've tried Voip Now, it was only to the extent of installing it to look at the GUI - I didn't try making any calls or registering any phones etc. I'm very familiar with the company through their Plesk add-on products though, so I have no doubt it works. I don't know which version of Asterisk it is based on. I am also unsure about hardware interfacing with this product - which I think is really going to be one of the main problems you will face for your project. Faris. Now my big question: What kind of virtualization should I run on the Server? I have already used VMware ESXi and Proxmox. It would be very nice if there was a way to make snapshots (for backup purposes). I read about clock problems (physical time != virtual time) and so on. If I'm right this does not matter when using OpenVZ but when using KVM, XEN, ESX, ... Please tell me your opinion. I definitely want to run the Asterisk via virtualization - so we have to find a solution for this ;-) Thank you very much! felix -- _ -- 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] iaxmodem, chan_capi, hylafax problem and faxing in general
Can anybody point me in the right direction please? I'm having some issues getting iaxmodem and hylafax to talk to each other. I have no doubt that someone has had this type of issue before but I can't find anything useful in the archives or on Google. Under RH9, with chan_capi 7.1, Asterisk 1.2.24, with an AVI Fritz ISDN2e BRI card all working perfectly... I've downloaded and installed iaxmodem (2 days ago -- latest version - I'm not able to check exact release from where I am right now). It runs OK, and registers with Asterisk. I've downloaded and installed hylafax (again 2 days ago -- latest version). It installs OK and runs. I copied the suggested config file from iaxmodem directory to /var/spool/hylafax/config/etc (unchanged) I run iaxmodem before starting hylafax, and a device /dev/ttyIAX appears. Running iaxmodem in non-daemon mode shows no errors. I've not modified the default iaxmodem config other than in terms of username and password etc to register with asterisk. The problem is that when I run faxstat, it does not show hylafax connected to any tty. And when I try to run faxaddmodem (just to see what might happen) and select ttyIAX, I get an error saying that hylafax can't detect the speed of the device and that I should set it manually, and then iaxmodem promptly crashes at that point. I've poked and prodded generally but I'm getting nowhere. Does anybody have any suggestions as to where to start looking for the cause of the problem? For that matter, is this the right way to go? The idea is to have a central fax server running on the Asterisk box, receiving and sending via the ISDN BRI line. Ideally I'd like fax to email facilities and some point, and the ability for clients across our network to send faxes. I've seen that there's also another option, Astrifax, but that seems to require a separate installation of spandsp, java, and other bits which seems like a bit of overkill. Pointers or suggestions would therefore be appreciated. Faris. ___ --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, chan_capi, hylafax problem and faxing in general
Thanks Lee. No, I'm definitely not running faxgetty - I didn't realise I was supposed to :-( And no, I'm not using the + version. Back to square one. At least I'm going in the right direction again! Only I've sidetracked and am currently trying to use capi4hylafax instead of iaxmodem which seems to work wonderfully but I'm having some issues with root verses uucp permissions which is spoiling my fun. Anyway, thanks again! Faris. (please excuse my top posting) -Original Message- Faris Raouf wrote: The problem is that when I run faxstat, it does not show hylafax connected to any tty. You're probably not running faxgetty (and your later comments below confirm this...) And when I try to run faxaddmodem (just to see what might happen) and select ttyIAX, I get an error saying that hylafax can't detect the speed of the device and that I should set it manually, and then iaxmodem promptly crashes at that point. You're probably not using HylaFAX+. The hylafax.org releases kill iaxmodem when faxaddmodem is run. See: http://hylafax.sourceforge.net/ Lee. ___ --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] Linksys SPA3102 - PSTN hangup detection in the UK
Henry.L.Coleman wrote: Yep, just swop the two wires. Sometimes the Tip and Ring get reversed and most loop start interfaces don't really care (they work either way). It's worth a try since if the disconnect is a reverse polarity flash then the card may see not see this condition as it is already reversed. I have a similar problem with Foriegn Exchange line (FX) but I haven't had time to visit the client to check this out yet. Thanks Henry. I'll definitely give this a go. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with CallerID (UK) TDM400P ( CID timed out waiting for ring )
[EMAIL PROTECTED] wrote: We have a problem where callerid works 50% of the time on both lines. What we are seeing in the logs is: Hi Phil, Unfortunately your configuration looks OK to me. Here's mine, which works 100% with CID (but not dratted hangup detection!). There are some duplications and things - just ignore them. I note that you have sendcalleridafter=2 and I have =1 but I think =2 is just fine. The only thing I can suggest is to play with the RX gain in case things are just too quiet: usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=no callwaitingcallerid=no threewaycalling=no transfer=no canpark=yes cancallforward=yes callreturn=no echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes group=1 immediate=no signalling=fxo_ks language=en context=sip2 channel = 1 signalling=fxo_ks language=en context=blah2 channel = 2 usecallerid=yes cidsignalling=v23 ;cidstart=ring cidstart=polarity sendcalleridafter=1 busydetect=yes busycount=3 ;callsrogress=yes progzone=uk rxgain=2.5 txgain=2.0 ringtimeout=5000 signalling=fxs_ks polarityonanswerdelay=1000 answeronpolarityswitch=yes hanguponpolarityswitch=yes resetpolarityonring=true language=en context=blah channel = 3 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Do any *UK* users have an SPA3102 (the newer version of the SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call has hung up? I've read everything I can find, including an SPA3000 UK setup PDF that lists UK ring etc tone settings, port impedances, disconnect tone settings and so on, but I'm still not getting PSTN hangup detection to work. Any help would be appreciated. Thanks, Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Conrad Wood wrote: On Tue, 2006-10-24 at 12:05 +0100, Faris Raouf wrote: Do any *UK* users have an SPA3102 (the newer version of the SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call has hung up? I've read everything I can find, including an SPA3000 UK setup PDF that lists UK ring etc tone settings, port impedances, disconnect tone settings and so on, but I'm still not getting PSTN hangup detection to work. I got an spa-3000 that works perfectly well now. (UK) I had some trouble at first though. What firmware are you using and what's the symptom? does it not hang up or does it hang up during calls? Conrad Thanks Conrad, It is brand new so I assume the firmware is the latest?: Software Version: 3.2.6(GWa) Hardware Version: 1.1.5. It just doesn't detect real hangups at all. If the person calling hangs up, either before and after the call is answered, the SPA will eventually timeout after about 30 seconds and hang up - in other words it does not detect the disconnect tone like it should. I have other small niggles but I'm sure I can sure them with some config tweeking but right now the hangup problem is really my top priority. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Conrad Wood wrote: It is brand new so I assume the firmware is the latest?: Software Version: 3.2.6(GWa) Hardware Version: 1.1.5. It just doesn't detect real hangups at all. If the person calling hangs up, either before and after the call is answered, the SPA will eventually timeout after about 30 seconds and hang up - in other words it does not detect the disconnect tone like it should. Interesting, I had it the other way round. It detected the disconnect tone during conversations. I disabled disconnect tone detection. *I think* it detects a polarity reversal instead. (It's been a while and once it worked I forgot about it) I posted my settings here http://www.conradwood.net/sipura.pdf Are they any different from yours? Conrad There are some differences, yes. Before I begin I should say that I think I've sorted out my main problem. I basically experimented with putting the values that were in the disconnect detect field into the ring tone field so I could hear them. Although the 400Hz tone was correct with my initial setting [EMAIL PROTECTED];20(*/0/1) and subsequent settings (including the same as you have), they just didn't sound right. There would be two distinct volumes on the 400Hz tone. In the end I'm using [EMAIL PROTECTED];2(0/*/1). This may seem backwards but seems to work. The higher the value of the number of repeats (2 in my case), the longer it takes to detect the disconnect tone. Also in this version of the firmware it does not seem to be necessary to put two tone values. But yes we have differences in our config other than that. You have polarity reversal detection and I do not (I did try with it on, but it didn't help even though there I have measured a polarity reversal on disconnect) You also have long silence detection off, which would seem logical to me. I have it on, but will probably switch it off in case it drops the line if I put the phone down to look for something etc. You have min CPC at 0.085 and mine is at 0.09. We also differ slightly at the bottom of the page. You have 3ms On hook speed, I have less than 5ms. You have Line In Use Voltage 30 and I have 25. You have Ring Validation 100Ms and I have 256. You have Ring Indication Delay of 256 and I have 0. I will now try your settings to see if it helps with my next big problem --- I'm not getting a CLI number. Instead I get the Username I've allocated to my SPA. Once again thanks hugely for your help on this. It is really good to be able to compare configs. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
ah. Do you have callerid from BT (bt line?). I signed up for something called BT Privacy or so which is free and gives you callerid. If you turn on logging (debug) on the sipura it'll log the received callerid via syslog. Also helpful to check under info Last seen number or so. There is CLI on this particular line. I even managed to get it to work with the TD400P (or whatever the analog card with 4 modules is that Digium sells for Asterisk) in the past -- but no luck with hangup detection on that at all so I gave up on it. I'm not seeing any caller id in the syslog nor the last seen number thing. (which helpfully just says , :-) I thought your suggestion about the filter was excellent so I tried a few different ones (we are an ISP so I have a few hanging around ;-) ) but to no avail. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Henry.L.Coleman wrote: Just a thought ... try reversing the Tip and Ring Henry L.Coleman CEO Henry, Apologies for answering the wrong message in my last post. I thought I was answering the one from Conrad. Sorry! By reversing the Tip and Ring you mean physically in the wiring or somewhere in the SPA? I can see Forward/Reverse settings for Line1 in the config, but nothing on the PSTN side? Thanks, Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Conrad Wood wrote: I'm not seeing any caller id in the syslog nor the last seen number thing. (which helpfully just says , :-) I'd be pretty sure that the device doesn't detect the cli. My one does list the number under the 'last seen number thing'. What sort of line is it? Straight BT? telewest? Some converter? I thought your suggestion about the filter was excellent so I tried a few different ones (we are an ISP so I have a few hanging around ;-) ) but to no avail. Thanks. It's unlikely that it would affect cli. It was meant as a possible explanation for the disappearing polarity reversal. Conrad I think I've found where the polarity reversal is going ... I think my lightning/spike filter is eating it or something. When I look at the syslog with the filter removed I see messages about polarity reversals. With the filter they are missing. Yet the phones I normally have plugged in still seem to read the CLI with no difficulty with or without the filter, and the SPA can't read them with or without the filter. Very fustrating. Yes, it is a bog standard BT line. I've tried using the CLI detection mode without the PR but that doesn't work either. I'm not sure what to try next Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is 1.2.12.1 production ready
Just in case it helps anyone: We had 1.2.12.1 crashing on us on a daily basis, and sometimes several times a day. I found that by disabling all qualify lines in iax.conf and sip.conf the problem went away. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 601 Expansion Module: Light the LEDs???
Doug wrote: Hey Folks, Been wrestling with the 601 and the expansion module. Finally figured out how to populate both with speed dial entries. Also hints are showing in Asterisk with the show hints command. But how do I get the LEDs to light when one of these other extensions is either off-hook, or ringing. Reading the 'Net and Polycom's documentation doesn't give a clear solution. Is there a genius out there who has this working?? Please help!!! I can't help with the extension module, but on the phone itself...I can't quite remember exactly what you do but the trick is NOT to have lines programmed for all the line key buttons on the phone itself. Any free line key buttons will then get populated by the speed dials, and the respective LEDs will show the status of those speed dials (assuming the corresponding hints are correctly configured in asterisk) You also have to enable to buddy feature on the phone itself using the XML config file. I think that part at least is documented somewhere. Search the mailing list for polycom buddy or polycom hints or similar and you will find more detailed instructions and a cry for similar help from me six months to a year or so ago. My problem was that I had defined all the line keys as lines, and freeing up those solved the issue. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream SX2000 attended tranfer
magnus wrote: Hi all, could anyone share how to perform attended transfers with Asterisk and Grandstream SX2000's - we are able to perform blind transfers with no problem, but attended transfers fail - is it necessary to set two line identities on the phones to be able to do this? Appreciate all input, thanks - Magnus Funny you should ask -- I was going to ask the exact same question about the GXP-2000 (is that the model you mean or is there a new similar phone?). At any rate they both seem to have the same problem: In order to do an attended transfer on the Grandstreams we have to have two accounts defined on the phone (both on separate usernames/numbers in our case - maybe you can do it with one?), one on Line 1 and one on Line 2. Call comes in on Line 1. Put caller on hold. Dial person you want to transfer to on Line 2. Then transfer. I've tried pressing Line 2 until the identity of Line 1 comes up - i.e. reuse Line 1 - but this does not work. Instantly fails. The instruction manual gives completely different instructions but these simply do not work. And what is not clear is how the transfer works when using the strange two account situation - is the transfer going * - phone - person you are transferred to once transferred? (can reinvite = no incidentally) or is the phone This is all completely unlike the case with a Polycom where it just lets you transfer with no problems and just one line. I'm using the latest stable firmware on the Grandstreams - it has been like this for all firmware versions I've used for over a year now. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct settings for UK (BT) FXO
Brian Candler wrote: Is there a document somewhere giving the correct TDM400P FXO settings for use on a BT PSTN line in the UK? All I can find is http://www.voip-info.org/wiki/view/UK+Asterisk+Details A patch was written for a previous version of Asterisk that got halfway there. I found some bugs in it which took us all the way and it worked perfectly. The original patch and my modifications and some other modifications/enhancements were added in a later release version of Asterisk but unfortunately for no apparent reason although it worked in other EU countries it no longer does so in the UK. I'm afraid I forget the exact details. Unfortunately I've not had time to investigate the code to try and figure out what is going on and how to fix it. Basically it is all down to the polarity switches that happen or don't happen on hangup. What is really required is for the code to sense the constant tone you get when you hangup in the UK, but this is far too complex for me to be able to deal with. So personally I've given up on analog and I'm sticking to the digital realm of ISDN. One day I may invest in a Sipura FXO box (or are they now Linksys or something?) which does sense the disconnect tone. Incidentally I think there are people on this list who have no issues with the TDM400p in the UK, but I have no idea how/why. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom_acd_functions SIP trouble
James Fromm wrote: Yeah, we tried that. Tried every combination of variables in sip.conf. Only solution that works is removing the requirement for a secret. Faris Raouf wrote: One thing to try is setting type=peer instead of type=friend. I'm a bit dazed and confused at the moment, but if I remember correctly Polycom phones just don't work with type=friend. Of course this doesn't explain why SJPhone won't work either so maybe I'm totally off-track, but it might be worth giving it a try just the same. Don't give up just yet. I spent hours with exactly the same problem (in the mainstream * release) until I sorted it out with the type=friend. How about re-trying but changing the password in both the polycom and sip.conf? Try a 1 digit password. Also is there no way to get some debug output in Asterisk that can give more details? Something that can show the password being sent and the password expected rather than just saying it is wrong (seems like a very useful thing to have if it isn't there already)? Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom_acd_functions SIP trouble
Dovid Bender wrote: I am sure you prob. know this but in your configs it shows secret commented out. Also it with a softphone if it dosent work then, then its your configs. Also did you remember to reload asterisk ? - Original Message - From: James Fromm [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, July 24, 2006 2:24 PM Subject: [asterisk-users] Polycom_acd_functions SIP trouble I'm trying to use the latest revision of Bweschke's branch from SVN for polycom_acd_functions. Asterisk builds and runs without error but all SIP devices can't register when specifying a secret in sip.conf. The Polycom 601 I'm testing with and a copy of SJphone will not register. IAX from Idefisk works without error. One thing to try is setting type=peer instead of type=friend. I'm a bit dazed and confused at the moment, but if I remember correctly Polycom phones just don't work with type=friend. Of course this doesn't explain why SJPhone won't work either so maybe I'm totally off-track, but it might be worth giving it a try just the same. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Presence
Forrest Beck wrote: Does anyone have a working implementation of SIP Presence? I have a new Grandstream GX-2000 phone with the supported hardware and I am not sure how to setup presence with asterisk. I've just been through this myself. It is relatively simple once you manage to figure it out but really hard until you do! 1) Upgrade to the latest beta firmware for the gpx-2000. (details here: http://www.voip-info.org/wiki/view/GXP-2000) 2) Assign a speed dial button as type Asterisk BLF in the drop down in the Basic page of the grandstream web config system and have it watch a particular extension. Lets call it extension 100 for the purposes of this example. It does not matter what name you give this speed dial entry - it is just a label. In the above step you are effectively telling the grandstream to watch a special hint extension, number 100. 3) Now here's the confusing bit. In extensions.conf you need to use a hint priority which you need to define in the same section as you have your normal extensions defined (you can do it elsewhere but for the purposes of this explanation we'll keep things simple). We are using 100 as our example in step 2. BUT extension 100 does not have to exist in your current dial plan! This is a key thing to get in your head. And if you do have 100 in your current dial-plan it doesn't matter either because adding a hint for that extension will not harm the existing extension. You do not need to have your hint priority anywhere near the lines where you define extension 100 in your dial-plan either -- you can have a block of hints anywhere (as long as they are in the same section as your normal extensions) The syntax is: exten = xxx,hint,sip/y where xxx would be 100 in our example. But what is y? Basically it is the name of the phone/device you want to monitor, as defined between the [ and ] in your sip.conf for that device. You can monitor more than one phone/device at a time by using a syntax like this: exten = xxx,hint,sip/ysip/ (add as many as you like on that line with a inbetween) But lets get specific: In extension.conf, in the same [heading] as your other sip extensions are defined, add: exten = 100,hint,sip/phone1 (where phone1 is a phone as defined in sip.conf and is what you want to watch) IMPORTANT: It is not necessary for phone1 to be configured as extension 100 in extensions.conf. hint extensions and real extensions are separate entities. This is crucial to understand. There is no link between them. As mentioned previously it is not even necessary for extension 100 to be defined previously in extensions.conf at all. Now, at this point, any device set to watch hint extension 100 will be alerted to the status of phone1. (In step 2 we set the grandstream to watch 100, so it will respond to changes in status on that hint extension). It is THAT simple. Only it isn't, because there are some gotchas. First of all, in sip.conf you need to have type=peer in your phone's definition, NOT type=friend. It just doesn't work if you have type=friend for Grandstream phones (polycom phones, on the other hand, won't work if you have type=friend -- they have to be type=peer but at least hints work with them when set to type=peer) The other thing you need for granstreams at least is call-limit=1 in your phone's definition in sip.conf. You may like to experiment with this though, as I'm not 100% sure it really is required. In any case it prevents more than one call ever going into the phone at the same time, which may not be what you want. So, having done all this, restart asterisk, then reboot your phones (an asterisk restart confuses hints/presence on grandstream phones sometimes) At the asterisk command line, enter the command: show hints You should see that extension 100 is shown, and that it was status Idle and 1 watcher. If you use the phone (you have to dial something not just lift the handset) and then use show hints again you should see status = InUse and the red light next to the speed dial button on the watching phone will light up like magic. It is wondrous when it works. Additional info: When setting up a speed dial in the grandstream gxp-2000 as asterisk BLF you can also define which account you want this to work on. If each of the four possible accounts (sip registrations) is connected to different asterisk servers (as opposed to all being configured to register on the same one), depending on which account you select in the Basic page when defining the speed dial, the grandstream will watch the extension defined on that account. This allows you to monitor presence on up to four different asterisk servers. In contrast (and VERY annoyingly), Polycom phones always use the account defined in in the first account (Line 1) - there does not seem to be any way at all to get them to watch extensions on multiple asterisk servers. Faris.
Re: [Asterisk-Users] SIP Presence
Viggiani Domenico wrote: Wonderful explanation! Just a note: So, having done all this, restart asterisk, then reboot your phones (an asterisk restart confuses hints/presence on grandstream phones sometimes) It seems that Asterisk = 1.2.7 solved this issue. Thank you! I'll try 1.2.7 shortly. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 600 presence indication on *LED*?
Jerry Jones wrote: Create a contact entry with their extension and enable buddy watch on it It will then show up on an unused line key On May 27, 2006, at 3:26 PM, Faris Raouf wrote: I've somehow managed to battle may way through hinting issues with type=peer type=friend and various other oddities and now have presence working correctly on my Polycom 600 and Grandstream GXP-2000 phones. However, on the Polycom I have to press the Buddies softkey in order to see if an extension with a hints priority is in use or not. I've spent all day going through google and my local archive of the mailing list, and from what I can see it appears that I should be able to set up one of the 6 line keys on the left of the phone to somehow show presence indications. But I simply cannot figure out how. I only know how to configure a line key as, well, a line key (i.e. mapped to a particular SIP registration in sip.conf). What do I need to do in order to get a nice LED or something to flash or light up or whatever on the phone to show that a particular extension on another phone is in use? This is driving me totally insane. Any help would be appreciated! Thanks, Faris. It will then show up on an unused line key That was the key! I had all my line keys in use. I got rid of one that I didn't really need, and BOOM! It works. Thanks! Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 600 presence indication on *LED*?
I've somehow managed to battle may way through hinting issues with type=peer type=friend and various other oddities and now have presence working correctly on my Polycom 600 and Grandstream GXP-2000 phones. However, on the Polycom I have to press the Buddies softkey in order to see if an extension with a hints priority is in use or not. I've spent all day going through google and my local archive of the mailing list, and from what I can see it appears that I should be able to set up one of the 6 line keys on the left of the phone to somehow show presence indications. But I simply cannot figure out how. I only know how to configure a line key as, well, a line key (i.e. mapped to a particular SIP registration in sip.conf). What do I need to do in order to get a nice LED or something to flash or light up or whatever on the phone to show that a particular extension on another phone is in use? This is driving me totally insane. Any help would be appreciated! Thanks, Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK experts only. BT Outgoing caller ID not showing
Paul Redstone wrote: Hi guys Thanks for help on this so far. There was no typo - old exchange was System X and new one System Y. Also caller ID is enabled on the new DDI range so we get incoming caller ID. BT are looking at this - the guys I talked to is being very helpful and has referred this to a colleague (why do we find this so surprising in the UK - BT helpful!). Paul Redstone ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This may be a red herring, but worth a punt... If you have a look at the BT website they have two CLI type options on ISDN. There is COLP (Connected Line Identity Presentation) and CLIP (Calling Line Identification Presentation). Way back in the mists of time I remember investigating COLP but I can't remember what the heck it turned out to be, nor if I decided if I actually needed it or not :-) Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hints/subscriptions accross IAX
(I hope this isn't html - Thunderbird is so annoying) I'm new to using hints/subscriptions on * so please be patient with me. I have two * systems in different geographic locations, connected via IAX Location1 has a Polycom 600 and a GXP-2000 phone Location 2 has a single GXP-2000. With the latest GS firmware, at Location1 I've managed to get an LED to light up on the GS phone when a line on the Polycom is in use. This is great. But I need to get an LED to light up on a GS in Location2 when a line on the Polycom at Location1 is in use. Is this possible? If so, can anybody give me any pointers as to how? Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems if GXP-2000 phones and Asterisk are not on the same network
Mimmus wrote: Hi, I have a lot of GXP-2000 phones not registering with Asterisk server. After two days of attempts, it seems that problem is due to the fact that phones and server are not on the sme network. Do you know if this is known issue? We have GPX-2000s connecting via different networks with no problems. However, at one point I had a real struggle to get them to register on certain lines but not others. The solution was to do a complete reset, wiping all settings and starting again. They now work OK most of the time. I'm using the current beta firmware (not alpha). Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues Not Reporting Estimated Hold Time
I'm getting the same thing since upgrading from 1.0.x to 1.2.x - no queue hold time announcements. There are other oddities in queues in 1.2.x compared to 1.0.x too. But I'm always afraid to raise them as bugs in case they are not, and 1.0.x was going things the wrong way and 1.2.x is going things the right way :-) Faris. Michael J. Liberatore wrote: Nope, never removed them, they are still there. It doesn't report an error either, it just never says playback . If this works for someone please let me know, otherwise I will report it to the bug tracker. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Thursday, March 16, 2006 7:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queues Not Reporting Estimated Hold Time When you upgraded to 1.2.5 did you remove your old asterisk-sounds but forget to reinstall it? (Not positive, but) could be that the prompts you need are in asterisk-sounds Michael J. Liberatore wrote: I am running 1.2.5 with a simple queue and have announce-holdtime = yes in queues.conf for that queue. The person is being told their posistion in the queue and the CLI says the estimated hold time, but it never plays it for the caller. It worked previously, i am not sure when it stopped, i think after 1.2.1. Is this a known bug? I dont want to report it to the bug tracker if its already been discussed, but a search yeilded no results. Thanks Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI and UK Premium Rate Numbers
This is pretty standard Asterisk behaviour exten = whatever,1,NoOp exten = whatever,2,Dial(SIP/nSIP/n+1SIP/n+2) exten = whatever,3,Hangup The incoming ISDN call will ring the specified SIP phones, and will not be answered until one of them picks up. As simple as that? Thanks!! That's perfect. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN BRI and UK Premium Rate Numbers
Can anyone help point me in the right direction please? I'm based in the UK and I want to start using a Premium Rate number with Asterisk - I think the equivalent in the US would be a 900 number. Effectively the caller pays much more to call such a number than a normal national or local call. The problem with these is that I don't want Asterisk to actually signal to the telephone network that the call has been answered until someone really does answer it, otherwise the caller will be paying a premium rate just to listen to an Asterisk-generated ring tone until someone answers the call. My setup would be chan_capi-cm and an ISDN BRI line with several MSNs (not DDIs -- this line does not support point-to-point only point to multipoint but we do have another line that does do point to point and has DDIs, and if necessary we can use it), and of course Asterisk and various SIP phones. I have very little idea where to start, as everything I normally do with Asterisk involves the call being answered immediately then put in a queue, which is no good in this case. What I really want is for the call to come in then: 1) One or more SIP phones will ring (unless they are on a call) but for Asterisk not to signal an answer just yet 2) Only when someone is free and answers the call does asterisk answer and put them through. Ideally I'd also like the caller and the person answering the call to hear a recorded message saying that calls to this number cost X per minute ... blah blah, this message being triggered only when someone answers the call. This will warn the caller *and* the person answering that this is a premium-rate call. The person answering the call will know to speak after this message has been played. But that's just an ideal situation. Right now I'm more concerned about how to stop Asterisk answering until someone is available to take the call. Can anyone help please? I don't really know where to start. The Wiki seems to be pointing me towards using DID/DDIs, but that's about as far as I've got. NOTE: We don't need the actual Premium Rate numbers themselves. We have those already (we used them with an old telephone system until recently). My problem is just to get Asterisk to work with them in the way I've outlined. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 alternative?
Thank you Lee, Dave, Rich, Joel and of course also Kevin. Between your various messages I finally understand what's happening and how it works, and have actually converted everything to alaw, ulaw, slin and gsm and am not actually using the mp3 side of things at all anymore. The difference is very noticeable in terms of MOH quality except when using g729 on the link between Asterisk and the phone - the sound quality seems worse there. I have two related questions though which I'm hoping someone can help with: We use alaw, ulaw, gsm and g729 between phones and asterisk. Sox can convert files to ulaw, alaw and gsm (not to mention slin) but what about g729? Is there such a thing as a format that won't need transcoding when using g729 links, or is this not something that is possible? And what is the signed linear (slin) format used for? Thanks, Faris. Lee Archer wrote: Check out the musiconhold.conf.sample in the asterisksource/configs folder. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faris Raouf Sent: 23 February 2006 18:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mpg123 alternative? Ah! Now this is actually something I've not been able to get my head around: Note: As of Asterisk 1.2.0, Mpg123 is no longer used by Asterisk, which has its own MP3 player. Can anybody tell me where this built-in MP3 player is in 1.2.x/how do I use it ? I still seem to have the usual two mpg123 processes running with 1.2.4, with whatever music on hold is set in musiconhold.conf I'm sure it is very obvious, but I can't for the life of me figure out what I'm supposed to do to use the built-in MP3 player facilities. I just have the following in my musiconhold.conf: [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 random=yes Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem
I've having a big problem after having upgraded to 1.2.4 and
chan_capi-cm 0.6.4
When making outgoing calls I don't seem to have any control over the CLI
that is presented to the called party -- it can be any one of the MSNs
allocated to the line, allocated on what seems to be a random basis.
This is on a BT Business Highway line (which is essentially an ISDN2e
line with two built-in analog ports), configured with 8MSNs alongside
the single the master digital telephone number for the line.
With a much older version of chan_capi-cm (0.5.4 I think) and Asterisk
1.0.9 it was always the master number that was presented, and that is
actually what I want.
Obviously the format of capi.conf has changed between these two versions
of chan_capi-cm, so maybe I'm doing something wrong. Any suggestions
would be appreciated.
Here's my capi.conf (actual numbers changed to protect the innocent!)
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;ulaw=yes;set this, if you live in u-law world instead of a-law
[123456]
; Master number for line - i.e. number for line before MSNs were allocated
; and which still works and still accepts incoming calls.
isdnmode=msn
msn=01234123456
;incomingmsn=*
incomingmsn=123456
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
[123457]
;first MSN
msn=01234123457
;incomingmsn=*
incomingmsn=123457
isdnmode=msn
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
{repeated for next 7 MSNs}
And in extensions.conf I have:
[globals]
ISDN1=CAPI/123456
[mysip]
;GET OUTSIDE LINE (ISDN1 - dial 9)
ignorepat = 9
exten = exten = _9.,1,Dial(${ISDN1}/${EXTEN:1}/b)
exten = _9.,2,Playback(busy)
exten = _9.,3,Hangup
*
I've tried using ISDN1=CAPI/contr1 but it makes no difference.
I've tried leaving out the isdnmode=msn but it makes no difference
I've tried entering 01234123456 as the msn= line on all of the msn
entries in capi.conf but it makes no difference either.
And now I'm out of ideas and any help would be appreciated.
Thanks,
Faris.
p.s. sorry if this message is HTML. I've switched to using Thunderbird
and it is confusing the heck out of me. It claims this is a plain text
message but it doesn't look like plain text to me from this end!
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Re: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem
Thanks for that Peter!
I think your message solved my problem: I set the master number to be in
group 1 (group=1) in capi.conf and called Dial with CAPI/g1 and it
worked perfectly.
However, with group=1 in capi.conf for the master number, at the moment
no matter what I do I'm getting the master number presented as the CLI.
This is fine by me because it is exactly what I want, but it is all very
confusing :-)
Faris.
Peter Braidwood wrote:
I have ISDN2 and 10 MSN numbers and also have just upgraded to 1.2.4 and
chan_capi-cm and have this working completely perfectly
Capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=en
[ISDN1]
isdnmode=msn
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=from-isdn
group=1
devices=2
bit of extensions.conf, I dial 9 for an outside line
[pstn]
exten = _9./321,1,Dial(CAPI/g1/01234567890:${EXTEN:1})
exten = _9./322,1,Dial(CAPI/g1/01234567891:${EXTEN:1})
exten = _9./323,1,Dial(CAPI/g1/01234567892:${EXTEN:1})
exten = _9./324,1,Dial(CAPI/g1/01234567893:${EXTEN:1})
exten = _9./326,1,Dial(CAPI/g1/01234567894:${EXTEN:1})
exten = _9./327,1,Dial(CAPI/g1/01234567895:${EXTEN:1})
exten = _9./328,1,Dial(CAPI/g1/01234567896:${EXTEN:1})
exten = _9./350,1,Dial(CAPI/g1/01234567897:${EXTEN:1})
exten = _9./351,1,Dial(CAPI/g1/01234567898:${EXTEN:1})
exten = _9./352,1,Dial(CAPI/g1/01234567899:${EXTEN:1})
So when extension 326 dials out the cli that is presented would be
01234567894
Contact me off list if you want any further help.
Peter Braidwood
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: 23 February 2006 13:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem
I've having a big problem after having upgraded to 1.2.4 and
chan_capi-cm 0.6.4
When making outgoing calls I don't seem to have any control over the CLI
that is presented to the called party -- it can be any one of the MSNs
allocated to the line, allocated on what seems to be a random basis.
This is on a BT Business Highway line (which is essentially an ISDN2e
line with two built-in analog ports), configured with 8MSNs alongside
the single the master digital telephone number for the line.
With a much older version of chan_capi-cm (0.5.4 I think) and Asterisk
1.0.9 it was always the master number that was presented, and that is
actually what I want.
Obviously the format of capi.conf has changed between these two versions
of chan_capi-cm, so maybe I'm doing something wrong. Any suggestions
would be appreciated.
Here's my capi.conf (actual numbers changed to protect the innocent!)
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;ulaw=yes;set this, if you live in u-law world instead of a-law
[123456]
; Master number for line - i.e. number for line before MSNs were
allocated
; and which still works and still accepts incoming calls.
isdnmode=msn
msn=01234123456
;incomingmsn=*
incomingmsn=123456
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
[123457]
;first MSN
msn=01234123457
;incomingmsn=*
incomingmsn=123457
isdnmode=msn
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
{repeated for next 7 MSNs}
And in extensions.conf I have:
[globals]
ISDN1=CAPI/123456
[mysip]
;GET OUTSIDE LINE (ISDN1 - dial 9)
ignorepat = 9
exten = exten = _9.,1,Dial(${ISDN1}/${EXTEN:1}/b)
exten = _9.,2,Playback(busy)
exten = _9.,3,Hangup
*
I've tried using ISDN1=CAPI/contr1 but it makes no difference.
I've tried leaving out the isdnmode=msn but it makes no difference
I've tried entering 01234123456 as the msn= line on all of the msn
entries in capi.conf but it makes no difference either.
And now I'm out of ideas and any help would be appreciated.
Thanks,
Faris.
p.s. sorry if this message is HTML. I've switched to using Thunderbird
and it is confusing the heck out of me. It claims this is a plain text
message but it doesn't look like plain text to me from this end!
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[Asterisk-Users] Problems mixing audio in queues and playing queue positions
Hi folks, Over the weekend I finally decided to upgrade one of our Asterisk systems from 1.0.9 to 1.2.4 I had no significant problems and all is well in general - as usual Asterisk rules! However, I did run into two small issues. Can anyone help me solve them please? The first one involves queue position announcements, and the second one is regarding monitor-join. A) In 1.0.9, as soon as a caller enters a queue they are played the position announcement (which is what I want) and then it is replayed every X seconds depending on what I have for announce-frequency in queues.conf This is not the case in 1.2.4 though. Effectively the queue position is not played until after the sum of times set for timeout and retry. e.g. from queues.conf: [myqueue] timeout = 10 retry = 5 wrapuptime=5 maxlen = 0 musiconhold = default strategy = ringall announce-frequency = 60 announce-holdtime = yes announce-round-seconds = 0 monitor-format = wav49 monitor-join = yes member = sip/phone1 member = sip/phone2 member = sip/phone3 With this queues.conf configuration, in 1.2.4 the caller won't get their queue position played until after they have been in the queue for 15 seconds, while in 1.0.9 they got it immediately. Any suggestions? I really think it makes more sense for it to be played immediately when the caller joins the queue rather than waiting for the first timeout, which for many configurations might be much longer than the 15 seconds in mine if timeout and retry are set to higher values. B) My second issue is that monitor-join = yes in queue.conf does not seem to work for me - I still get individual -in and -out files for calls in the queue. Admittedly I had this problem in 1.0.9 too, but not in 1.0.7 I don't think. A very significant bit of information here is that using the m option in Monitor() in extensions.conf does not work for me either (I still get individual -in and -out files). The correct soxmix command gets executed (at least it appears on the console) but does not actually have any effect on the files. Manually running the exact same command on the command line does work, and joins the files correctly, so sox and soxmix are there, and are in the path, and work correctly in theory. Any suggestions would be appreciated! Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Draytek Vigor 2900 Asterisk
Jonathan Attwood wrote: I'm in conversation with Draytek's pre-sales dept.. Here's the most recent reply: Hello, We really don't know of anyone who has run an Asterisk server on a Vigor2900. There are doubtless people around, but it's relatively rare. Most people don't run SIP servers. Regards, All I want to know is, if I buy one of these routers, will it break my setup or not - ie. assuming I set up the relevant port-forwarding, can I expect any one-way audio issues. Can't get a definitive answer from suppliers or the manufacturer, so I hope someone here uses this model with Asterisk.? ___ I have a 2900G behind a Cisco 1720 with dual ADSL WICs in one office, and a 2600VGi standalone in another (I don't use the 2600's built-in FXS ports -- they aren't very good - seem noisy). I have Asterisk servers in both offices, linked via IAX. I have incoming voip services going independently to both Asterisk servers. I've had no problems whatsoever -- everything has worked perfectly. The QoS facility in both routers allows you to reserve a certain amount of bandwidth (in or out) for IAX and SIP and this seems to work fine though it isn't necessary on our networks. I'm using port forwarding on both routers to route IAX and SIP to the private IPs of the Asterisk boxes. But you will need to open the appropriate ports on the firewall in the router, or firewall the Asterisk boxes and DMZ the Asterisk boxes. However, the new Dreytek 3300 series of routers is even more interesting. Multiple WAN ports for backup/load balancing, and optional hardware FXO/FXS ports. I hope this helps. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] suddenly iax calls don't work anymore
Gerald Dachs wrote: Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial(Zap/2-1, IAX2/username:password@sip.coco-connect.de/number) in new stack -- Called username:password@sip.coco-connect.de/number -- Call accepted by 62.180.50.221 (format g729) -- Format for call is g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1758 ast_set_write_format: Unable to find a path from gsm to g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? ... Gerald Someone will probably correct me, but it looks like you are trying to use the g729 codec for your calls (or coco-connect.de is forcing you to use g729), but this requires a license from Digium and is not installed on your machine. Try using a different codec if possible or, if you do have a g729 license try re-installing the codec and re-activating it. I think this may solve the problem. But as I say, someone may correct me - I may be completely wrong about this. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording queue calls
Dov Bigio wrote:
Hi,
When I set monitor-format=wav49 on file queues.conf for a queue,
Asterisk records calls at /var/spool/asterisk/monitor. But the file
names it users are the call-ids of the calls.
Is there a way to change that, and use information such as date, time,
agent and queue to build the filename?
It would make the localization of such files much more easy.
Other useful that I miss is the capability to to allow the files to be
stored in different directories, such as
/var/spool/asterisk/monitor/queue1, /var/spool/asterisk/monitor/queue2,
and so on, based on the queuename. Is this possible by any means?
Hi,
Yes. All you need to do is use the following in your extension.conf at
the point before you call the queue
SetVar(MONITOR_FILENAME=foo)
or, if you are using 1.2.x
Set(MONITOR_FILENAME=foo)
For example, I have:
Set(MONITOR_FILENAME=${TIMESTAMP}-${CALLERIDNUM}-to-SALES-${UNIQUEID})
and then a little later on:
Queue(salesqueue|t|||60)
in my extensions.conf
Which sets the monitor filename to start with a timestamp, then the CID
of the caller, then the to-SALES is what I use to differentiate
between queues (I'd have a different Set command for a different queue).
I then add the UNIQUEID as a just in case to make absolutely sure
there's no way I'd ever have two files of the same name.
I hope this helps,
Faris.
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[Asterisk-Users] in and out recorded audio mixing in queues
Way back I was still on Asterisk 1.0.7, I configured my systems to mix
the incoming and outgoing audio call recordings into one file per call
for both normal calls and queued calls using:
exten =
_9.,1,Monitor(wav49,${TIMESTAMP}-${CALLERIDNUM}-to-${EXTEN:1}-${UNIQUEID},m)
; m option merges audio into one file and deletes the parts
in extensions.conf
and
monitor-join = yes
in queues.conf
As far as I remember this worked perfectly.
But I was only on 1.0.7 for a very short while, and quickly updated. One
system is on 1.0.9 Stable and the others are on the very latest SVN HEAD.
I recently decided to have a bit of a spring clean on the audio files,
and to my horror found that only the very first few files on all
machines were mixed (I presume those were the ones when I was still on
1.0.7) while the rest were all still there as separate -in and -out
files. This is despite the console showing that soxmix was being called
to join the two files and remove the individual parts each time a call
was made or received, and with no errors. Soxmix is installed, and does
work - I can copy and paste the command from the console output at the
command line and the single audio file is then properly created and the
individual in and out parts deleted.
I've just tried changing to using MixMonitor in extensions.conf on the
1.2x machines and this works perfectly for normal outgoing or incoming
calls that don't involve queues.
But this obviously doesn't solve my queues mixing problem on the 1.2
machines, nor any of the mixing problems on the 1.0.9 machine.
Has anyone else come across this issue? Any pointers please?
Thanks,
Faris.
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Re: [Asterisk-Users] recording queue calls
Tom Lynn wrote: Faris, Is there a way to have * send save these in an off-server location? Or have * e-mail them via smtp and then delete them from the server automatically? I'm sure there is a very technical way of doing it. For example if I remember correctly you can set your own script to run to join the two sides of an audio recording (something I tried using to solve the problem I'm having with joining two sides of a conversation, but with no luck). You could add a mail command to the script to do what you want. I'm afraid I don't remember the exact details of how this is done, but I think I came across it when searching for asterisk call recording on Google. There was a full script for an alternative mixing solution. Or you could use rsync, running every hour or every day as a cron job, to synchronise the /var/spool/asterisk/monitor directory on the machine tasking the calls with a second server. e.g. rsync -e ssh -avz /var/spool/asterisk/monitor/ [EMAIL PROTECTED]:~/monitorbackup You'd need to set up a passwordless private/public key combination for this to work automatically though. There may also be issues with the rsync job using too much bandwidth and causing audio quality problems. Hmm... Well, I'm sure someone who know more than me on this topic will pipe up on this! Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with current chan-capi-cm
Armin Schindler wrote: On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote: Armin Schindler schrieb: On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote: Hi, as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in any direction. Going back to a version from Dec 4th gives a working system again. [..] error 0x1103 is 'queue full', so the capi driver (isdn card) does not accept further voice packets. Did you try latest CVS (11.12.)? I cvs checkouted today. Can you please provide a full log and one with the older, working version too? set verbose 50 enough? Or another type of log? The file is massive, and I don't want to waste everybodies bandwith. Use 'set verbose 5' and 'capi debug'. You can send the logs to me directly. Armin I know this won't help anybody debug or solve the issue, but I thought it might help to know that others are having the same problem. Mind you I'm using quiet an old Asterisk 1.2 svn version (three weeks ago), with the chan_capi-cm of about three weeks ago too. I didn't even realise I had a problem until a few days ago. For me it works fine after Asterisk is restarted, but at some point later it just stops - it dials, but no audio. I will check out the latest Asterisk and chan_capi-cm and try again over the next week or so. (This is with an AVM Fritz card (BT Speedway) under RH9 btw) Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN 2e Cards
Simon Faulkner wrote: I have tried Billion HFC-PCI and Eicon Diva cards for ISDN 2e in the UK but both seem to have drawbacks/advantages. I need to build a new Asterisk box for my tiny business (1 x ISDN2e from BT and 1 x IAX link from Gradwell) Is anyone prepared to go out on a limb and say which card they prefer and why? TIA Simon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users BT SpeedWay PCI cards are ideal. Work perfectly with chan_capi-cm for ISDN2e and Business Highway here in the UK. They are basically AVM Fritz cards badged by BT. I have a stock of brand new ones if you need, or alternatively they are often advertised on the auction sites (new and used). Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN 2e Cards
Avi Miller wrote: John Daragon wrote: I'd second that. For a single ISDN2e connection the AVM Fritz card is really hard to beat/ Yeah, single is the key word there. I have 2x ISDN2 (OnRamp2 in Australia) and the AVM Fritz cards are a nightmare. Replaced the two cards with an Eicon Diva V-4BRI (so I have two extra ports if necessary) and my Asterisk box is just incredible now: Almost zero echo across the board and much lower processor utilisation. cYa, Avi Absolutely right. I have managed to get two of these cards running correctly in one of my machines thanks to the instructions on voip-info, but I can't say I'd be able to easily reproduce it -- I seem to remember I had to fiddle around with the drivers for ages and ages :-( Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] In France asterisk never detect hang up. Why ?
[EMAIL PROTECTED] wrote: Hello everybody :-) This are my first line french zapata.conf settings. I have 3 like this, with only rx/tx gain a little bit different levels. Running well. Best Regards, Francois BERGERET, France. usecallerid=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2 txgain=6 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=3 busypattern=500,500 signalling = fxs_ks channel = 1 -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de asterisk user dupont Envoyé : vendredi 18 novembre 2005 13:33 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] In France asterisk never detect hang up. Why ? Hello. I am sorry my english is not good at all. When i have a call from a fxo port of a tdm400p, asterisk waits one minute before detecting that the caller has hang up his phone. I have in my extension conf : answer background (the prompt is 40 second long) dial (on fxs port) confgured for 30 seconds ringing. if the caller hang up at the begining of the background prompt, asterisk waits until he make ring the phone on the dial command for the all 30 secondes before detecting the hang up. Do you know if there is a way to repair that ? here is what i see on asterisk when the caller hang up IMMEDITALY after the test prompt begins : *CLI -- Starting simple switch on 'Zap/4-1' -- Executing Answer(Zap/4-1, ) in new stack -- Executing NoOp(Zap/4-1, 0675458745) in new stack -- Executing Set(Zap/4-1, TIMEOUT(response)=20) in new stack -- Response timeout set to 20 -- Executing BackGround(Zap/4-1, barge) in new stack -- Playing 'test' (language 'fr') -- Executing Dial(Zap/4-1, Zap/2|0675458745|30) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/4-1 -- Attempting native bridge of Zap/4-1 and Zap/2-1 -- Hungup 'Zap/2-1' == Spawn extension (reseau, s, 5) exited non-zero on 'Zap/4-1' -- Executing Hangup(Zap/4-1, ) in new stack == Spawn extension (reseau, h, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' In my zapata.conf i have : language=fr default=fr relaxdtmf=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 cidsignalling=v23 usecallerid=yes group = 1 context=reseau signalling=fxs_ks callprogress=yes busydetect=yes callerid=asreceived busycount=5 pulse=yes In my zaptel.conf i have : loadzone=fr defaultzone=fr fxoks=1-3 fxsks=4 If anyone can see what is wrong he will really help me. thank you. Your English is better than my French :-) Making the TDM400p detect hangups can be hard. I had it working OK with pre-1.2 versions, but now in 1.2 stable I'm also having some problems again. I'll investigate in more details eventually. For now, the only thing I can suggest is that you add: hanguponpolarityswitch=yes in your zapata.conf In the UK, hangups are signaled by a polarity switch, and since sometimes the UK and Europe do the same thing, I'm hoping this will be the case for you too. However, even with this option enabled, like I say, I'm having some small problems with 1.2 stable. I hope to have time this weekend to investigate and see what is going on. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] small chan_capi-cm 0.6 capicommand(echosquelch) problem?
I now have chan_capi-cm 0.6 working with Asterisk 1.2 RC2. But I have discovered a small problem. I have a mix of analog and ISDN (BRI) lines coming in to my Asterisk box. Both types of lines are fed into the same set of contexts. In the previous version of chan_capi-cm that I was using (0.53 I think), I was using to CapiNoES command before all voicemail commands in order to disable echo suppression because messages left in voicemail was unintelligible without it. However, with chan_capi-cm 0.6 I need to switch to capicommand(echosquelch|no) instead of CapiNoES. This is fine, but while CapiNoES didn't fall over when executed on a call that came in via an analog line, with capicommand(echosquelch|no) asterisk stops processing the call when called and displays an error to the effect that you can't use this command with a non-capi call. As a result I'm going to need to add a lot of code to my extensions.conf to take this into account which I'd really rather not do unless I have to :-) Armin - is there any chance of changing the behavior of capicommand(echosquelch|no) to simply generate a warning rather than cause Asterisk to stop processing the call? Using it on an analog channel isn't really a fatal error in the grand scheme of things? Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs
[EMAIL PROTECTED] wrote: Thanks Armin, this version is working, but I still have an undefined symbol in another module: [pbx_wilcalu.so]Nov 5 18:51:12 WARNING[11348]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Nov 5 18:51:12 WARNING[11348]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed! Can you also help me on that issue? Thanks and Regards Markus To my knowledge, that module has nothing to do with CAPI. I don't honestly know what it does. (will call you) What I can say is that with 1.2 RC2 (latest from CVS) and chan_capi-cd 0.6 (latest from sourceforge cvs) as of 20:00 12/11/05 GMT on RedHat 9, I get exactly the same error when loading on a freshly sanitised system with all traces of previous asterisk installations removed. HOWEVER, if you add a noload = pbx_wilcalu.so in modules.conf you can make the error go away. (but this is probably a bad thing since I don't know what that module does!) But unfortunately, for me at least, I then end up with errors about: app_capiCD.so app_capiHOLD.so app_capiRETRIEVE.so app_capiECT.so and app_capiMCID.so For example: [app_capiCD.so]Nov 12 20:24:55 WARNING[19197]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: ast_capi_MessageNumber Nov 12 20:24:55 WARNING[19197]: loader.c:554 load_modules: Loading module app_capiCD.so failed! # Ouch ... error while writing audio data: : Broken pipe No matter which of the modules you comment out above, the same thing happens -- the error is always about app_capi_MessageNumber Armin (or anybody) -- have I missed something out/done something wrong, or is it a compatibility issue between chan_capi-cm 0.6 and Asterisk 1.2 RC2? Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A2Billing Authentication Refused
Sam Tam wrote: Try o reupload the mysql database again to see if that work? Sam *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Omar McKenzie *Sent:* 03 November 2005 00:27 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [Asterisk-Users] A2Billing Authentication Refused Hi I installed A2Billing on Asterisk (running on FC4) , mysql When attempt to logon using username/password: root/myroot , or username/password: admin/mypassword gets error ‘Authentication Refused, please check your login/password’ Also check to see if using [EMAIL PROTECTED] instead of just root will solve the problem. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fritz!Card PCI ver2.0
Patrick wrote: On Wed, 2005-11-02 at 19:33 +, Faris Raouf wrote: Please note, however, that somewhere in the wiki it suggests that you modify the AVM driver code slightly. I found this stopped it compiling, and that simply leaving the code as it is worked fine. Then please add a note to that page on the Wiki or change the text to reflect that. Regards, Patrick Good idea. Will do! Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Time based call direction
Rene Nelson wrote: I would like to manipulate phone call direction to voicemail for lunch, after hours etc, but am unsure how to do this. Could someone point me to a howto or quickly explain the concept? Thanks Neri Hi Neri, The command GotoIfTime() if your answer here. See http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime for more info. Now, assuming we are talking about a situation with say one main voicemail extension to collect messages from callers calling the main company number The call comes inthen do a gotoiftime to branch to two places: first place is normal, second place is lunchtime. Now, for each of these, first play an appropriate message with the Playback command, then record the message left using the voicemail command with the s option. The s option means play nothing, so basically you aren't using the built-in outgoing messages that the voicemail system has and instead will have first used some custom message via the playback function e.g. exten = 4321,111,Playback(lunchtime) exten = 4321,112,voicemail,s12345 where 12345 is your main voicemail box. 4321 and 111/112 are also just numbers picked at random for use in this example. See http://www.voip-info.org/wiki-Asterisk+cmd+Voicemail for more info on using voicemail in this way. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fritz!Card PCI ver2.0
Stephen Arulraj wrote: Anyone knows how I can use this ISDN card for asterisk as a BRI trunk interface? Thanks, Stephen Hi Stephen, Is this a new version of the AVM card? If not (or even if it is), you may find the following pages helpful: http://www.voip-info.org/wiki/index.php?page=Asterisk+AVM+Fritz+CAPI+Driver+Install http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI Please note, however, that somewhere in the wiki it suggests that you modify the AVM driver code slightly. I found this stopped it compiling, and that simply leaving the code as it is worked fine. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000
Erick Baum wrote: We're having a rather serious echo problem using the Grandstream GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking that might be an easy fix. The echo seems to be worst on internal SIP to SIP calls but you do get it every once in a while on outgoing calls through the PRI. It's not the speakerphone echo problem, we're running the 1.0.1.12 http://1.0.1.12 firmware that pretty much fixes that. It seems like most of the echo cancellation functions are for outgoing calls through the phone company. Is this a more likely a phone problem? We've got about 50 of these phones all doing the same thing. -- | Erick Baum Hi Eric, I only have two of these but have not come across an echo problem with them on SIP at all. Nothing unusual needs to be done to the config at all. So although I don't know how to help you, you can be assured that the problem is solvable and not down to the actual phones themselves, if you see what I mean? Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 601 XHTML microbrowser
Chris HARIGA wrote: Gary Reuter wrote: On 10/26/05, *Chris HARIGA* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have a show parked calls php script for my Polycom IP600 phones. If U are interested let know and I can email it. Even if Sean doesn't want it, I do! All examples can be helpful. :-) Why not put up a page on the wiki linked from the polycom page(s)... If formatting is problematic, just note it on the page and I (and others) can help make look nicer for the wiki. -Gary ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, I will edit the wiki and I will upload my polycom scripts: parked calls, sip users status, meetme status, queues list and phones status tonight. Best regards, Chris HARIGA Please! I've bee wondering if anything was available along these lines. All that space on the LCD with nothing to do! This will be of huge benefit to a large number of people - thanks you. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 601 XHTML microbrowser
[EMAIL PROTECTED] wrote: At 08:38 AM 10/27/2005, you wrote: http://www.voip-info.org/wiki/view/Polycom+XML+browser+scripts+for+asterisk Best regards, Chris HARIGA Thanks. Is it possible for someone to provide a basic explanation of how to implement this for us less technical minded people? From what I can tell, it looks like one needs to modify the ipmid.cfg file. I'm guessing the mp.proxy, mp.main.home, and/or mb.limits.nodes values need to be modified. My guess is that I simply copy the files to an appropriate folder and modify the mp.main.home setting to point to that folder. The mp.proxy and mp.limits.nodes values can be left null? Thanks, Doug Um, well the easiest thing to do is: 1) stick the files on your webserver somewhere (e.g. www.mydomain.com/pcom) 2) Modify the top lines of each .php file so that the ip address is that of your asterisk server, and the username and password match a username and password configured in manager.conf 3) Change the config on your polycom phone via the web browser rather than hacing away at the xml. Once logged in, click on the microbrowser link option (I think it is in the general section), leave the proxy server line blank, and just put www.mydomain.com/pcom or wherever in as the address. Click on OK. The phone reboots and the xml config files will automatically update (assuming you allow TFTP uploads on your TFTP server). And then it just works! Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID setup from goiax.com
trixter aka Bret McDanel wrote: I dont know then that was cut and paste from what I have working ... maybe actual log dumps of the error? On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote: That is What I stated in the email.. my GOIAX #. not the DID #. That is not the issue. Is this still ongoing? If so... when you get an error like [EMAIL PROTECTED] in the log, it is a good indication that something is looking for a priority s in the context (I think). In my case I set up goiax yesterday and had this exact error. The solution was simply to have s priorities in the context in extensions.conf that my context in iax.conf was pointing to for goiax. NOTE: In the following, mygoiaxnumber should be replaced with the actual number (not DID number) that you see on your screen when you first register, just above you password. iax.conf: register = mygoiaxnumber:[EMAIL PROTECTED] [mygoiaxnumber] context=goiaxinwards etc etc extensions.conf: [goiaxinwards] exten = s,1, Answer() etc AND NOT: exten = mygoiaxnumber,1,Answer() etc (which is what I originally had and which did not work for me in my particular case - I got the [EMAIL PROTECTED] type error) Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 Disconnect tone detection in Spain, Peru and Colombia ?
makevuy wrote: Hello everybody, I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip installation. I actually have a problem when a PSTN user calls and hangs up. The disconnect tone is not detected by the SPA, the the call continues and, for example, leaves an empty message on the voicemail before hanging up (because * hangs up). How could resolve this problem?. I set, Detect Polarity Reversal:yes Detect Disconnect Tone: yes, with the default value. Thanks a lot for your help ;) I've never used one of these (but I'd like one). However, if it is not detecting the disconnect tone, it could be that your telephone service provider is providing a tone is not the same as the one the unit is expecting. For example in the UK you need to change the settings for the disconnect tone from the defaults. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hang-up Detect - Yet Again
* answers the call, but if the incoming caller hangs up, * does not release the line. Is there a polarity reversal on hangup (those clicks you hear maybe)? If so then you may find that using the CVS-HEAD version of Asterisk will help hugely. Put hanguponpolarityswitch=yes in your zapata.conf But I'm positive that the definitive answer to most people's hang up detection problems would be some code in chan_zap to detect a tone other than busy on hangup. For example on my line is it a continuous tone. On yours you get a dial-tone. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100p Problem, randomly hungup pstn line
Right at the end of your Zapata.conf you have: #include zapata_additional.conf hanguponpolarityswitch ;Include genzaptelconf configs #include zapata-auto.conf Remove that hanuponpolarityswitch as you already have hanguponpolarityswitch=yes earlier on, and I don't know what having the second one, with no =yes/no would do. Then, with regards to logging, in logger.conf (and not logging.conf like I said in my original message -- but you noticed that already :-) ) Looks for the console = and messages = lines. At the moment they will be something like console = notice,warning,error and messages = notice,warning,error If you add ,debug without the quotes to either line, debug information will then be shown on the console (if you add it to the console line) or in /var/log/asterisk/messages (or somewhere similar) if you add it to the messages line. To view the contents of /var/log/asterisk/messages in real time (constantly updated), use the following command from the command line (not asterisk's command line but your Linux box's command line) tail -f /var/log/asterisk/messages tail, by itself, shows the last 5 or so lines. Tail, with -f, keeps looking, and so you get a scrolling log of what is being added. Very useful. Restart asterisk for the new logging changes to be shown. You should then be able to see some hopefully useful debug messages as your call progresses. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100p Problem, randomly hungup pstn line
I installed this card, everything work, i can make call and receive call with no echo and great sound quality, but after between 5 to 50 secs the call disconnect by itself, in the log i don't see nothing revelant. In logging.conf, try enabling debug logging to the console and/or to /var/log/asterisk/messages to see if you can find the cause. chan_zap.c displays a lot of useful debug info if you enable the debug level logging. Also please post your zaptel.conf and zapata.conf config so we can have a look. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice Prompts, what do you think? Good voice.
Gregory, My advice is to go for it. Allison is nice but there are times when her accent doesn't pass the International test ( e.g. everyone I've ever spoken to in the UK roll about on the floor laughing when they first hear her in the Voicemail prompt, telling you to leave a message ). Others will probably disagree with me (in fact there was a discussion on this very recently), but if you do go for it, I would personally like to see the recordings in .wav format (8k, mono, PCM, 16 bit) - Wavelab allows this to be done very easily. I save all my final prompts in this format because they provide great sound quality compared to GSM, and also allow for high quality sonic idents (something I'll be posting about soon. Watch this space). If people prefer them in different formats, they can then use the wavs as the basis and re-encode them (e.g. gsm). But like I said, that's just my opinion. I'm not saying this is what everybody wants, or what you should definitely do. Faris. -Original Message- From: Gregory Wiktor - ADCom Corp. There is a good chance I will do it, but want some feedback. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: channel offhook state
Yes indeed. There have been huge changes to chan_zap.c in CVS-HEAD compared to 1.09. In 1.09 Stable there are a lot of problems with handling call hang-ups. CVS-HEAD, of 28/08 was much better. But even though it did improve things, it wasn't quite right. In particular I found two problems with polarity reversal detection in chan_zap.c for which I have created a patch (this is now in CVS-HEAD). Please see http://bugs.digium.com/view.php?id=5191 for more details. Please note that you'll need to use answeronpolarityswitch=yes and/or hanguponpolarityswitch=yes in your Zapata.conf to make full use of the polarity detection code. You will also need to be very careful if CID is sent on a polarity switch too -- you may need to make it detect on the 0th ring or you could suffer from immediate hang-ups on ring. Unfortunately I've received a problem report with this modification. Any updates Magnus? I'm hoping it is all down to the ring that CID is detected at, and that by changing it to 0 or 1 all will be well again. But anybody who has had problems with hangup detection in the past should try CVS-HEAD and play with the options above to see if it improves things. Having said all this, things are still not perfect: For UK (and possibly other European countries) we still require a way for Asterisk to detect the continuous tone that indicates a remote party hangup on a POTS line. The Sipura 3000 uses this method and I believe it works quite well, though I've not tried it myself. Faris. -Original Message- FWIW, there were a couple of channel zap changes made in the last couple of days to cvs-head. Don't have a clue whether those fixes addressed the problem you're talking about. Has anyone else experienced the same problem, where a Zap channel gets stuck in off-hook state? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 1:45 PM To: asterisk-users@lists.digium.com Subject:[Asterisk-Users] FW: channel offhook state -Original Message- From: Jacqueline Lee [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 11:46 AM To: asterisk-users@lists.digium.com Subject:channel offhook state We are using a digium card (TDM400) with asterisk for our access to the PSTN. Initially when the server starts, all the zap channels on the card are in the onhook state. As soon as a channel is used (for inbound or outbound PSTN calls) the corresponding channel goes into offhook state, and stays in offhook state, even after the call ends; Asterisk log shows that the channel was hungup. Most of the time, the channel is still usable to make more PSTN calls, even though it shows in offhook state. Occasionally the channel becomes unusable for making PSTN calls (usually channel 1). The symptom is Asterisk and the client show the PSTN call was established, but the destination PSTN number never really receives the call. Shouldn't the channel go back to onhook state once the call hangs up? Is the persistent offhook state causing the channel to eventually become unusable? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P not detecting hangup and not hanging up
Canuck15, No, I hadn't played with the gains. But I've now done so and no difference unfortunately. Thanks for the suggestion though. I have discovered that after Asterisk has answered the call and the remote caller has hung up, if I lift the receiver on a phone connected to the line (in parallel with Asterisk), Asterisk then DOES instantly hang up. Would it be reasonable to assume the voltage drop caused by lifting the receiver causes this? It only happens when I set the BATT_(whatever it was) in wcfxs.c to 8. If I set it to a lower level, Asterisk won't even answer at all and so nothing works. Also possibly relevant: When I disconnect Asterisk completely from the equation and just answer the remote caller myself, when the remote caller hangs up the line does not actually drop: Instead I just get a disconnect (or number unobtainable) tone. Could this be the problem (i.e. there's no actual voltage drop happening to signal the call has ended)? Or is there some sort of other change in the line that I wouldn't detect audibly? Could it be that any inaudible voltage drop might be happening too quickly for zaptel to detect? What might I change in the source code to see if this is the case? Does nobody else in the UK use these cards? I'm sure that's not the case. So if you do use them, please stand up and be counted -- did you have to make any adjustments or did it just work out of the box? Incidentally, when callprogress=yes, Asterisk goes nuts and keeps detecting strange things happening: Essentially every time the CLI comes through (polarity reversal) between rings, asterisk picks up and hangs up (though not physically - the caller hears ringing). This may or may not be related but have you tried adjusting your RX and TX gains? I see both are at the default (0.0) which leads me to believe you have not. Search the Asterisk Wiki for the procedure. Stevanus, I think the hanguponpolarity switch is relevant to a patch to to Zaptel that may or may not have actually been added to the released version. I'm not sure. However, thanks for pointing this out -- I've tried it too and didn't get anywhere. I have similar problems like you. In the past, I did adjusted my RX and TX gain, but didn't know if it has been optimal yet. Fxotune is seemed do not working, perhaps caused of my asterisk's version ( I use stable v1.0).. Just curious, is rx and tx gain really a sole setting option here in order to make things the way it's meant to be? Or is there others? FYI, my tdm04b occasionally don't detect call-in as well as hangup signal. I've searched in the wiki and have activated hanguponpolarity swicth. But I don't notice any difference at all. Any help would be greatly appreciated. (I've asked this in another thread, but got no respon :( ) SUMMARY OF THREAD: hardware=TDM400P 2xFXS, 1xFXO. Location=UK. *ver=1.0.9. Zaptel 1.0.9.1. Problem: Asterisk does not detect that the remote caller has hung up and carries on as though nothing has happened. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P not detecting hangup and not hanging up.
Can anyone suggest where I might begin looking for an answer please?
I have just installed a TDM400P (2x FXS and 1x FXO modules installed)
The first problem is that it does not seem to be able to detect if the
remote party has hung up when a call comes through on the FXO. For example,
if someone calls in, and then hangs up at any time after it starts ringing,
Asterisk carries on as though the caller never hung up.
I've tried raising BATT_THRESH to 8 in wcfxs.c and re-compiling zaptel (this
was the only thing that Google came up with to help me, although others do
seems to have had similar problems to mine at various times), but it has
made no difference at all.
The second problem is that Hangup does not hangup. The channel stays open
until I stop asterisk.
Note: When MAKING a call on the FXO, when I terminate the call on my SIP
phone the line does drop correctly. The problem appears to be related to
incoming calls only.
I'm based in the UK. Using RedHat 9 with zaptel-1.0.9.1, asterisk-1.0.9 (and
chan_capi-0.5.4)
Thanks in advance for any ideas.
Faris.
*
Here's my initialisation script:
modprobe zaptel
modprobe wctdm opermode=UK
/sbin/ztcfg -
capiinit
safe_asterisk
zapata.conf
[trunkgroups]
; nothing in here
[channels]
rxwink=300 ; (I tried commenting this out. Make no difference)
usedistinctiveringdetection=no
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=no
sendcalleridafter=1
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
immediate=no
progzone=uk
; module 0 on card is an FXS
signalling=fxo_ks
language=en
context=sip
channel = 1
; module 1 on card is an FXS
signalling=fxo_ks
language=en
context=sip
channel = 2
; module 2 on card is an FXO
signalling=fxs_ks
language=en
context=faris
channel = 3
zaptel.conf
fxoks=1-2
fxsks=3
loadzone=uk
defaultzone=uk
and in extensions.conf
[faris]
exten = s,1,NoOp(cid=${CALLERID})
exten = s,2,Wait(10)
exten = s,3,Answer
exten = s,4,Wait(1)
exten = s,5,Playback(some-long-message)
exten = s,6,Hangup
The long wait(10) is just there to see what happens. Removing it makes no
difference. Basically whenever a call comes in, no matter when the caller
hangs up, Asterisk continues with the call to the end (i.e. plays long
message).
What's more, the Hangup at the end has no effect. The line is not dropped.
The line is not ever dropped in fact.
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[Asterisk-Users] RE: GrandStream GSX-2000 strangeness
Thanks to all who replied on this. But amazingly I think I've solved the problem. Basically I did a factory reset (select reset via the Menu key then enter the MAC address [as shown on the white label under the phone], then press Menu key again) and re-entered the necessary config details on both phones. And this has solved the problem completely (so far). Can you possibly give it a go to see if it solves your problem too Mark? What I don't (yet) know is the cause. It could be that the last firmware update somehow corrupted some of the existing settings, or it could be that prolonged use causes the problem, requiring a factory reset. I can still duplicate the sound problems in a way though. If you login to the phone's web config page, while listening to the phone giving a dial-tone, I can hear the same type of glitches happening every time I click on any links. So it seems that the source of the glitches is probably the phone doing something internal and getting stuck in a loop. GrandStream support also replied to my email on this subject, suggesting the possibility that it may be a hardware problem and asking for me to send them a copy of the phone's config. Faris. Message: 19 Date: Wed, 10 Aug 2005 22:37:13 +0100 From: Mark Brown [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] GrandStream GSX-2000 strangeness To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I have exactly the same problem with my GSX-2000. And am running the latest firmware. Although they seem like really cool phones in theary, practically I think they still have a far way to go. I personally can't believe they actually launched the 2000's with all the problems they actually have. Many of the advertised features on the GS website have still never been implemented in the actual phones themselves. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GrandStream GSX-2000 strangeness
I have a really baffling problem. A couple of months ago I purchased a pair of GrandStream GSX-2000 phones for use with Asterisk. At first all was well. But recently I've noticed terrible sound quality problems. Basically the sound will glitch or stutter randomly from time to time. Now, what is interesting is that this happens even with the phone totally disconnected from any network. You can hear it on the dial-tone that the phone itself generates if you press SPEAKER or lift the handset, with no network cable plugged in. So whatever is causing this glitching would seem to be being generated within the phone itself, and not coming from an external source. (It isn't the power, as the two phones are in different buildings in different towns. Besides, both are connected to UPSs). To me this seems to indicate a fault with the phones themselves. But for both of them to develop the same fault at the same time seems odd. I asked on the voipuser.org forum if anybody else had had similar problems, but everybody who responded said all was well with their phones. But given the wider reach of this list, I thought I'd ask here as well. Both phones have the latest firmware from the GS website. Does anyone have any ideas? Has anyone had anything similar happen to them? Faris. p.s. This is my first post here. Please be gentle with me :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
