[Asterisk-Users] Compile problem SuSE 8.2
I am trying to compile * on SuSE 8.2. When doing the make install in /usr/src/zaptel I get the following error. ** /usr/src/linux/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules freeIn file included from /usr/src/linux/include/linux/highmem.h:5, from /usr/src/linux/include/linux/vmalloc.h:8, from /usr/src/linux/include/asm/io.h:47, from /usr/src/linux/include/asm/pci.h:40, from /usr/src/linux/include/linux/pci.h:654, from zaptel.c:38: /usr/src/linux/include/asm/pgalloc.h: In function `flush_tlb_page': /usr/src/linux/include/asm/pgalloc.h:201: internal compiler error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://www.gnu.org/software/gcc/bugs.html for instructions. make: *** [zaptel.o] Error 1 *** Any ideas about where to look for the problem would be appreciated. Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Results SUSE 8.2 + server size
Hello All, Thanks to those that responded to my problem of compiling on SUSE 8.2. I was not able to get the compile done so decided to put RedHat 9 on this system. After getting a RedHat supported NIC and RedHat installed, Asterisk compiled cleanly, one SIP phone is connected and voice mail works. No other tests have been run yet. A couple of days ago, Michael Farnworth asked about the smallest system that was running Asterisk. This one is a Pentium 100, 32 MB RAM, 8 GB disk. I don't expect it to handle much load but for a test platform it seems ok to use while trying to find a low cost P4 system. Regards, Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No ISA tormenta card message]
I am getting the following messages that seem to be coming from Asterisk. In the system there are no ZAPTEL cards installed. I did uncomment ztdummy in the Makefile in /usr/src/zaptel before running make install. Any ideas on how to get rid of this message. I looked through all the config files (installed the sample ones then modified sip.conf, extensions.conf and voicemail.conf, rest are as installed) but did not find anything that looked right. Can someone please point me toward what I am overlooking? Thanks Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Setup
My config that works for number 1 is below. Everything works including the voice mail waiting light. All of this for * was copied from or based on: http://www.automated.it/guidetoasterisk.htm. This is an EXCELLENT getting started site. Can't help you with #2 but am sure others can. sip.conf for extension 2000 [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=9overthruster7 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=2000 ; Activate the message waiting light if this ; voicemailbox has messages in it extensions.conf exten = 2000,1,Dial(SIP/2000,20) exten = 2000,2,Voicemail(u2000) Budge Tone config: SIP Server: 192.168.0.110 (my * box) SIP Userid: 2000 (userid is same as extension Authenticate ID: 2000 Authenticate password: 9overthruster7 Send DTMF: Via SIP info (in order for the dtmf to be recognized by voicemail) Hi People, Ok i've tried everything I can think of but cant get this to work. Can someone please give me an example of their sip.conf settings and also the details of the settings in their grandstream phone to allow: 1. Grandstream phone to register with asterisk when on same lan. 2. Grandstream phone to register with asterisk when phone is behind a nat. Regards, Aaron. - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Proper Credit: Re: Grandstream Setup
I was incorrect in my citation of credit in the below email. Properly the credit goes to John Todd for the Asterisk config examples. His excellent article is at: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1 Sorry for the goof-up. Robert My config that works for number 1 is below. Everything works including the voice mail waiting light. All of this for * was copied from or based on: http://www.automated.it/guidetoasterisk.htm. This is an EXCELLENT getting started site. Can't help you with #2 but am sure others can. sip.conf for extension 2000 [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=9overthruster7 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=2000 ; Activate the message waiting light if this ; voicemailbox has messages in it extensions.conf exten = 2000,1,Dial(SIP/2000,20) exten = 2000,2,Voicemail(u2000) Budge Tone config: SIP Server: 192.168.0.110 (my * box) SIP Userid: 2000 (userid is same as extension Authenticate ID: 2000 Authenticate password: 9overthruster7 Send DTMF: Via SIP info (in order for the dtmf to be recognized by voicemail) Hi People, Ok i've tried everything I can think of but cant get this to work. Can someone please give me an example of their sip.conf settings and also the details of the settings in their grandstream phone to allow: 1. Grandstream phone to register with asterisk when on same lan. 2. Grandstream phone to register with asterisk when phone is behind a nat. Regards, Aaron. - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec
Do you have a 100 or 101? You have indicated different models in your postings. Were you able to get Call Transfer and Call Waiting working with your Asterisk system and other phones? Which version of the Grandstream firmware do you use? There most recent on their website this weekend was at least 2 version numbers higher than what came on my phone in August. Think that they are making improvements rather frequently. Robert On Wed, 15 Oct 2003, Jon Pounder wrote: The Grandstream 101 I'm using is a piece of junk but I don't have the same problem with it. What don't you like about the grandstream ? (I am not looking to flame you, but was considering buying and if there are problems would rather find out beforehand) Nothing works. Call transfer and call waiting, in particular. (Well, almost nothing; vm notification does work) There is no place to plug in a headset, and since I do a fair amount of tech support and longish conference calls, that's a big deal for me. However, keep in mind that I have an old, no-longer-manufacturered model (the Budgetone 100). Don't take my frustration with my outdated phone as a sign that you should dismiss Grandstream out of hand - I just don't like my 100. -- JustThe.net Internet Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec
I only have 1 but the absolutly only time it has to be rebooted is when I change a parameter or upgrade the firmware. It has run for weeks without any problem. Another poster mentioned the 10 vs. 100 Ethernet speed. Maybe Grandstream can upgrade the interface in future hardware. I don't imagine that the price point for 10/100 is much different than 10 these days. One option I would definatly like is the ability to turn off the ringer. Since my testing ususally happens after my wife goes to bed it would help NOT to have the audible ring but only the visual indication! Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream ringer
Michael, That would work for me too. If the volume can be reduced (maybe to zero or almost zero) then my request for the ability to disable it is not needed. Since the volume of the speaker and handset can be controlled maybe the GS folks can include a patch in the next release of the firmware to also handle the ringer. They monitor this list so maybe will jump in with some feedback to us. Robert Better still I would like volume control over the ringer as the default tends to be rather loud and annoying to other people in the same room. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
The only thing that is wrong is that there seems to be some expectation of Digium that they have to tell things... The source code is available. If someone isn't happy with the Digium methods then they should find a solution and post it to the list and/or one of the several Asterisk Wiki's that are around. Digium has no obligation in this regard. OSS doesn't mean free, OSS doesn't mean no secrets. It means Open Source Software. Alot of folks (me included)sometimes incorrectly equate the term OSS to mean FREE. Digium has already given for free much more than the typical telephony hardware manufacturer. I think its pretty clear that they like using common timing source across all platforms. I am looking forward to the postings of alternative timing solutions. Robert Also as I'd written. It seems we're arguing the same side of the argument. :-) My compliant was not that the timing was needed, but that Digium seems so damned secretive about it. I mean this is OSS -- just tell us that having a common timing source across all platforms makes things really easy, you don't have to screw with looking at writing an alternative driver for RTC or USB or XYZ and hey, we happen to make some money selling these boards too. If you are running a SIP-only * box then here are some alternative timing drivers, point them at some URLs and oh by the way, we didn't write 'em, we don't support 'em, they seem to work fine with others though. What's wrong with that? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
Andrew, I am running it rather well on a original Pentium 100 Mhz, 32 MB RAM, no USB adapter. I agree with you this would not be an ideal setup for a business but in a home it will work rather well. I think it'll handle 2 CO analog lines fine. Yes, my wife thinks its overkill. Probably is, but guess what, if I want to change it I can. If I want to try and integrate another type of card, I can. If I want to connect and control my ham station, I can. AND best of all, if I want to develop and use another timing source, I can. and so can you. Regards, Robert I don't think you'll be running Asterisk on anything older than a P2 to be honest, and even then the utility is severely hampered due to everything being done in software on *. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
Chris, Good point. As I understand it, the Asterisk software requirement was to be a PBX between normal telephone lines and VoIP. Maybe even it was just to replace the expensive PBXs. As such seems to me that it clearly met and exceeded its design requirements since it utilizes the hardware boards that were in the original design requirement. Don't think anyone can dispute that. Its created by Digium for Digium hardware. Everything lese is gravy. 73, Robert Any discusion about PCI cards, RTC timmers and the like is in a complete vacuum unless you know what exactly it is that the software is required to do. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie question: Meetme
Yes, I am a newbie too. I am having a problem with meetme. From what I have seen it will work without a Digium card but with audio problems. My goal is just to see how it works not the quality of the audio. When I dial into the conference room the following message is played: That is not a valid conference number. On the console I get: unable to open pseudo channel. As indicated in previous posts I do not have any Digium cards in the system. When making the zaptel part of the system I did uncomment ztdummy.o in the MODULES= line. Extensions.conf contains: exten =2663,1,Meetme,9876 meetme.conf is: [rooms] conf = 9876 If meetme doesn't work at all without a real card that is ok. It can wait. If it'll work at least somewhat with ztdummy then obviously I've missed something. Any ideas? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie question: Meetme
On Wed, 2003-10-15 at 14:16, rnc Info Lists wrote: Yes, I am a newbie too. I am having a problem with meetme. From what I have seen it will work without a Digium card but with audio problems. My goal is just to see how it works not the quality of the audio. When I dial into the conference room the following message is played: That is not a valid conference number. On the console I get: unable to open pseudo channel. As indicated in previous posts I do not have any Digium cards in the system. When making the zaptel part of the system I did uncomment ztdummy.o in the MODULES= line. did you actually install the module into your running kernel? -- Steve, I have the following line in /etc/modules.conf: post-install ztdummy /sbin/ztcfg Is that what you mean or did something else need to be done that I missed. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie question: Meetme
Did you modprobe ztdummy before running asterisk ? I have meetme running in one * box without zaptel harware. I just tried that. The following messages are given: /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed ztdummy.o is in /lib/modules/2.4.20-8/misc so I tried: modprobe /lib/modules/2.4.20-8/misc/ztdummy.o and got: modprobe: Can't locate module /lib/modules/2.4.20-8/misc/ztdummy.o Permissions of that file are: -rw-r--r-- I noticed in the Makefile of zaptel that PRIMARY=torisa. Can zaptel be remade (or whatever that is called) without having to redo Asterisk? If so maybe I try PRIMARY=ztdummy. What do you think? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I give up!!
Asterisk... Linux... You get what you pay for. And it's free :P Thats true but free (cost) doesn't have to mean cheap (quality). Maybe what we need is to collect business requirements and build a configuration for a typical system. (hardware spec. and actual config files) What Dave has listed is a good start. Then folks will have a starting point. If cost is the driving factor then obviously there has to be a compromise in functionality. Knowing what a specific functionality costs to implement would help people quoting installations. (example the transfer situation that GS phones don't handle but seem to work with one of the more expensive phones and the rest as GS). While I don't have the hardware or even Asterisk knowledge (yet) to do this, I'll be glad to document results in a set of webpages (or maybe we should use one of the already existing sites). Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie question: Meetme (looking for ztrtc)
look at the rtc driver then. you do have a rtc chip already on the system. I looked back in the list and looks like the message that mentioned who wrote ztrtc I deleted. Can someone please let me know where to obtain ztrtc? I did a google on it and came up empty. Thanks, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie question: Meetme (looking for ztrtc)
Seems you used my abreviation. It is really known by zaptelrtc. It seems to be written by Klaus-Peter Junghanns [EMAIL PROTECTED] and is distributed at http://www.junghanns.net/asterisk/. Thanks for the info Steve. I got it but the make didn't work. Will work on it over the weekend. Not trying to stir up old flame wars, and not directed at the person requesting the information above. This was found with a combination of google and grep -ri over my mail directories. Proof positive that a web only based version of this list is not a good option. You are right.. but at least there are archives available. maybe if I had looked for rtc and ASterisk then might have gotten a hit. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The Start extension
I have my sip phones going into the context [from-sip] and would like to play an introduction message and then have the caller enter the extension. The message (dir-info was picked just to have something) doesn't play. Maybe I misunderstood the s extension. According to what I read it is executed everytime something enters the context. Obviously something was misunderstood. The following is in extensions.com: [from-sip] exten= s,1,Answer exten= s,2,Background,dir-intro exten= s,3,DigitTimeout,3 exten= s,4,ResponseTimeout,10 exten = 2000,1,Dial(SIP/2000,20) exten = 2001,1,Dial(SIP/2000,21) Any ideas are appreciated. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Start extension
The s extension is used when there is no known called number. In other words, if you are dialing 2000, the dialplan will always prefer the priority list for 2000 instead of going to 's', so that is why your current system doesn't work. John, Thanks for the details. Actually what I want to do is to play an announcement and then pass the person along to the extension that they dialed. Use of Background was probably not the correct command. (sb. Playback). YOur details clear up the order of processing. Think I can get it from here. Thanks again. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
7 - Ringer volume control 4 - plug in module of user programmable buttons for frequently called numbers. Not everyone would need this so being able to add as an optional module would keep the base phone cost effective. 9 - ability to switch back and forth between speakerphone and handset 7 - message waiting light under the message button. The LCD light blinking is nice but is not easy to see when the room is well lit. 4 - headset jack Thanks for taking the survey. You might also encourage David to have his folks actively participate in the lists. I mentioned it to him before and his reason for not having a more active presence was to avoid the appearance of being commercial on the lists. Personally, I think that it would help to build a better relationship between his technical folks and their userbase. Robert Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 21 Oct 2003, Low, Adam wrote: Maybe I am missing something here but why would it downgrade their network speed to 10mbps, its very rare to find a 100bT switches these days that don't also support 10bT. In a switched ethernet network there would be no performance loss for the other ports !? The cable goes into the phone and then out of the phone into the computer. That switch in the phone is 10Mbit so the computer ends up on 10Mbit too. Perhaps the best way to avoid this is to join all the phones together since they are all 10Mbit anyway, so you will then just need one extra ethernet socket in the room for all the telephones. Michael Michael, How would you be able to connect all phones in a room to one socket? The Ethernet specificiation has a limit to the number of hubs/switches that can be inline. (or at least it used to). The only way I can see to connect all phones to one socket would be to daisy chain them. This would not be a good solution since: - all phones would use the same 10mbps segment, chances for collisions would be high - rules of Ethernet would be violated so even if it did work it may stop at any point with some other normally minor change. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_?_from_mysql.pl files??
Are you manually updating the mySQL tables or do you have a web app. to do that? Robert Steve Creel wrote: You'll want to #include it. This leaves the burden of the [general] and any static configs on sip.conf but allows the script to blindly write out from the database to sip_additional.conf in sip.conf: #include sip_additional.conf Steve Excellent, Thanks for that.. I didn't know there was an include command.. Do you know if include is available in other .conf files eg extensions.conf?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Ouch.. you hit one of my pet peeves.. See below.. This is not meant to be a FLAME but rather SOAPBOX. Robert John Brown (CV) wrote: http == hyper text transport protocol So are the entries on your hard drive with a .htm or .html extension not files? (sorry a little sarcastic I know) *** Big difference beween httProtocol HyperTextMarkupLanguage :-) tftp == trivial FILE trasfer protocol thus using tftp to do updates seems better. Its also a smaller foot print code wise and in boot loader thats important. The boot loader size is the the best argument I have heard so far for using TFTP, but memory is pretty cheap now compared to the days gone by.. :) SOAPBOX Yes, memory is cheap, disk space is pratically free and processors increase in power every year. But that is not a reason to ignore memory usage or write inefficient programs. IF we used the same programming standards as we had in the last century :-) (70s and 80s) then WinXP would probably run on a 486 with 64MB RAM. /SOAPBOX From what I have seen, the Asterisk code must be fairly good. Its running quite nice on my P100, 32 MB system. MusicOnHold ran for 2,5 hours last night without any noticable distortion. Ok.. I don't have many phones hooked up but was fairly surprised that it does as well as it does. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inbound IAXTel failing?
Is anyone else having trouble receiving IAXTel calls? I don't know if it's my config that's broken or IAXTel that broken. Several people have given me their IAXTel numbers and calls to them all fail. I can call FWD numbers via IAXTel just fine. --Eric Eric, I am having a similar problem but am just starting to try and use IAXTEL today for the first time.. Had thought my issue was config but if you had a working config and now its not then maybe its IAXTEL. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH problems
... Still: When I call my Asterisk box (which has a fixed IP and is located within a university network) using X-Lite I get choppy sound to say the least. In fact I can hear only the first half second of what I am supposed to hear followed by permanent silence. Note that this * box has no telephony hardware at all. Any clues or suggestions what else to try? There is no hardware in between that could be responsible for silence suppression, but maybe there is a paramter in Asterisk that I can tune? I tried to use the loud MOH class instead, but it didn't make any difference. :-( Cheers, Philipp When you look in the process list do you see mpg123 processes running. I think there should be 2 for each class you have in the conf file. (at least thats what it seems like on my system. I also copied the mpg123 executable to /usr/bin instead of a link. Not sure if that makes a difference. In theory I would think it would not. With a Grandstream phone and examples from the last days on this list the MOH function when dialed via an extension works absolutly excellent. I've run it for over 2 hours with perfect audio. Gruß aus Friedrichshafen, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gastman crashes on Win32
Can anyone please point me toward the source/binary (linux and Win32) for Gastman?? Robert Hi, The Win32 binary of Gastman crashes on Windows 2000 SP4. Same case on all my machines, no error, no log. Although, the CVS version works great on Linux. Is it anybody who knows how to compile it with mingw32 ? Or better, could anyone, who already has mingw32 installed, make a binary snapshot ? Thanks in advance, Jean-Christophe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI questions..
Jarad, I would be interested in one or 2 of your examples to get an idea of how to get started. Thanks, Robert Friedrichshafen, Germany On Fri, 2003-10-24 at 05:54, WipeOut wrote: First off, can AGI scripts be created using PHP??.. This is where our skills are and since PHP can be run from a command line it would be easier to create and maintain.. Yes, you can use PHP just fine for AGI scripting. I recommend, however, that you use PHP version 4.3.0 or later, due to the updated CLI stuff. Feel free to contact me off-line if you'd like some examples. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS update
Okay, at the CLi i did a show version and it's still showing the old version. What I'm attempting to prevent the overwriting of my already established config files and sound files. Any further suggestions? When I did the make on Asterisk the first (and only) time, I had to do make samples to get the config files. Maybe if you don't make samples you won't get any overwrite of the conf files. (no guarantees here since I've never done a CVS Update) You can always employ the long proven method of making a backup of /etc/asterisk and /var/lib/asterisk/sounds before doing the update. I do a backup of /etc/asterisk fairly often anyway in order to snapshop my config. Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
... At this moment, Asterisk behind a NAT can't connect to an outside SIP provider. If you put asterisk outside your NAT, your inside clients can connect to Asterisk and Asterisk will be able to connect to your providers. I suspected this would be the case. The problem is that I have no control over the NAT. I guess I'll just have to work on my provider a bit more to support IAX. Jonathan, I have the same problem and have solved it by using iaxtel.com. Asterisk talks to IAXtel quite well on inbound and outbound from behind my NAT router. While I don't have the dialplan inside Asterisk completed yet it does do the following: - outbound calls from any internal extension to any service reachable over iaxtel.com. I've tested the following: - USA toll-free numbers (until they stopped working this week.. seems to be an IAXTel problem) - other IAXtel numbers - FWD numbers (1 700 99 x) - inbound calls from FWD to my IAXTEL number ring into the Asterisk box. Currently I play a message then forward them to an internal extension as proof of concept. If you would like the parts of extensions.conf and iax.conf that seem to make it work let me know. I pulled bits and pieces from various places, including a number of the postings on this list over the last 2 days. All of this is rather impressive for me but my wife really wonders if I've lost my sanity... Hunker down everyone.. here comes the solar flare. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nextone softswitch testing and Asterisk long distance
Alexander, I will be happy to help with the testing but since I am behind NAT am not sure it will be of much help to you.. I have 2 Grandstream phones and Asterisk. Robert Friedrichshafen, Germany Hello All, We are looking to test interoperability between Asterisk and Nextone softswitch. Please let me know who is wishing to participate. We will open free US Long distance service for testing. Please email me for more details and to be added to testing participants. To qulaify you need to have already configured Asterisk software any kind of IP Phone , i.e: SIP IP Phone, H323 Phone, PC2Phone, etc/ Thanks for your time. - Alexander You can also contact by ICQ: 2851311 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
My asterisk server(s) are behind NAT, and I am a customer of Vonage (thrice-over), iconnecthere, and Net2Phone. There are still some rough edges (especially with iconnecthere) but overall it is not correct to say that they won't work. B. Thats great to hear. Can you please share your config files that connect iconnecthere and net2phone via SIP? I think there are a number of people here who have tried and not been able to get it to work. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iconnecthere connect problem
I have an Asterisk box behind NAT and am trying to connect to Iconnecthere as was indicated possible earlier. Am getting the following on the Asterisk console: -- Executing Dial(SIP/2001-12c8, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] == No one is available to answer at this time sip.conf is: [delta3] type=peer username= secret= host=213.137.73.140 the extension.conf entry is: exten =_1706NXX,1,Dial,SIP/[EMAIL PROTECTED] Am I missing something?? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
Interesting. Someone thinks that a strategic use for * should be off this list. Someone thought my FAX modem for * should be off this list. However, nobody seems to think a 1000 messages about Grandstream phones should be off this list. Personally I would welcome seeing more of what people are doing in the softswitch area. Regards, Steve Steve, I agree with you. If the discussion involves * then it should be here. In the case of your fax program I think some people who jumped in after the initial introduction might have thought it was totally separate and didn't make the connection. What I find really good about the fax discussion last week was that in the course of 48 hours it went from a non-working integration to functional in Asterisk. There is a tremendous resource base here... If we aren't interested in a discussion then the delete key or mail filters work wonders. Personally I read at least the beginning of all threads... Never know when a new idea or resource is mentioned. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnecthere connect problem
Hello..
Thanks for the reply.. I'll give this a check later today. Is the first
x in the register command your phone number at ICONNECTHERE? I am
using them with the demo account only as outbound so don't have a phone
number. Maybe this could be the problem.
Regards,
Robert
Friedriedrichshafen, Germany
Hi!
try to use in sip.conf :
register =x:[EMAIL PROTECTED]/xx
[iconnect]
type=friend
secret=
username=xxx
host=sipauth.deltathree.com
dtmfmode=inband
context=yourcontext
and in extensions.conf:
exten = _7X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
This works for me
regards
Miklos
- Original Message -
From: rnc Info Lists [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, October 25, 2003 5:17 PM
Subject: [Asterisk-Users] Iconnecthere connect problem
I have an Asterisk box behind NAT and am trying to connect to
Iconnecthere
as was indicated possible earlier. Am getting the following on the
Asterisk console:
-- Executing Dial(SIP/2001-12c8, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
== No one is available to answer at this time
sip.conf is:
[delta3]
type=peer
username=
secret=
host=213.137.73.140
the extension.conf entry is:
exten =_1706NXX,1,Dial,SIP/[EMAIL PROTECTED]
Am I missing something??
Robert
___
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[Asterisk-Users] SwissVoice MGCP IP10S
I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line = aaln/1 The portion of extensions.conf is: exten = 3001,1,Dial(MGCP/aaln1,20) exten = 3001,103,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Citeren rnc Info Lists [EMAIL PROTECTED]: I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Were you able to configure the phones through their webinterface ? You could try entering 'mgcp debug' and then power up your phone to see if it registers at all... Yes, web config. of the phone works ok. The IP for the Asterisk server is in the call agent field and port 2427. The following comes on the Asterisk console at powerup. The items between the repeat. MGCP Show endpoints doesn't show anything. Evidently the phone isn't registered but not sure why since there doesn't seem to be a place to associate a userid or password. MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427MGCP read: RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1529', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart ** from 192.168.0.5:2427MGCP read: RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1529', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, -Original Message- The portion of extensions.conf is: exten = 3001,1,Dial(MGCP/aaln1,20) exten = 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20) Or aaln/1@ip should do just fine. However this doesn't explain why there is no dialtone on the phone.. Oh, one thought: Did you set your toneconfiguration to Europe or US ? If you choose custom you need to configure it another way... Florian Update: I changed the tone config to USA to match Asterisk. No change. I did notice that when I booted up everythign tonight that the MGCP SHOW ENDPOINTS now shows: Gateway 'ip10' at 0.0.0.0 (Dynamic) -- 'aaln/[EMAIL PROTECTED] in 'from-sip' is idle In the messages at start up there is: == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) -- MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate [chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '192.168.0.5' (and thus its endpoint 'aaln/1') does not exist -- Setting hookstate of aaln/[EMAIL PROTECTED] to ONHOOK MGCP DEBUG shows the below lines repeating every couple of seconds: from 192.168.0.5:2427MGCP read: RSIP 1375 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1375', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Still no dialtone and not able to send or receive calls. Evidently there is a problem finding the phone. I can ping it from the Asterisk server so isn't a raw IP issue. On the phone there is the message Waiting for call manager Additional ideas are appreciated. Will keep plugging away at it. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD connection
As far as I know they do only SIP. If your Asterisk box is behind a NAT firewall then you probably will have problems. Hi All, I have a FWD number and wish to connect it to Asterisk to receive my FWD calls. How I do? Is it a register in sip.conf or iax.conf? Regards Dave html xmlns:v=urn:schemas-microsoft-com:vml xmlns:o=urn:schemas-microsoft-com:office:office xmlns:w=urn:schemas-microsoft-com:office:word xmlns=http://www.w3.org/TR/REC-html40; head meta http-equiv=Content-Type content=text/html; charset=us-ascii meta name=ProgId content=Word.Document meta name=Generator content=Microsoft Word 9 meta name=Originator content=Microsoft Word 9 link id=Main-File rel=Main-File href=cid:[EMAIL PROTECTED] !--[if gte mso 9]xml o:shapedefaults v:ext=edit spidmax=2051/ /xml![endif]--!--[if gte mso 9]xml o:shapelayout v:ext=edit o:idmap v:ext=edit data=1/ /o:shapelayout/xml![endif]-- /head body lang=EN-GB link=blue vlink=purple div style='mso-element:header' id=h1 div cite=mid:Unknown20031101T182611378; p class=MsoHeaderfont size=3 color=black face=Times New Romanspan style='font-size:12.0pt;color:black;mso-color-alt:windowtext'!--[if gte vml 1]v:shapetype id=_x_t75 coordsize=21600,21600 o:spt=75 o:preferrelative=t path=[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL PROTECTED]@5xe filled=f stroked=f v:stroke joinstyle=miter/ v:formulas v:f eqn=if lineDrawn pixelLineWidth 0/ v:f eqn=sum @0 1 0/ v:f eqn=sum 0 0 @1/ v:f eqn=prod @2 1 2/ v:f eqn=prod @3 21600 pixelWidth/ v:f eqn=prod @3 21600 pixelHeight/ v:f eqn=sum @0 0 1/ v:f eqn=prod @6 1 2/ v:f eqn=prod @7 21600 pixelWidth/ v:f eqn=sum @8 21600 0/ v:f eqn=prod @7 21600 pixelHeight/ v:f eqn=sum @10 21600 0/ /v:formulas v:path o:extrusionok=f gradientshapeok=t o:connecttype=rect/ o:lock v:ext=edit aspectratio=t/ /v:shapetypev:shape id=_x_s1026 type=#_x_t75 style='position:absolute; margin-left:68.4pt;margin-top:.55pt;width:300pt;height:93pt;z-index:1' o:allowincell=f v:imagedata src=cpclear/ w:wrap type=topAndBottom/ /v:shape![endif]--/span/font/p /div /div div style='mso-element:footer' id=f1 div cite=mid:Unknown20031101T182611378; p class=MsoFooterb style='mso-bidi-font-weight:normal'font size=3 color=navy face=Times New Romanspan style='font-size:12.0pt;color:navy; font-weight:bold'![if !supportEmptyParas]nbsp;![endif]o:p/o:p/span/font/b/p p class=MsoFooter align=center style='text-align:center'b style='mso-bidi-font-weight: normal'font size=3 color=navy face=Times New Romanspan style='font-size: 12.0pt;color:navy;font-weight:bold'Registered Office: - 23 First Street, Low Moor, Bradford, West Yorkshire, BD12 0JQ.o:p/o:p/span/font/b/p p class=MsoFooter align=center style='text-align:center'b style='mso-bidi-font-weight: normal'font size=3 color=navy face=Times New Romanspan style='font-size: 12.0pt;color:navy;font-weight:bold'Company Registration Number: - 03807643.span style=mso-spacerun: yesnbsp; /spanVAT Registration Number: - 734-3363-42o:p/o:p/span/font/b/p p class=MsoFooter align=center style='text-align:center'b style='mso-bidi-font-weight: normal'font size=3 color=navy face=Times New Romanspan style='font-size: 12.0pt;color:navy;font-weight:bold'Telephone / Fax: - 44 (0) 7092 154039. SIP_Phone: - 1 (747)669 1957o:p/o:p/span/font/b/p p class=MsoFooter align=center style='text-align:center'b style='mso-bidi-font-weight: normal'font size=3 color=navy face=Times New Romanspan style='font-size: 12.0pt;color:navy;font-weight:bold'a href=http://www.codepipe.ltd.uk/;http://www.codepipe.ltd.uk/a / a href=http://www.codepipe.com/;http://www.codepipe.com/a / E-Mail: - [EMAIL PROTECTED]/span/font/bb style='mso-bidi-font-weight:normal'font color=navyspan cite= style='color:navy;font-weight:bold'o:p/o:p/span/font/b/p /div /div /body /html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] recording files for menues
Besides, even if I didn't have the files ready, I wouldn't use my lovely voice for it - I'll go to a recording studio with a professional (talking about a production environment) so it's good to know how to do this yourself, in case the studio doesn't know how to record them in this format. For professional recording you can use the same voice as the original prompts.. For details see http://www.digium.com/index.php?menu=thevoice The price seems reasonable to me.. According to John Todd's site the turnaround can be rather fast. (http://www.loligo.com/asterisk/sounds/Sounds-README.txt) http://www.loligo.com/asterisk/ for access to his directory of additional prompts. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi , I even think to avoid using an installer mainly because the installer part is bigger that the application himself. What do you think? Dan, I agree that if an installer or registry entries are not needed then it makes an automated rollout much easier. Also makes it possible to run the program from a diskette/CD so as to be really portable between systems. However, the installer will be necessary for the acceptance by the non-geeks. I only had a short time to run your program last night but it worked well. Configuration was easy and it worked the first time! The problem with changing address book entries was encountered but that has already been reported. Will do more extensive testing tonight with the version from today. Thanks for a good program. Looking forward to it being GPL and the further development. Robert Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: IAX2 Java library (was Re: [Asterisk-Users] New IAX softwarephone (for WIndows platform))
On Mon, 2003-11-03 at 16:27, Alastair Maw wrote: On 03/11/03 20:03, Steven Critchfield wrote: Sounds like you really need a C programmer and get into the guts of asterisk. Can't get more flexible than having the source code yourself to do anything you want. You could add your DSP routines into the dsp.c file and call them when needed. You can also write a asterisk application and have direct access to all the audio in every direction just as you want it. But C isn't as maintainable as nice Java apps, and it's as simple as that. Basically, I'm after the most powerful interface possible to Asterisk, but trying to make it as friendly as possible to code things against. As far as our organization is concerned, that pretty much means Java objects. So you bought that line of Marketecture didn't you. I think there are several large open source projects that prove that C is maintainable. Maintainability is really a function of organization. If you can't be organized, you will not produce very maintainable C code. I'll point out that I am not a C programmer, but making patches to asterisk isn't that difficult. I have also made patches to the kernel without too much hair pulling. -- Steven Critchfield [EMAIL PROTECTED] Steve, You are right... Lots of proof that C is maintainable. I don't profess to be a C, VB or JAVA expert but have programmed for longer than I care to admit. What matters most is good solid and tight code regardless of the language. It all comes down to the number of CPU cycles needed to perform a given function. When doing real time processing, a few cycles here and a few there can add up to make a real difference. Object Oriented is nice for ease of writing/maintaining code but all of those objects have blocks of code behind them. A slight inefficiency there can really impact performance. Sure we have faster processors and lower cost memory every 6 months but thats no excuse for not writing the most efficient code possible. Asterisk does rather well on my Pentium 100/32 MB RAM. Wish I still had the Pentium 75 to try it on. It must really boogy on the bigger boxes. I contend that the most powerful interface is one that meets the requirements of the customer (1st requirement), is written to be the most efficient (2nd requirement) and maintainable (3rd requirement) as possible. The language to be used is the selection of the person doing the development. I'm not a fan of any Microsoft product but they do have a place in the world (for now). Kudos to Dan for his IAX phone. It works. He is responsive to bug fixes. Hopefully he will continue the development. Mark's offer of direct help I think speaks volumes about the importance of GPL IAX softphones for Win32. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
Daniel, the MGCP log you sent shows you sending the digits and asterisk receiving them, however after that either nothing happens (infinite digittimeout) or you cut the log short. Can you also send some console output with 'mgcp no debug' :-) It saves clutter. Maybe a peek at your extensions.conf might be usefull as well ? Also, can you tell us your phone's firmware ? (the IP10) I had one minor issue with the IP10 because of an older firmware version, a simple upgrade resolved it (by the way, in my case it was interpreting digits twice in some cases, i.e. dialling 326 would make asterisk think I was calling 33226) Best regards, Florian FLorian, What version of the IP10 firmware are you using?? I have experienced the multiple digit problem. Seems that this happens when dialing more than 2 digits. My 2 digit extensions seem to work fine but the ones greater than 2 digits get this repeating issue. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP10S and Handset
With some lively IP10S discussions here maybe someone knows about this issue: I can use the speaker phone ok. However the handset and switch hook do not seem to work. If I enable headset then I can get audio via the handset but still have to use the speaker phone button to take ot off hook. Seems a bit wierd.. I have sent it to Swissvoice but no answer back yet. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP10S and Handset
Hi Robert, I haven't the HeadSet model but the lan switch model so I can't be of any help for you. Daniel I have the IP10S LAN Switch model too.. Thats why I find it wierd that the headset setting makes the difference ! Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Demo Weather Report AGI v2.0
Some of you may know me as ManxPower from #Asterisk at irc.freenode.,net I've posted my demp weather report Asterisk AGI script at http://www.fnords.org/~eric/asterisk/downloads/ Eric, Can you comment on the difference in installation ease for Festival and Cepstral? Regards, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Demo Weather Report AGI v2.0
Cepestral was installed and working within 10 mins of my decision to purchase it. It's $30.00 and can be purchased on their web site and they give you a download. They have a demp on their website that will do text-to-speech and give you a .wav file to download and listen to. Download, unpack it, run their install.sh, answer a couple of questions, read the man page and you're done. With Festival I had to figure out exactly which tarballs to download (there was a total of 18 tarballs to download if you count all the Festival voices plus the MBROLA voices), then I had to figure out how to install Festival, then MBROLA, I never have figured out how to actually INSTALL festival, I just run it out of the source directory. It's very picky about paths and such. I'm not a big fan of commercial software. For TTS most of the software either is Windows only or costs several thousand dollars (and sometimes both). If it's a choice between spending two thousand for something like Rhetorical TTS or using Festival, I'll pick Festival. If it's a choice between spending thirty dollars for a TTS system or using Festival, I'll happily spend the $30. Thats a very easy ROI since one hour of a technical resource to setup Festival is easily double the 30 USD. Maybe the Cepestral folks have figured out that making a little money from alot of people will be much better than alot from only a few. I'll buy Cepestral and skip the pizza on Friday night. Net result will be about break even ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail RFC
Earlier today someone posted a RFC number related to voice mail. Unfortunatly I deleted the message so have lost the number and don't see it yet in Google. Can you please resend that to me? Thanks, Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
I think it is time to start commercial Pro version (not expensive !!!) of Asterisk. In my company we already made decision to do it, to offer people ready-to-go solution. But is is hard to do anykind of such product without Digium and Mark's support. Mark I think you are very overloaded with all projects, maybe we can help with Asterisk project. Asterisk Basic will stay as it is now, but we will be developing Asterisk Pro. Correct me if I am wrong, but unless you have a license from Digium directly then you must sell your Pro version software under GPL. What you do for documentation/packaging is probalby not covered under GPL. You make some good points but I think that the solution is not to commercialize everything. There is starting to be a trend of businesses (and governments) turning away from commercialization (ever so slowly but it is in that direction). Pick something that is missing and contribute that to the community. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 1
Is anyone running Asterisk under Fedora Core 1 (http://fedora.redhat.com/)? If so, did everything with Asterisk work properly? I'm looking to migrate from Red Hat 8.0 to Fedora this week. Thanks. Interesting question... Since RedHat will in the future have only their Enterprise version I wonder if Digium/Mark will develop running on Fedora or move to some other system. Guess the jury is still out on what Novell will do with the SUSE distribution long term. Hopefully they maintain the current distribution package scheme. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone-101 MWI
Max, That is what worked for me. if you want the MESSAGE button on the GS to dial the VM then put whatever extension you have defined for VM in the field Voice Mail UserID via the GS Admin Web Interface. Robert Hi Folks, Bit of a newbee here, so please be gentle. :) I'm trying to get the message waiting indication working on a budgetone-101. Is it as simple as putting `mailbox=n' where n is the mailbox number into sip.conf? Is there anything else I should check or set? -Cheers Max. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EU SIP Phone providers
Does anyone know of SIP phone providers (Grandstream in particular) who are located in Germany (or the EU) Thanks for any info. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Background only responds to 1 digit
I have a problem where the Background application only seems to work if one digit is pressed. Extensions with multiple digits just timeout and asterisk hangs up. Below is the relevant excerpt from extensions.conf. In this example, pressing 2 will access the service menu. Then pressing 1 will do the echo test ok but pressing 8463 or 33 will cause an invalid extension message. Any ideas for a solution are appreciated. Robert [default] exten= s,1,ResponseTimeout,10 exten= s,2,Background(rnc-mainmenu) exten= 1,1,Goto(local-extensions,2001,1) exten= 2,1,Goto(services,s,1) [services] exten= s,1,ResponseTimeout,10 exten= s,2,Background(rnc-svcmenu) exten = 1,1,Answer exten = 1,2,Playback(demo-echotest) exten = 1,3,Echo() exten = 1,4,Playback(demo-echodone) exten = 1,5,Wait(1) exten = 1,6,Playback(vm-goodbye) exten = 1,7,Wait(1) exten = 1,8,Hangup exten =33,1,Answer exten =33,2,MusicOnHold(random) exten =8463,1,DateTime() exten =8463,2,Wait(2) exten =8463,3,Playback(vm-goodbye) exten =8463,4,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Business discussion again
Why don't we just add it on the DIgium list server, wouldn't that make more sense, to have a single place for all list memberships? Mark OR even just leave the discussion on asterisk-users... If we create new lists everytime some people disagree with a topic being on-list then we will have not 2 or 3 lists but many more. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnect (DeltaThree) config on Asterisk
Are you also able to make outgoing calls via Iconnecthere? If so do you mind posting your config? I tried their 10 minute trial a couple of months ago but was not able to get a connection. Thanks, Robert I'm receiving calls on my asterisk server from iconnecthere. My asterisk server is behind nat but it still seems to be working fine. AJ On Fri, 21 Nov 2003, Chris HARIGA wrote: Hi, Is anyone using the iconnect on Asterisk to receive and to place calls? Best regards, Chris HARIGA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Netphone SIP phone
Does anyone have experience using the Netphone SIP phone from Ortena Networks (http://www.ortena.com). I contacted them, and they will sell me 10 units for 95 euros/unit. At least i -looks- better then the Grandstream :-) The phone looks interested and appears to have been on the market for a while. The price is good, particularly if they get into the EU without any additional customs taxes... Will be interested to hear of your experiences. Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk European Tour: was RE: [Asterisk-Users] * Party in Paris
Hey, surprise! Just discovered it on the web: http://graphics.cs.uni-sb.de/~rainer/tour.jpg Mark is going on tour! Not sure if this is real info or just a JPG that someone created. Is Stuttgart a definate date on the 30th? If so, where in Stuttgart?? Robert Friedrichshafen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IaxTel seems down
Is anyone other than me having trouble dialing out via IAXTEL? I havn't changed my config files in weeks but seems that IAXTel calls (800 and FWD) stopped working in the past week sometime. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IaxTel seems down
Yes, I've been having problems as well but had not taken the time to diagnose the problem. Just did some looking and it appears iaxtel.com has removed the iax v1 support. iax2 seems to be working fine. Rich, That solved the outbound problem.. Thanks for the hint... 800 numbers are accessable but FWD numbers don't seem to work either me to FWD or FWD to me. It might be that the systems at FWD have the same problem that I had in the ASterisk system. I'll check over on that email list. Thanks again for the help. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] John Brown from Chagres!
On Fri, Dec 12, 2003 at 01:57:02AM -0500, Brian Capouch wrote: John Brown (CV) wrote: Hi List, Just a quick note that we have cleared all back logs of Grandstream product. If you have been awaiting shipment, its shipped. Everyone should be getting tracking numbers shortly. We also have NEW STOCK that can ship within 2 to 3 days of order BT-101 BT-102 HT-286 (YES IN STOCK) Any word on pricing out of the European warehouse? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] John Brown from Chagres!
it's a firmware problem on GS, they are working on that but it seems its not that simple to make volume higher on the speaker and echo go away, anyway 4.26 seems stable for now and with many new features! Miguel, What are the new Features? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Garbled VoiceMail
I tried again at runlevel 3 but to no avail. I'm pretty sure I have sufficient horsepower since I'm running on a box with half gig memory and a speedy CPU. burak I run on a Pentium I /100 Mhz, 32MB RAM with RedHat 9.0 and have no trouble with voicemail audio or Music On Hold. This is a total SIP/IAX(2) machine with no interfaces to the PSTN. Granted this is a much smaller machine than reccomended but it does work. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] new CVS Checkout
Today I deleted the files in the asterisk, libpri, zaptel directories that are in /usr/src and did a new CVS checkout (not update). After doing the make installs and starting asterisk the show version is the same as before: Asterisk CVS-10/09/03-20:33:57 built by [EMAIL PROTECTED] on a i586 running Linux It looks like there are executable asterisk files in /usr/sbin and /usr/lib with a change date of today. I would have expected a newer data on the Version. Is there something I missed doing? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new CVS Checkout
On Sat, 2003-12-13 at 16:41, Joe Dennick wrote: I just updated yesterday, but I did a complete rm -Rf for all of the following directories: /usr/src/zaptel /usr/src/zapata /usr/src/libpri /usr/src/asterisk Then I did a new cvs checkout for all four of those items before recompiling them (make clean; make install) in the same order. My Asterisk now states that its running Version CVS-12/12/03-09:47:51. I use make update. In the Makefile under update: is the instruction rm -f .version -- Dave Cotton [EMAIL PROTECTED] Thanks to everyone who replied to this question. I got rid of the hidden files and reloaded from CVS. It is ok now.Guess over the next days I'll go through the archives and learn a bit more about using CVS to maintain updates. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail Password problems
Hi! I don't get why people always say dtmfmode=info mine works fine with rfc2833. bkw Dunno. I tried rfc2833 first, and had exactly the same problem as described below with voicemail (but only there). Info then worked just fine (as obviously also confirmed by this user here). Is there any other setup/setting that has influence on DTMF detection? Like NAT (yes for me) or anything else? However, more likely it's simply a GS firmware thing (4.17 on mine) - or production (hardware) issue with GS. Incorrect Password '4433211' for user '2000' (context =any) This is a FAQ: use dtmfmode=info in your sip.conf for your Grandstream Note: Don't forget to reload after modifying sip.conf. I am running GS Firmware 1.0.3.78 with Send DTMF = Via SIP INFO sip.conf for that phone is: [2001] type=friend username=2001 secret=test2 host=dynamic context=local-extensions Am able to access VM with no problem and use the phone via *-IAXtel to access other VM systems at USA toll-free numbers. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Quality Survey.... :P
From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Quality Survey :P Reply-To: [EMAIL PROTECTED] ... I have 2 of these phones and they work fine for my application. Granted its not the most intensive use and definatly not the most critical users but... With all of the companies that are running into cash problems in the next year I think that the demands for systems that do everything including make coffee will decrease. Basic functionality will take the place of complicated functionality. Granted GS needs to be more responsive but if they are going to maintain a low price level we need to be a bit understanding about the responses If GS phones don't meet your needs then by all means spend more money on some of the other brands. For some of us, GS does meet the requirements and we will continue to use them. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Quality Survey.... :P
Message: 11 From: Asterisk online forums [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P Date: Wed, 24 Dec 2003 11:23:14 -0500 Reply-To: [EMAIL PROTECTED] Brian, ... We are looking now to improve GS products and start collecting all bugs/probs and send them to GS. Idea is that we are opening Online forums and special Grandstream products mailing list. Some support people from Grandstream will be participating in Forums and Mailing lists, so we will have direct communication between GS and Online community, hopefully it will help us to solve more probs. Grandstream is very interested to make nice product and sell more, so they will be fixing bugs for sure, otherwise they will be out of business. Alexander, I agree with your email but setting up MORE forums and mailing lists is not productive. GS phones have problems interacting with the VoIP services and Asterisk. The BEST places for the GS folks to get feedback AND to interact with the people who are using their phones are on these already existing mailing lists. I don't know why you insist on creating even more websites/email lists for VoIP support. Why not encourage GS to get visible on these lists and interact with their customers here, where they can get the most concentrated feedback (good and bad). Also, a comment for the general list. To me BETA code means that it is NOT yet RELEASED as PRODUCTION code. For anyone to think that Beta code comes without problems is being a bit shortsighted. If you get beta code that works without problems then that is great, otherwise give the developer feedback so that he can fix the bugs and don't complain about the problems it caused you. Otherwise wait on the official production releases. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream 102 flashing display
The phone powers up and I can make calls through my Asterisk gateway to other endpoints. However the four leds under the keypad are permanently illuminated and the backlight slowly flashes on and off. When I pick up the handset there is a repeated tone before I get a dial tone. I know it's trying to tell me something, but the manual does not give anything away. Can't say for the LEDS being illuminated but a flashing backlight and stutter dialtone is the normal message waiting indicator that the phone gives when Asterisk tells it that a mesasge is waiting... I don't remember the exact syntax in sip.conf since am away from my Asterisk box. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD problems
Still, there seems to be a you get what you pay for theme to many of today's posts and this clearly applies to support on FWD. Naybe we should remove the signature from * that enables FWD to identify * systems :-) That certainly seems the case for today's theme... It is certainly the right of any company or person to define the rules of their service. Since I don't pay for either Asterisk or FWD then I appreciate the service that is provided and try not to crusify them when things don't go right. This entire VoIP is still rather experimental. If I want guaranteed service then I'll pay some provider for it... THEN.. and only then will a service level be expected. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vocera Communication Badge
Hi there, yesterday I came across the Vocera Communication Badge and now I'd like to know if anyone here has played with that thing (or even just seen it in real life), and if a price tag can be found for this device? Too bad they don't use SIP... ;-( http://www.vocera.com/ http://www.heise.de/newsticker/data/tol-25.12.03-001/ Cheers, Philipp Looks interesting. I seem to remember something a while back regarding VoiceXML and the TellMe folks. Maybe there is something that is open sourced regarding speach recognition on the server side. An open sourced hardware, as recently discussed, should be able to be packaged so that it could also be a wearable device even if part is on the belt and the speech I/O as separate. Will check with come contacts and see if anyone was involved in the U.S.S. Coronado tests. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I wanna buy a new X100P
I'm trying to buy a new X100P but http://shop.store.yahoo.com/bsdmall/wisifxoin.html is failing to check the order Anybody knows any other way to purchase it? Isamar Try http://store.yahoo.com/asteriskpbx/wildcardx100p.html You won't get the whopping 95 cent discount from BSD Mall but you'll be buying it directly from Digium AND have their support. http://www.digium.com has likes for ordering their hardware. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Happy New Year!!
Where can I find that Howto? I'm new to Asterisk and am looking for all the doc I can find. TIA, Eric Eric, You will find at at: http://members.lycos.co.uk/wipe_out/asterisk/ Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] POTS interfacing recommendation
Check http://www.telappliant.com for their VoIP Starter kits or Telephony Cards sections. Robert -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello there, . for pointing me at a friendly/knowledgeable UK supplier of such cards. Any advice would be greatly appreciated: once I have some known-working hardware in place, I'm cocky enough to believe I can set the software up with enough head banging :) cheers, - -- Matthew Bloch Check http://www.telappliant.com for their VoIP Starter kits or Telephony Cards sections. I don't have any experience with the X100 or voice cards since my implementation is VoIP only (so far). Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Matrix Orbital (usbl LCD or VFD) (oops, wrong list I think)
Sorry 'bout that. -Original Message- From: Kris Edwards [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 3:38 AM To: '[EMAIL PROTECTED]' Subject: Matrix Orbital (usbl LCD or VFD) This probably isn't practical for anyone other than home users, but I would like to use a USB LCD display in my case to display things such as: Answering Caller ID Info Current Context Etc. I am very new to asterisk (in fact, I won't even be getting my digium hardware until the 15th), so I'm sorry if this question isn't up to par with the other discussions going on. Does anyone know of any info on this? If not, is there a particluar file that I can grep out what I need and send to the display? Kris Edwards icq*5661686 Kris, Thats an interesting thought... Since the source code is available you could always modify it to either send the data to the serial port or into a file that you could monitor and then extract what you are looking for. Also the Manager Interface (http://www.voip-info.org/wiki-Asterisk+GUI) might be another source of the data. When I start * I redirect the console log to a file.. That file could be displayed on the LCD as an indication of current activity. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
