Re: [asterisk-users] What is the best softphone work with Asterisk
iaxcomm pro?? On Tue, 24 Jul 2007 19:40:45 -0400, Jim Archer [EMAIL PROTECTED] wrote: I tried several and had very poor luck with each I tried. These included IaxComm, IaxComm Pro, Diax and Firefly II. Also, One other one from I think Germany that had just changed it's name. All of these had issues. I could not get Firefly configured at all to talk to Asterisk. Diax, when the user places a call, just keeps ringing even when the person answered. Both IaxComms would crash. I'm sure there is one out there but I have not found it, although I have not yet tried the SIP soft phones. --On Tuesday, July 24, 2007 2:09 PM -0700 bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal _ ___ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off Topic: Open Source USB Softphone
Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk
On Thu, 06 Jul 2006 19:34:01 -0400, Brian Capouch [EMAIL PROTECTED] wrote: Thomas Kenyon wrote: For some reason when I do this, It only works if I have callerID switched off, otherwise I get authentication errors. Do you know of anyway to bulk-save the contents of all the config screens on that unit? Try NewSipuraUtil at http://www.dualarrow.com If so, I could scrub the passwords and send you the config for the one I'm using. I just checked; I am getting the CallerID just fine when I bring calls into my Asterisk box via the SPA3K. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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/IAX softphones for use in call centre environments
Chris, I've done several customized versions of iaxComm (including two for call centers) Contact me off-list if you're interested. On Thu, 5 Jan 2006 05:37:59 -, Chris Bagnall [EMAIL PROTECTED] wrote: I've been working my way through the softphones listed on voip-info over the last few weeks and I've not really found anything to fit the bill. Has anyone had more luck? The environment is a small call centre of 5 users. Operators often need to be able to transfer calls to other operators with different specialties, so the softphone needs to be easy to use and quick to transfer calls. Operators also have a full-screen web application open most of the time to assist them with callers, so if possible, the softphone needs to either run always on top, or (possibly) have keyboard hotkeys for common functions. Most importantly it needs to work with 96dpi fonts (rather than Windows' default of 72dpi). The TFTs they have are 1280x1024 and operators prefer the larger font size. Many of the softphones I've tried end up with data elements appearing in weird places (or not visibile at all) with the larger font size. Any thoughts / suggestions / pointers? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bluetooth headset with softphone or direct asterisk
On Tue, 15 Nov 2005 16:00:46 -0500, Jerry Geis [EMAIL PROTECTED] wrote: Anyone doing anything with bluetooth headsets? Both DIAX and iaxComm will use the bluetooth headset for audio. I think that DIAX has some additional support for the hook button, but I'm not sure. I'd love to add such a feature to iaxComm, but a cursory google search turns up a flood of bluetooth stacks, and places to buy bluetooth headsets. I'd like to use one with asterisk. If that means through a softphone OR directly connected to asterisk - either way is fine. I found the chan_bluetooth app and downloaded it but it does not compile and have not been successful in reaching the author. I have a kensington USB bluetooth module and a motorala bluetooth headset. Any one have information on how to pair bluetooth modules under linux? THanks Jerry ___ --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 ___ --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] USB phone for Linux?
On Thu, 13 Oct 2005 08:41:17 -0400, Paul [EMAIL PROTECTED] wrote: Tony Mountifield wrote: Hi, Can anyone recommend a USB phone that can be used under Linux, either interfacing directly with Asterisk in some way, or using a soft phone program on Linux that doesn't need screen interaction (only using the phone's keypad)? The idea is to be able to plug it into the USB port of an Asterisk box in a rack, where screen, kbd and mouse may not be available. Thanks in advance! Tony Find me a USB phone with sufficient hardware docs available and I will see what I can do. I could use the same type of thing. I have remote customer servers and would love to have them setup so my contractor tech can just plug in and become extension on my pbx here. Tigerjet makes a USB handset that is based on the same chipset as the S100U. In fact, it looks like there's enough info in the wcfxsusb.[ch] files in zaptel to get the keypad running. I like the AU100 USB phone a lot better, but it looks like it will be windows only. (ironically the AU100 was a lot easier to get working with iaxComm than the Tigerjet phone) What I would do is base the softphone on something like iaxclient. I would have it launched when the usb hotplug was seen. I suppose this could be initially done with 2 devices. One would be a good usb headset and the other would be a keypad with lcd display. ___ --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 ___ --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] telephony that just works
On Mon, 10 Oct 2005 11:01:12 -0300, Ivan Stepaniuk [EMAIL PROTECTED] wrote: On Mon, 2005-10-10 at 13:28 +0200, lenz wrote: I am looking for a way to have multiple remote Windows users download a package and get connected to *. My idea would be that they run a simple app, it connects without any setting to an * box (maybe via IAX) and then people press a button to talk. It would be okay if they had to enter a username and password, but not more than that. i've tried IaxComm http://iaxclient.sourceforge.net/iaxcomm/ it works, it's iax, and it's open source so you can re-package - re-compile it with you own default settings (or even hide those settings you don't want final users to see) Rather than recompile with presets, you'd probably want to change the reg keys used in the installer. When I was first developing iaxComm for family and friends, I distributed the executable with a .reg file with their username/password, the asterisk server , and a few speed dials preset. I finally wrote the installer script for an ITSP that has a really neat approach: The user provides username and password on the web page. The server modifies the username and password in the nsi script, and rebuilds a new installer for each user. -- Ivan Stepaniuk [EMAIL PROTECTED] ___ --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 ___ --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] iaxcomm huge latency
On Wed, 17 Aug 2005 14:34:26 +0200, Juraj Bednar [EMAIL PROTECTED] wrote: Hello, I use iaxcomm-latest from the iaxclient.sf.net page (binary release) on linux, also tried Mac OS X version with the same result and Asterisk 1.0.9 from Debian. Iaxcomm has a huge latency -- tens of seconds, constantly changing over time. It was run on two different machines, always to a SIP phone (which otherwise works correctly even with VoipBuster, which also uses IAX with no latency and other SIP phones). Is it a known bug? What results do you get when the iaxcomms call each other (both via the asterisk server and peer to peer)? There have been some changes to the jitterbuffer code in the iaxclient library since those iaxcomm binaries were posted. You might also want to compile the CVS code. Juraj. ___ 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 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] Alsa and lag
On Thu, 19 May 2005 18:11:47 -0700, Tyler Spivey [EMAIL PROTECTED] wrote: My problem is this. I am connected via fwd's iax protocol. I try to call the echo test, sometimes I get an answer, sometimes not. When I do, sound is about one second lagged - I still hear the ring about a second after I get the answering on console message, and the echo test takes about a second or second and a half to get back the answer. When I use asterisk with dmix and dsnoop, calling just produces the killed. message, and asterisk exits. I'm just looking for a good quality softphone for the console. Linphone doesn't seem to work properly, and is kludgy. Can someone help me fix my alsa setup or the alsa driver? I'm using asterisk CVS. You could probably use the tkphone that comes with iaxclient. ISTR that it runs in the background, and uses a tcl interface to control it. I would think that you could control it from the console. ___ 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: iaxcomm
On Thu, 14 Apr 2005 09:33:21 +0500, amna saleem [EMAIL PROTECTED] wrote: No actually i have successfully installed (from scratch) and been using asterisk for more than 4 months now...i have been using diax phone ...but i came across this iaxcomm just thought about transfering a calljust playing around ..but i can`t really get it working ... maybe i am not getting the one hint can u help thanx Prior to 1.0rc3, you had to hit the OK button in the dialog box to complete the transfer. Enter key did not work. Now anything but Cancel should work. On 4/13/05, amna saleem [EMAIL PROTECTED] wrote: Hi! I was using iaxcomm but due to some reason am not able to transfer calls to some other extensionwhat maybe the problem do i have to make some changes to my extensions.conf??or iax.conf to be able to transfer calls Thanks Amna ___ 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to compile iaxclient with MinGW/Cygwin
On Tue, 01 Feb 2005 10:12:52 +0100, Guan [EMAIL PROTECTED] wrote: Hello, I cant compile iaxclient, because one needs to compile the new version wiax.dll. I tried to compile it under MinGW/Cygwin, but I had the messages like: The instructions for building iaxcomm in iaxclient/simpleclient/iaxcomm/README will work for you. (You will just need steps 1,2 and 4 if you're not compiling iaxcomm, since you won't need wxWindows/wxWidgets) You can use MinGW 3.1.0-1 instead of the older 2.0.0.3 in the README, but don't use 3.2.rc3 if you plan to build the dll rather than the static lib. ___ 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] Soft phones that _actually_ work under Linux?
On Wed, 02 Feb 2005 07:12:54 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Surely there has to be one soft phone that works under Linux. I've tried: kphone - it sometimes complains about the need to release the sound device linphone - lowww iaxcomm - needs some strange widgets What does it ask for that you can't find? various others - either only supplied as binaries, or just plain don't work, or won't compile. Is there just one out there that is guaranteed to work with adequate performance with FC2 or FC3. I don't mind whether its SIP or IAX2 - I just need it to _work_. ___ 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] Soft phones that _actually_ work under Linux?
On Wed, 02 Feb 2005 08:05:13 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: On Wed, 2005-02-02 at 07:41, Michael Van Donselaar wrote: On Wed, 02 Feb 2005 07:12:54 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Surely there has to be one soft phone that works under Linux. I've tried: kphone - it sometimes complains about the need to release the sound device linphone - lowww iaxcomm - needs some strange widgets What does it ask for that you can't find? This is the version: -rw-r--r-- 1 lannet lannet 1392640 Feb 1 06:37 iaxcomm-lin-1.0rc1.tar and this is the error: $ ./iaxcomm Error wxWindows Fatal Error : Couldn't Initialize IAX Client . That means that the iaxclient library couldn't initialize, which is almost always due to a problem with audio initialization. Can you contact me off list with a bit of info about your hardware and OS version? ___ 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] iaxComm version 1.0 released
On Sat, 29 Jan 2005 15:07:48 +0900, Kuniyoshi Murata [EMAIL PROTECTED] wrote: Hi, Is MacOSX version yet to come? I don't have any hardware to compile, so I've been depending on people to send me binaries. I have made a request to the builder of the previous binary. As soon as I get something from someone, I'll post it. Date: Fri, 28 Jan 2005 21:11:48 -0600 [zone:Chicago/Mexico City], [EMAIL PROTECTED] mentioned in msg: [Asterisk-Users] iaxComm version 1.0 released that ... iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems. Recent Changes: * Improved jitterbuffer code * Steve Underwood's Packet Loss Concealment Code Features Include: * iLBC support * GSM support * speex support * ulaw and alaw support * Blind Transfer. * Custom Ringtones per CallerID * Speakerphone mode. * Register with multiple servers (ie enterprise server and iaxtel). * Multiple call appearances. * User selectable audio devices. * User defined ringtones. * Autoanswer intercom calls (with password protection). http://iaxclient.sourceforge.net/iaxcomm-win-1.0rc1.zip http://iaxclient.sourceforge.net/iaxcomm-lin-1.0rc1.tar ___ 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] iaxComm version 1.0 released
On Sat, 29 Jan 2005 10:46:21 -0200, Denis Galvão - iSolve [EMAIL PROTECTED] wrote: Hi Michael. Any work to support some USB Phones!? The ability to dial using the phones keypad!? Not yet, but I'll probably add suport for the TigerJet phone eventually. Thanks. Denis. Em Sáb 29 Jan 2005 01:11, Michael Van Donselaar escreveu: iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems. Recent Changes: * Improved jitterbuffer code * Steve Underwood's Packet Loss Concealment Code Features Include: * iLBC support * GSM support * speex support * ulaw and alaw support * Blind Transfer. * Custom Ringtones per CallerID * Speakerphone mode. * Register with multiple servers (ie enterprise server and iaxtel). * Multiple call appearances. * User selectable audio devices. * User defined ringtones. * Autoanswer intercom calls (with password protection). http://iaxclient.sourceforge.net/iaxcomm-win-1.0rc1.zip http://iaxclient.sourceforge.net/iaxcomm-lin-1.0rc1.tar ___ 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 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] iaxComm version 1.0 released
On Sat, 29 Jan 2005 15:07:48 +0900, Kuniyoshi Murata [EMAIL PROTECTED] wrote: Hi, Is MacOSX version yet to come? Thanks to Andreas Wrede for the binary! http://iaxclient.sourceforge.net/iaxclient/iaxcomm-mac-1.0rc1.zip ___ 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] iaxComm version 1.0 released
iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems. Recent Changes: * Improved jitterbuffer code * Steve Underwood's Packet Loss Concealment Code Features Include: * iLBC support * GSM support * speex support * ulaw and alaw support * Blind Transfer. * Custom Ringtones per CallerID * Speakerphone mode. * Register with multiple servers (ie enterprise server and iaxtel). * Multiple call appearances. * User selectable audio devices. * User defined ringtones. * Autoanswer intercom calls (with password protection). http://iaxclient.sourceforge.net/iaxcomm-win-1.0rc1.zip http://iaxclient.sourceforge.net/iaxcomm-lin-1.0rc1.tar ___ 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] IAX Softphone
On Wed, 26 Jan 2005 16:50:13 +0100, Germán Micale [EMAIL PROTECTED] wrote: Hi, Does someone know an ActiveX IAX softphone? I need a free softphone to connect with Asterisk from a web page. Check here: http://www.geocities.com/babarnazmi/index2.htm Regards ___ 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 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] DIAX 0.9.9g more features and higher stability
On Fri, 14 Jan 2005 15:18:36 +0200, Dan [EMAIL PROTECTED] wrote: snip Is there anything that I can do to solve this issue!? Have you trioed to play with the 'Latency' parameter in Audio Configuration form? Try between 40 and 200. IAXCOM I think use the default which is 200. The iaxclient default latency for windows was changed about two months ago to 40. There were a couple of reports of audio distortion, so it was kicked up to 67. I think you can get pretty agressive with this, just remember to check on the latency if you get distortion. Best regards, Dan ___ 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 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 1/12/2005 ___ 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] DIAX 0.9.9g more features and higher stability
On Fri, 14 Jan 2005 15:36:29 -0200, Denis Galvão - iSolve [EMAIL PROTECTED] wrote: Em Sex 14 Jan 2005 15:11, Michael Van Donselaar escreveu: The iaxclient default latency for windows was changed about two months ago to 40. There were a couple of reports of audio distortion, so it was kicked up to 67. I think you can get pretty agressive with this, just remember to check on the latency if you get distortion. I think the problem is not related to latency. I tried from 20 to 200 latency time, but the problem is the same: Agreed. I was just pointing out for the archives that iaxComm has been using a lower latency as default for over a month (I followed Dan's lead on this). In your first message, you say that you're not having the problem with iaxComm. Which version of iaxComm? DIAX and iaxComm both use the iaxclient library, which has been in flux lately: lots of new features added. If your iaxComm is version 0.99pre4 or later, then I *think* it might be using a more bleeding-edge version of the iaxclient library. Some of the recent library work has been on the jitterbuffer code, which is much more likely to produce the kind of results you describe than portaudio latency tuning. When: Jon - call - Fred Fred listen Jon without problems, but Jon listen Fred with 10 seconds of delay. When: Fred - call - Jon Jon listen Fred wihtout problems, but Fred listen Jon with 10 seconds of delay With Firefly Softphone(IAX2) I dont get this problem, everything works great. Thanks for any help. Denis. ___ 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 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 1/12/2005 ___ 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] iaxComm 0.99pre11 binaries posted to Sourceforge
iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol. It is distributed as part of Steve Kann's iaxclient library. I've just posted new Windows, Linux and Mac OSX binaries to sourceforge. The Windows binary was compiled on WinXP. The Linux binary was compiled on RedHat 9. The OSX binary was compiled by Andreas Wrede on 10.3 and was tested on 10.4 (Tiger) beta. These builds are from iaxclient CVS of 8 JAN 2005. http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-0.99pre11.zip http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-mac-0.99pre11.zip http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-lin-0.99pre11.tar -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 1/12/2005 ___ 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] iaxComm 0.99pre11 binaries posted to Sourceforge
On Sat, 15 Jan 2005 12:41:42 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: On Sat, 2005-01-15 at 12:27, Michael Van Donselaar wrote: iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol. It is distributed as part of Steve Kann's iaxclient library. I've just posted new Windows, Linux and Mac OSX binaries to sourceforge. The Windows binary was compiled on WinXP. The Linux binary was compiled on RedHat 9. ...and when I try to run this on FC2 it complains: # ./iaxcomm Error wxWindows Fatal Error : Couldn't Initialize IAX Client . This means that the iaxclient library couldn't initialize. Most always due to inability to initialize audio, or trying to run on a system that already has asterisk running. WTF is wxWindows? An insidious plot to drive people to www.google.com The OSX binary was compiled by Andreas Wrede on 10.3 and was tested on 10.4 (Tiger) beta. These builds are from iaxclient CVS of 8 JAN 2005. http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-0.99pre11.zip http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-mac-0.99pre11.zip http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-lin-0.99pre11.tar -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 1/12/2005 ___ 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] Recommended IAX softphone.
On Thu, 23 Dec 2004 22:43:43 +0100, Bruno Hertz [EMAIL PROTECTED] wrote: After having been toying around with asterisk and various VoIP stuff for a couple of weeks now, I want to recommend a preferred protocol and softphone to friends and family for calling me up. As SIP and H323 are such a mess to set up in NATed environments, the only reasonable protocol option right now seems to be IAX. After looking at http://www.voip-info.org/wiki-Asterisk+IAX+clients and trying some of those, I'm still not sure which the best client might be. Aside from firefly, I think all IAX2 softphones are based on Steve Kann's iaxclient library: http://iaxclient.sourceforge.net/index.html I've put links to all of the softphones of which I'm aware on the index page. Do you guys have any recommendation, in view of robustness, codec support, general feature richness? If so, I would like to hear ... DIAX and iaxComm are the only iaxclient-based phones right now that support other than GSM codec. The current version of iaxclient allows them to support alaw,ulaw,gsm,iLBC and Speex. iaxComm is Open Source, and currently runs on Win32 and i386Linux platforms. Earlier versions run on Mac OSX, but I don't have hardware to compile it, and have not had any recent reports. Thanks, Bruno. ___ 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 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] iaxComm to iaxComm
On Thu, 18 Nov 2004 17:23:28 -0800, Adam Fineberg [EMAIL PROTECTED] wrote: Having some trouble with segfaults and sound quality all of a sudden (since I recompiled from the latest source) when 2 iaxComm clients connect. First off immediately after the server reports: -- Attempting native bridge of IAX2/[EMAIL PROTECTED]:4569/1 and IAX2/4589/5 The iaxclient library is in flux right now. The echo cancellation code is likely the cause, although I have heard of some problems resolved by disabling speex. I'm going to try to post new linux and windows binaries for iaxcomm this weekend that disable echo cancellation, and prefer iLBC. If you want to try it out, I've just posted a binary based upon 12NOV2004 CVS modified as above. It's not listed on the web page, here's a direct link (not guaranteed past this Sunday): http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-test.exe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Just getting started...
On Fri, 19 Nov 2004 22:48:05 -0500 (EST), Rick Green [EMAIL PROTECTED] wrote: Last night, I attended a presentation on asterix by Greg Boehnlein, and I caught the bug. Today, I've spent the day reading, downloading, and trying to get started. Watch out for the first step, its a doozey! I have no hardware(FXO, FXS ports, VoIP phones) as yet, so I'm trying to move forward with just the commodity stuff I have on hand. Here's my current plan: 1) Install and learn an IAX 'softphone' application. I have some minimal experience with ohphone, which I hope will translate to an IAX softphone. I think that iaxComm is currently the only other iax softphone for linux http://iaxclient.sourceforge.net/iaxcomm/index.html So here I'm stuck. There were no documentation files in the .rpm, nor on the website. The README consists only of We released it, Hooray!. Is this worth pursuing? Is there another IAX softphone application out there? Greg mentioned in his talk a 'firefly IAX stack' but a google search tells me that is a windows app. Not an option. It's much better documented than gnophone: there's a README *and* a QUICKSTART OK, so the documentation isn't that fantastic, but there are some screenshots on the web page. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soft phone auth
On Mon, 1 Nov 2004 17:50:35 -0300, Guido Rebert [EMAIL PROTECTED] wrote:
Do anyone know about soft phones, commercials or not, that asks for
username/passw at launch?
It would be fairly easy to patch iaxComm to do so.
Guido Rebert
Network Manager
GrupoPyD - +54 11 4800
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Guido Rebert
Enviado el: Lunes, 01 de Noviembre de 2004 05:18 p.m.
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: RE: [Asterisk-Users] EIC *
As I suposed... You have there a pretty nice solution.
The T100p has an ethernet port I think your Dialogic´s CIC server board
also has an ethernet port... As mine dialogic is coaxial like...
You have over there a nice stuff!
Any idea with exchange/crm? Asterisk talking with eic/cic...?
Guido Rebert
Network Manager
GrupoPyD - +54 11 4800
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Tom Neville
Enviado el: Lunes, 01 de Noviembre de 2004 01:51 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] EIC *
We're doing just that. :) I was going to reply off-line, but this
should apply to just about any PBX that has PRIs going in and
T1/Channel banks coming out.
We've got a CIC system with 4 PRI ports and 10 Station ports (T1s going
to Adtran TA-750s.) Our current system is built on ISA cards. The CIC
VOIP boards are PCI, so we would have to replace all the hardware.
(Way too expensive.)
What we have working right now is a T1 (normally connected to an Adtra
TA-750) going into a T100P. This is used to bring stations into the
* box. Each channel corresponds to a specific user.. ie
station-cb10-01 in the CIC world is me. Channel 1 is me in the *
world. Using zapata.conf I map each channel to a context..
zapata.conf
---
context = exttomn
channel = 1
context = extdela
channel = 2
Those contexts then ring whatever VOIP lines we want:
extensions.conf
---
[exttomn]
exten = s,1,Dial(SIP/7001SIP/[EMAIL PROTECTED])
[fromtomn]
exten = _NXX,1,Dial(Zap/1/${EXTEN})
exten = _NXXNXX,1,Dial(Zap/1/${EXTEN});
exten = _1NXXNXX,1,Dial(Zap/1/${EXTEN});
exten = _9,1,Dial(Zap/1)
To place outbound calls, calls from the SIP phones are mapped into
[fromUSERNAME] contexts. This causes outbound calls from these
extensions to be placed through the users normal station. (The _9 is
to allow the user to dial 9 to pickup the channel without actually
dialing anything, incase you hit PICKUP in the CIC client by mistake.
:) If the user is not there, the dial to the station is allowed to
just timeout and CIC pulls the call back and drops them in the CIC
voicemail (Unified messaging stuff..) This allows me to stay
Available, no ACD to get my calls bounced to all of my phones...
As for inbound, we ran a PRI into a T100P with PRI_NET signaling.
(Providing a PRI to the Interactive box.) Through that, I am able to
take VOIP inbound calls (eventually routing them from an AS5400 also
providing dialup in other cities) into the CIC box. CallerID is
reported correctly to the CIC box and inbound DID also works as it
should.
I then created a new TrunkGroups in CIC and put the * lines into that.
(Our other lines are also grouped in TrunkGroups.) This allows me to
route outbound long distance out the PRI to the * box and on to
whatever VOIP provider we're using that day. :)
Tom
On Nov 1, 2004, at 10:56 AM, Guido Rebert wrote:
Has anyone done some integration between Interactive Intelligence
(cic,
eic...) and Asterisk?
Thanks all
Guido
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Re: [Asterisk-Users] ATT Cordless VOIP Phone?
On Wed, 27 Oct 2004 13:42:14 -0500, Me [EMAIL PROTECTED] wrote: http://att.onlinephonestore.com/browse/109cea22c2eca7d0b549e0b05cd0c899 Anyone know if this could work with Asterisk? It's not a VOIP phone. It's a combination 1-line POTS phone/900MHz cordless mic and speaker. You plug it into your computer and use it as a cordless headset for your softphone, and/or as a POTS phone. It's what I used to develop iaxComm. Cordless softphone. Unfortunately there's no interface for ringing the handset or detecting hook state, so I had to set up the call at the laptop, but then I could roam. $75 is a bit steep, though. When I bought mine last year, $60 seemed to be street price. Thanks, Todd Routhier Lightwave Technologies, LLC. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iaxComm now supports iLBC,Speex
Linux and Windows compiles of today's CVS are posted on http://iaxclient.sf.net/iaxcomm/index.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Client
On Tue, 31 Aug 2004 14:14:40 -0400, Jon Bebeau [EMAIL PROTECTED] wrote: Hello all, I'm working an a switchboard console for Asterisk and would like to investigate using IAX Client library to Asterisk. I don't seem to be able to find the source. I'm planning on a Win32 app. Guidance on where the source is or who to take to is requested. Jon iaxclient.sf.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxclient compile on win2k
On Mon, 21 Jun 2004 14:24:06 +0530, Navnit Chachan [EMAIL PROTECTED] wrote: Hi, I am not sure whether this is the right forum but anyway am posting my woes. I am trying to compile iaxclient on win2k. The iaxclient-devel list is three doors down on the left. You can subscribe here: http://lists.sourceforge.net/lists/listinfo/iaxclient-devel The list has been low traffic lately, but it is a good place to go for help. Using cygwin,The lib compiles fine but when i try to make any client, i get the following errors /iaxclient_lib.c:424: undefined reference to `__beginthreadex' eg. when i compile testclient, etc. I am using the Makefiles supplied with the distribution. The problem seems to be the process.h that soes not have the __beginthreadex define. If i use mingw/process.h, i get a ton of errors. Thanx Navnit ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip/IAX Clients for Linux
On Tue, 25 May 2004 11:07:12 +0200, [EMAIL PROTECTED] wrote: Hi There, i think all VOIP clients for Linux are unusable! for IAX2: http://iaxclient.sourceforge.net/iaxcomm/index.html I was going to point you to linphone for a SIP phone, but it didn't work for you. I haven't tried it recently, but I know that it did work for me last summer. i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip/IAX Clients for Linux
On Tue, 25 May 2004 16:57:44 +0200, [EMAIL PROTECTED] wrote: Sorry one question again iax2 does it gave debian packages? I don't have a debian system to build it on. You'll have to build from source. If anyone wants to send me a debian package, I'd be happy to post it on the site. On Tue, May 25, 2004 at 09:34:09AM -0500, Michael Van Donselaar wrote: On Tue, 25 May 2004 11:07:12 +0200, [EMAIL PROTECTED] wrote: Hi There, i think all VOIP clients for Linux are unusable! for IAX2: http://iaxclient.sourceforge.net/iaxcomm/index.html I was going to point you to linphone for a SIP phone, but it didn't work for you. I haven't tried it recently, but I know that it did work for me last summer. i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone Audio problem
On Thu, 20 May 2004 14:18:16 +0100, Andy Farnsworth [EMAIL PROTECTED] wrote: As a test, I was trying to use Iaxcomm and Iaxphone to connect to Asterisk and dial out to my other line. Using either of these soft phones, I can connect to Asterisk and listed to audio just fine. I can even connect across the net to another asterisk server and hear audio just fine, however, when I dial out to my second land line the audio that is transmitted is horribly broken up. It is as if the audio stream is broken into 8 parts every second and then every other part is dropped. I then tried the asterisk echo test and got the same thing. I am running Asterisk under RH9 on an AMD 2600+ 512 Mb RAM Desktop and the soft phones on my laptop running Windows XP (Laptop is Sony Vaio PCG-GRT815E, 2.8 Ghz processor, 512Mb Ram). Is this an asterisk problem or a soft phone problem? If asterisk, any ideas on how to fix it? What kind of PSTN interfaces are you using? I'm not sure from the description: are you seeing the problem of both lines, or only the second line? Do you get the same kind of results when using a SIP softphone? I'll be posting new binaries to sourceforge this weekend, because there have been some library changes related to jitter, but I haven't heard or seen anything as drastic as you describe. BTW what version of asterisk? Thanks, Andy Farnsworth ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux IAX client
On Tue, 4 May 2004 12:32:30 -0400, Tim Sailer [EMAIL PROTECTED] wrote: Folks, It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there another IAX2 client that is usable under Linux (Debian preferred)? Thanks, Tim Did it work before you upgraded asterisk, or you can't get it to work at all? I'll admit that the QUICKSTART is a bit terse. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * INSTRUCTIONS
On Fri, 23 Apr 2004 03:55:57 -0400, tmpm [EMAIL PROTECTED] wrote: Might I humbly request someone, somewhere in the community establish a dummies guide to asterisk kind of site, that explains in detail what the cryptic scripts actually do, line by line. The Wiki is helpful, but unless you were in on the movie from the first part, the scene discussions are moot. If you haven't seen the movie yet, the hardest part will be understanding the dial plan. http://www.asteriskdocs.org/stable/docs-html/c511.html gives some insight, but http://www.loligo.com/asterisk/current/ is a working example. I finally figured out what was going on by poring over his extensions.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can i do voice chat without using the hardware
On Mon, 22 Mar 2004 06:31:00 -0800 (PST), suresh kumar [EMAIL PROTECTED] wrote: Hi, PLEASE STOP YELLING INSTALLED ASTERISK PROPERLY AND WORKS FINE. AFTER I MADE A WRONG DECISION TO INSTALL iaxComm in the Client, it's created problem. When i type asterisk -r command, Now i got display as [EMAIL PROTECTED] asterisk]# asterisk -r Asterisk CVS-03/18/04-18:01:45, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-03/18/04-18:01:45 currently running on edventure17 (pid = 4462) edventure17*CLI *is* the command prompt WHY I AM NOT GETTING *CLI prompt? edventure12 i gave when registering the iaxComm. How can i get back *CLI prompt? You *do* have a command prompt. Installing iaxComm did not break anything. When you untar the tar file, you get an executable, two text files called QUICKSTART and README and a 16 bit pcm file called ringtone.raw. No shared libraries. No changes to any .conf files When run, iaxComm creates a lockfile ~/iaxComm.lock to prevent two instances from running at the same time. It also creates a config file of its own, ~/.iaxComm. Neither of these files will conflict with asxterisk in any way. I am using an on board sound card. HOW CAN I CHECK WHETHER IT'S CONFIGURED OR NOT? I'm also assuming that you still have the default extensions.conf. If so, you should be able to type dial 600 YES ... I didn't change the default extensions.conf. If i type dial 600 in edventure17*CLI prompt, getting display as No such command 'dial' (type 'help' for help). I think this is a result of asterisk not recognizing your sound card So i type reload in edventure17*CLI prompt and enter the key. Then i type dial 600 , i got edventure17*CLI prompt. Is there anything wrong with my configuration? If it's wrong, How to fix that? Now what can i do to get back to the asterisk solution part. I am not able to continue ... PLease advice me what can i do in this stage. Thanks Regards, Suresh wrote: snip I would like to get some help from you. My server ip is 192.168.1.1 and i would like to connect to another ip 192.168.1.2. So how can i specify the ip 192.168.1.2 so that make a call from 192.168.1.1? Basic configuration is described in the QUICKSTART that came with the binary. But, since you want to originate calls from the asterisk server, that's a bit different. I'm assuming that you have asterisk installed and working with your sound card. I'm also assuming that you still have the default extensions.conf. If so, you should be able to type dial 600 YES ... I didn't change the default extensions.conf. If i type dial 600 in edventure17*CLI prompt, getting display as No such command 'dial' (type 'help' for help). So i type reload in edventure17*CLI prompt and enter the key. Then i type dial 600 , i got edventure17*CLI prompt. If it's wrong, How to fix that? at the CLI prompt and get the echo test. If not, you'll have to get that fixed before going further. If you can dial extensions from the console OK, then just 1. Make and iax.conf entry for an extension [101] type=friend host=dynamic secret=foo context=default callerid=Remote PC 101 diasallow=all allow=gsm 2. Make an extensions.conf entry for that extension in the default context exten =
Re: [Asterisk-Users] Can i do voice chat without using the hardware
On Mon, 22 Mar 2004 08:00:58 -0800 (PST), suresh kumar [EMAIL PROTECTED]
wrote:
Hi Michael,
Thanks a lot for your help.
I can see the four files when i untar the iaxComm.
How can i get the *CLI prompt?
Just as you have been: asterisk -r
Is it possible to uninstall the asterisk package?
If so, how can i uninstall so that i can configure
asterisk with iaxComm.
You don't need to uninstall either package to get the other to run.
Asterisk and iaxComm can coexist in the same filesystem. You just can't run
them at the same time.
You can have asterisk and iaxComm installed on the same machine. No need to
uninstall either application. They have no files in common.
Thanks Regards,
Suresh
--- Michael Van Donselaar [EMAIL PROTECTED]
wrote:
On Mon, 22 Mar 2004 06:31:00 -0800 (PST), suresh
kumar [EMAIL PROTECTED]
wrote:
Hi,
PLEASE STOP YELLING
INSTALLED ASTERISK PROPERLY AND WORKS FINE. AFTER I
MADE A WRONG DECISION TO INSTALL iaxComm in the
Client, it's created problem. When i type asterisk
-r
command, Now i got display as
[EMAIL PROTECTED] asterisk]# asterisk -r
Asterisk CVS-03/18/04-18:01:45, Copyright (C)
1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-03/18/04-18:01:45
currently
running on edventure17 (pid = 4462)
edventure17*CLI
*is* the command prompt
WHY I AM NOT GETTING *CLI prompt? edventure12 i
gave
when registering the iaxComm. How can i get back
*CLI
prompt?
You *do* have a command prompt.
Installing iaxComm did not break anything. When
you untar the tar file, you
get an executable, two text files called QUICKSTART
and README and a 16 bit pcm
file called ringtone.raw. No shared libraries. No
changes to any .conf files
When run, iaxComm creates a lockfile ~/iaxComm.lock
to prevent two instances
from running at the same time. It also creates a
config file of its own,
~/.iaxComm. Neither of these files will conflict
with asxterisk in any way.
I am using an on board sound card. HOW CAN I CHECK
WHETHER IT'S CONFIGURED OR NOT?
I'm also assuming that you still have the default
extensions.conf. If
so, you should be able to type dial 600
YES ... I didn't change the default
extensions.conf.
If i type dial 600 in edventure17*CLI prompt,
getting
display as No such command 'dial' (type 'help'
for
help).
I think this is a result of asterisk not recognizing
your sound card
So i type reload in edventure17*CLI prompt and
enter
the key. Then i type dial 600 , i got
edventure17*CLI
prompt.
Is there anything wrong with my configuration?
If it's wrong, How to fix that?
Now what can i do to get back to the asterisk
solution
part. I am not able to continue ...
PLease advice me what can i do in this stage.
Thanks Regards,
Suresh
wrote:
snip
I would like to get some help from you.
My server ip is 192.168.1.1 and i would like to
connect to another ip 192.168.1.2. So how can i
specify the ip 192.168.1.2 so that make a call
from
192.168.1.1?
Basic configuration is described in the QUICKSTART
that came with the binary.
But, since you want to originate calls from the
asterisk server, that's a bit different.
I'm assuming that you have asterisk installed and
working with your sound card.
I'm also assuming that you still have the default
extensions.conf. If so, you should be able to
type
dial 600
YES ... I didn't change the default
extensions.conf.
If i type dial 600 in edventure17*CLI prompt,
getting
display as No such command 'dial' (type 'help'
for
help).
So i type reload in edventure17*CLI prompt and
enter
the key. Then i type dial 600 , i got
edventure17*CLI
prompt.
If it's wrong, How to fix that?
at the CLI prompt and get the echo test. If not,
you'll have to get that fixed before going
further
Re: [Asterisk-Users] Can i do voice chat without using the hardware
On Sun, 21 Mar 2004 04:00:39 -0800 (PST), suresh kumar [EMAIL PROTECTED] wrote: Hi, Thanks a lot for your help. After installing iaxComm, When I test Asterisk typing # asterisk c Are you running iaxComm on the same machine as asterisk? You can't do that. I got a display like this (Not getting any CLI prompt) [chan_iax.so] = (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory ) Why i am getting this error? How can i tackle this error? Before installing the iaxComm, i will get the CLI prompt. Now it's not getting it. So please help me to solve this problem. Thanks Regards, Sur --- Michael Van Donselaar [EMAIL PROTECTED] wrote: On Fri, 19 Mar 2004 05:53:44 -0800 (PST), suresh kumar [EMAIL PROTECTED] wrote: Hi, Thanks for your help. I had gone through the www.voip-info.org and got more information regarding the asterisk. Still now i am not clear, how can i test this software. I had gone through the mailarchieves, but didn't get any solution. My aim is that, i want to connect my PC (where i installed the asterisk) to another PC in my network for voice chating. For this purpose, what are the steps to be done? which are the files to be modified. I would like to make use of the existing Hardware (sound card, network card etc), i am not using any extra hardware. Is X-Lite work in Linux? or any compatible s/w that works under linux? iaxComm uses asterisk's native IAX protocol. It runs on Windows, Linux and OSX. Precompiled binaries for RedHat 9, Windows, and OSX (Panther) ara available at: http://iaxclient.sourceforge.net/iaxcomm/index.html linphone is a SIP softphone for Linux: http://www.linphone.org I am expecting an help from experienced person like you. Or can you please send me the link where i can get more information to tackle my problem. Thanking you, Best Regards, Sur --- Matt Ammerman [EMAIL PROTECTED] wrote: Sure thing. You're going to have to get SIP involved though. This means using sip.conf to create new sip users. Do a search on www.voip-info.org for sip.conf and it will explain how to configure a user for SIP. Then you'll need SIP clients (hard VoIP phones, or SIP soft clients such as Windows Messenger or X-Lite). You can make VoIP calls over an existing network infrastructure without analog hardware. For instance, I have an internal Asterisk PBX allowing VoIP conversations between X-Lite, Windows Messenger, and Pingtel clients - all over networking connections, no T1/E1/Analog needed. You need the hardware when you start interfacing with the PSTN for the most part. __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can i do voice chat without using the hardware
On Sun, 21 Mar 2004 07:38:17 -0800 (PST), suresh kumar [EMAIL PROTECTED] wrote: Hi, Yes.. i installed iaxComm in the same machine. Hope that was a wrong method. How can i uninstall iaxComm so that i can get the CLI prompt? Please help me to provide a solution for this. Thanks Regards, Sur You don't need to. From looking at another of your posts, it looks like you've got asterisk running in the background. Typing asterisk -r should get you the CLI of the asterisk that is running in the backgound. Michael Van Donselaar [EMAIL PROTECTED] wrote: On Sun, 21 Mar 2004 04:00:39 -0800 (PST), suresh kumar wrote: Hi, Thanks a lot for your help. After installing iaxComm, When I test Asterisk typing # asterisk c Are you running iaxComm on the same machine as asterisk? You can't do that. I got a display like this (Not getting any CLI prompt) [chan_iax.so] = (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory ) Why i am getting this error? How can i tackle this error? Before installing the iaxComm, i will get the CLI prompt. Now it's not getting it. So please help me to solve this problem. Thanks Regards, Sur --- Michael Van Donselaar wrote: On Fri, 19 Mar 2004 05:53:44 -0800 (PST), suresh kumar wrote: Hi, Thanks for your help. I had gone through the www.voip-info.org and got more information regarding the asterisk. Still now i am not clear, how can i test this software. I had gone through the mailarchieves, but didn't get any solution. My aim is that, i want to connect my PC (where i installed the asterisk) to another PC in my network for voice chating. For this purpose, what are the steps to be done? which are the files to be modified. I would like to make use of the existing Hardware (sound card, network card etc), i am not using any extra hardware. Is X-Lite work in Linux? or any compatible s/w that works under linux? iaxComm uses asterisk's native IAX protocol. It runs on Windows, Linux and OSX. Precompiled binaries for RedHat 9, Windows, and OSX (Panther) ara available at: http://iaxclient.sourceforge.net/iaxcomm/index.html linphone is a SIP softphone for Linux: http://www.linphone.org I am expecting an help from experienced person like you. Or can you please send me the link where i can get more information to tackle my problem. Thanking you, Best Regards, Sur --- Matt Ammerman wrote: Sure thing. You're going to have to get SIP involved though. This means using sip.conf to create new sip users. Do a search on www.voip-info.org for sip.conf and it will explain how to configure a user for SIP. Then you'll need SIP clients (hard VoIP phones, or SIP soft clients such as Windows Messenger or X-Lite). You can make VoIP calls over an existing network infrastructure without analog hardware. For instance, I have an internal Asterisk PBX allowing VoIP conversations between X-Lite, Windows Messenger, and Pingtel clients - all over networking connections, no T1/E1/Analog needed. You need the hardware when you start interfacing with the PSTN for the most part. __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can i do voice chat without using the hardware
On Sun, 21 Mar 2004 09:11:33 -0800 (PST), suresh kumar [EMAIL PROTECTED] wrote: snip I would like to get some help from you. My server ip is 192.168.1.1 and i would like to connect to another ip 192.168.1.2. So how can i specify the ip 192.168.1.2 so that make a call from 192.168.1.1? Basic configuration is described in the QUICKSTART that came with the binary. But, since you want to originate calls from the asterisk server, that's a bit different. I'm assuming that you have asterisk installed and working with your sound card. I'm also assuming that you still have the default extensions.conf. If so, you should be able to type dial 600 at the CLI prompt and get the echo test. If not, you'll have to get that fixed before going further. If you can dial extensions from the console OK, then just 1. Make and iax.conf entry for an extension [101] type=friend host=dynamic secret=foo context=default callerid=Remote PC 101 diasallow=all allow=gsm 2. Make an extensions.conf entry for that extension in the default context exten = 101,1,Dial(IAX2/101) 3. Configure iaxComm on the other machine to use the iaxconf entry (username 101, password foo) as described in the QUICKSTART. Should i install softphone s/w in server (192.168.1.1) and other machine (192.168.1.2)? You don't want iaxComm installed on the asterisk server. Just on the remote machines. In sip.conf file how can i specify the ip 102.168.1.2 iaxComm does not use the SIP protocol. It's config file is iax.conf If you have time, please help me to get a solution. Thanks Regards, Suresh --- Girish Gopinath [EMAIL PROTECTED] wrote: Suresh, From: suresh kumar [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can i do voice chat without using the hardware Date: Fri, 19 Mar 2004 05:50:00 -0800 (PST) Thanks a lot for your valuable information. I will go through it once again. Still i don't have any idea to connect two PC's. Hope i may get help from you. For configuring 2 softphones with Asterisk see this link: http://www.automated.it/guidetoasterisk.htm That helped me a lot in learning Asterisk. It explains configuring your sip phones with Asterisk. Is there any softwares like X-lite for Linux? Yes, I think you can use linophone. But i was not able to install linophone because of some make issues. Also i have tested the softphone from zultys. It works well with Asterisk. You can get it from their web site:http://www.zultys.com Regards, Girish _ Catch the formula fever! Get all the latest news. http://www.msn.co.in/formula2004/ Right here on MSN. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can i do voice chat without using the hardware
On Fri, 19 Mar 2004 05:53:44 -0800 (PST), suresh kumar [EMAIL PROTECTED] wrote: Hi, Thanks for your help. I had gone through the www.voip-info.org and got more information regarding the asterisk. Still now i am not clear, how can i test this software. I had gone through the mailarchieves, but didn't get any solution. My aim is that, i want to connect my PC (where i installed the asterisk) to another PC in my network for voice chating. For this purpose, what are the steps to be done? which are the files to be modified. I would like to make use of the existing Hardware (sound card, network card etc), i am not using any extra hardware. Is X-Lite work in Linux? or any compatible s/w that works under linux? iaxComm uses asterisk's native IAX protocol. It runs on Windows, Linux and OSX. Precompiled binaries for RedHat 9, Windows, and OSX (Panther) ara available at: http://iaxclient.sourceforge.net/iaxcomm/index.html linphone is a SIP softphone for Linux: http://www.linphone.org I am expecting an help from experienced person like you. Or can you please send me the link where i can get more information to tackle my problem. Thanking you, Best Regards, Sur --- Matt Ammerman [EMAIL PROTECTED] wrote: Sure thing. You're going to have to get SIP involved though. This means using sip.conf to create new sip users. Do a search on www.voip-info.org for sip.conf and it will explain how to configure a user for SIP. Then you'll need SIP clients (hard VoIP phones, or SIP soft clients such as Windows Messenger or X-Lite). You can make VoIP calls over an existing network infrastructure without analog hardware. For instance, I have an internal Asterisk PBX allowing VoIP conversations between X-Lite, Windows Messenger, and Pingtel clients - all over networking connections, no T1/E1/Analog needed. You need the hardware when you start interfacing with the PSTN for the most part. __ Do you Yahoo!? Yahoo! Mail - More reliable, more storage, less spam http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX Error
On Tue, 9 Mar 2004 12:35:40 +1300, Matt Riddell [EMAIL PROTECTED] wrote: Also, If I click a phonebook entry then dial it crashes... what are the contents of the phone book entry? The reason is that it goes to 2 for line and that doesn't exist ? If I click VM then 1 then dial, it's ok Matt P.S. is there somewhere I sould post this instead of here? Steve Sokol's IAX Phone is based upon iaxclient. Dan's DIAX is based upon iaxclient. iaxComm is based upon iaxclient. I think that [EMAIL PROTECTED] would be appropriate for any bug reports. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.614 / Virus Database: 393 - Release Date: 3/5/04 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VTGO-PG and IPP200
On Tue, 2 Mar 2004 13:22:05 -0500, Tim Sailer [EMAIL PROTECTED] wrote: On Tue, Mar 02, 2004 at 09:59:55AM -0800, Ed Rubright wrote: Hmmm. I thought it was just that I didn't have X-Lite and Asterisk configured correctly and I've been searching thru docs trying to figure out how to get a MWI working! Does X-Pro have a MWI? Not that I can find from the manual or web page. I have MWI working with the BT-100 hardphone, but using the same SIP setup for the extension, I get nothing from X-Lite. I'm looking for a softphone (SIP I guess. I don't see MWI indicated for IAX) that will support MWI and Hold, nothing fancy. $50-100 is about the range I'll look for, because the hardphone are more cost effective at that point. IAX Phone (Hi, Steve!) has a MWI indicator http://www.sokol-associates.com/IaxPhone.htm Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm updates at sourceforge
On Mon, 01 Mar 2004 07:34:33 -1000, Jean-Denis Girard [EMAIL PROTECTED] wrote: Download link at top of page is broken for linux iaxcomm-lin-20040228.zip should be iaxcomm-lin-20040228.tar Thanks. Fixed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iaxComm updates at sourceforge
There are new iaxComm binaries for Windows, Linux and Mac OSX posted at http://iaxclient.sourceforge.net/iaxcomm/index.html These binaries also have the recent library change that allows client to client connections to be handed off correctly. Recent changes include speakerphone mode, blind transfer, music on hold and custom ringtones based upon callerid. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USB Phones
On Fri, 27 Feb 2004 10:52:29 -0500, Tim Sailer [EMAIL PROTECTED] wrote: I have some mobile users that would prefer to have a 'real phone' instead of a computer headset. I've been looking around at the USB phone setups, which is (it seems) simply a softphone with a USB handset. The only ones I've found seen to be locked to a particular service provider. Has anyone used these, and are there any ones that can work as a general softphone, like X-Lite? Tim I the TigerJet phone does emulate a soundcard. I use it with iaxComm. Steve Sokol's IAX Phone supports the Eutectics handset. I have used the S100U with iaxcomm, as well. I am working on off hook detection and handset ringing for the TigerJet handset, but for now it works great as a soundcard/headset substitute. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web based UA
On Wed, 25 Feb 2004 11:58:54 -0700, [EMAIL PROTECTED] wrote: You may be right here. I was thinking of an ActiveX plug-in. I don't expect them to use public internet kiosks so they should be able to use the ActiveX approach. I was hoping that something IAX based could be found as it would make the connectivity easier and open port risk reduced. Michael None of the IAX softphones are very large. iaxComm is only 600K. Original Message Subject: Re: [Asterisk-Users] Web based UA From: Jonathan Moore [EMAIL PROTECTED] Date: Wed, February 25, 2004 11:16 am To: [EMAIL PROTECTED] I think xten is supposed to have an active X control version of their softphone that would probably do what you are talking about. On Wed, 25 Feb 2004, Michael Graves wrote: Hello All, Does anyone here have any experience with web based soft clients for *? I'm thinking about putting a page up on our corp web server that would let staff in the field connect to our in-house phone system via the internet. This could help staff making overseas calls while on trips, without demanding that they use a particular laptop/soft phone. They could use an PC on a broadband connection. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] It is dangerous to be correct about matters when the established authories are wrong. - Voltaire ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jonathan Moore Technology Coordinator Winfield Public Schools Office 316-221-5100 Fax 316-221-0508 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Iaxclient-devel] New Windows IAX Client
On Wed, 21 Jan 2004 18:01:38 -0600, Steven Sokol [EMAIL PROTECTED] wrote: Announcing a new Windows-based IAX/IAX2 client. Please download it and give it a try. Let me know about any bugs, and any missing features. I have yet to come up with a catchy name for it, so at this point it calls itself IAX Phone. (Suggestions? Non-derogatory suggestions, preferably). (There was a program out there at one point called iaxphone. I think it's defunct now.) Steve, this looks absolutely fantastic. Is there any possibility of a linux port? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Bug in iaxComm Solved
I've applied Steve Sokol's patch the the IAXClient source, and the IAX2 noanswer bug is solved in iaxComm, as well. Win32 and Linux binaries are available at http://iaxclient.sourceforge.net/iaxcomm/index.html Any feedback is appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kedpad less extension
On Thu, 8 Jan 2004 13:58:45 -0700, [EMAIL PROTECTED] wrote: Does anyone know of a resource for extensions in which the server (whether asterisk or custom scripts) can trigger the phone to be answered? So for example an operator can have a headset and when a call comes through the call is automatically (through a script) connected to the headset instead of the operator having to manually answer the call. Any responses, help or ideas of a type of supplier to contact for more information would be greatly appreciated. I don't know if you are wanting a hard phone or a softphone. If a softphone is acceptable, iaxComm can do this on a Windows or Linux PC with the intercom feature. Thanks Michael Blood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kedpad less extension
On Thu, 8 Jan 2004 18:48:35 -0700, [EMAIL PROTECTED] wrote: I was looking for a hardphone since it made sense to me that it would be better quality. But I like the scalability of this option since it is an IP Phone. I think that you'll be surprised with the audio quality. Especially after Steve put in the noise reduction code! Also may there is another option here. Is there any way that an server could have multiple instances of an iax type client and then have multiple headsets attached to each? I don't think that the audio library works and plays well with others. Thanks again. Michael Blood -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Van Donselaar Sent: Thursday, January 08, 2004 4:01 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Kedpad less extension On Thu, 8 Jan 2004 13:58:45 -0700, [EMAIL PROTECTED] wrote: Does anyone know of a resource for extensions in which the server (whether asterisk or custom scripts) can trigger the phone to be answered? So for example an operator can have a headset and when a call comes through the call is automatically (through a script) connected to the headset instead of the operator having to manually answer the call. Any responses, help or ideas of a type of supplier to contact for more information would be greatly appreciated. I don't know if you are wanting a hard phone or a softphone. If a softphone is acceptable, iaxComm can do this on a Windows or Linux PC with the intercom feature. Thanks Michael Blood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one way choppy sound problem !
On Mon, 5 Jan 2004 13:29:06 +0100, Dawid Mielnik [EMAIL PROTECTED] wrote: Hi Again, Apart from X-lite client I have also tried eStara, diax phone, iaxcomm and some others. I have tried different codecs - GSM, aLAW uLAW. They all give the same result. In the direction PSTN user --- Softphone user the sound is crystal clear (also tried on a dial-up connection), in the other direction however the sound is a bit choppy. The chops occur at regular intervals of time, at about 1-2 seconds !? Are the PSTN interface and a network card sharing an interrupt? I had similar problems with my X100P and a thunderlan dual ethernet card shring IRQs (also would make one of the ethernet ports fails until reboot) Are you still using the P133? I tried using a P120, but it wouldn't do the trick with GSM conversion. DIAX and iaxComm, since they use the iaxclient library, need to use GSM. When analyzing *'s ethernet interface with tcpdump (raw tcpdump -i eth0) I have noticed that the scrolling slows down during the times when chops occur in the sound. I have tested things using different softphones and different internet connections (user side) - always yelding the same result. In other words this is probably a problem on asterisk, either the hardware (ehternet interface/E100p) or a swoftware bug, incoming RTP buffering maybe ? Has anyone actually obtained a good quality sound in a similar setup ? Internet 2 x E1 x-lite --- Asterisk --- PSTN Any help appreciated ! Best regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nicolas Gudino Sent: Friday, January 02, 2004 6:35 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] one way choppy sound problem ! I have a similar problem, with GS phones, X-Lite or Kphone. I tried all the codecs with the same result. Choppy sound in the direction SIP-Phone - pstn, but crystal clear sound the other way around. The only difference in my case is that I have two asterisks servers connected together via IAX2, the PSTN call is received in one asterisk, while the sip phones are in the other asterisk. Ex: pstn - * --iax2-- * -sip phone (GS, Xlite or Kphone) If I use an Xlite in the same asterisk as the pstn line, the sound is perfect in both ways. But when I answer the call in the second asterisk, the sound from the sip phone to pstn is choppy, with or without silence detection, and the sound from pstn to sip phone is perfect. The asterisk server with the pstn line is an old pentium 133, maybe thats the problem, I will try with a better machine and see how it goes. On Fri, 2004-01-02 at 06:23, Dawid Mielnik wrote: Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite --- Asterisk --- PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite - PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data were lost. The strange thing is that the x-lite user hears the PSTN user fine ! In x-lite, I have swithed off sience detection (transmit silence - yes), this has improved the sound quality but did not eliminated the problem. I have fed a countinious sound into the microphone and still got chops in the sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get the same problem with all of them. Maybe the problem lies somewhere in audio buffering settings on x-lite ? Has anyone ever had this sort of problem and managed to deal with it ? I would greatly appreciate your help ! Best regards, Dave ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring DG-104s
On Thu, 18 Dec 2003 14:10:08 +0200, Anton Yurchenko [EMAIL PROTECTED] wrote: Hello, I asked on the asterisk mailing list about dlink DG-104SH, some people wrote that they had DG-104S working, so I kicked that 104SH , and got an 104S. And now I`m having trouble configuring it( I`m kinda new to MGCP) What do I put in the Config Call agent IP section? I now have it like this: Notify Entity RGW Name DNS IP . . . DNS State SDP IP address for NAT . . . Make sure that Notify Entity is [EMAIL PROTECTED]:2427 (172.20.0.50 is your asterisk box, right? Make RGW Name DG104S and in the mgcp.conf i have: [general] port = 2427 bindaddr = 0.0.0.0 [172.20.0.98] Make this [DG104S] Let me know if this helps ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxclients missing calls
On Mon, 15 Dec 2003 23:05:56 -0500 (EST), [EMAIL PROTECTED] wrote: Hello All When I open up iaxcomm, it registers fine with the asterisk server. If I call into it, iaxcomm will ring; however if I leave iaxcomm sitting idle for awhile (I haven't figured out exactly how long) it seems to miss calls. I can see the calls coming in on the asterisk server but they never ring through on iaxcomm. If I close it and reopen it, it takes calls again fine. I thought I saw someone else who had experienced this problem too but I don't recall seeing a solution. Any thoughts or solutions? I'm trying to use iaxcomm in my every daybusiness but it's kind of hard when I miss calls. It complicates things a bit. Although I used iaxcomm as an example here I've experienced the same problem with DIAX. Thanks a bunch. AJ Hi, AJ. It sounds like this is related to the other incoming ring issue. I'll check through the archives to verify, but I think a couple of other users were seeing this even soon after launching iaxComm or DIAX. I'm wondering if the iaxclient lib might be having a problem with registration renewal, or if it's a problem with event dispatching. Any other users seen this yet? (That is, seen incoming ring failures become more likely as uptime increases). The next time you get an ignored incoming call, could you try doing an iax2 debug on the cli? Do you see the same problem with DIAX when using IAX instead of IAX2? ___ 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] Dlink DG-104SH
I have the MGCP-only version, the DG-104S Works great for me. mgcp.conf: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.0.254 [DG104S] host = 192.168.0.130 threewaycalling = yes transfer = yes callwaiting = yes context = from-internal callgroup = 1 pickupgroup = 1 canreinvite=no callerid = General 301 mailbox = michael line = aaln/1 callerid = Michael 302 mailbox = michael line = aaln/2 callerid = Alex 303 mailbox = alex line = aaln/3 callerid = Katie 304 mailbox = katie line = aaln/4 On Fri, 12 Dec 2003 17:59:54 +0200, Anton Yurchenko [EMAIL PROTECTED] wrote: Hello, Anybody has it working with asterisk? Could you share your experience ( good/bad) Thank you ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm, etc.)
On Tue, 2 Dec 2003 12:14:24 -0600, Steven Sokol [EMAIL PROTECTED] wrote: Hi, I seem to be having problems with IAX clients based on the iaxClient library. I have been working on my own client (an augmentation to the Call Manager I released last week) and it seems to regularly miss incoming calls entirely. It also occasionally misses the drop signal when the remote end drops a call. Has anybody else seen this kind of behavior? I have tested with my client, with DIAX and with iaxComm and all three act the same way. I would really like to know if it has something to do with my asterisk setup, or if it is specific to the client library I am using. I am seeing the same behaviour ocassionally, but had attributed it to something wrong in my application code, rather than the library. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm Update available [Ringtones, Intercom, UI improvements]
On Mon, 01 Dec 2003 00:46:22 +0100, Brancaleoni Matteo [EMAIL PROTECTED] wrote: Hi. Isn't possible to have a statically linked version for linux? [EMAIL PROTECTED] iaxcomm]$ ./iaxcomm ./iaxcomm: error while loading shared libraries: libwx_gtk_xrc-2.4.so: cannot open shared object file: No such file or directory [EMAIL PROTECTED] iaxcomm]$ I replaced iaxcomm-lin-20031129.tar.gz with a new file (same name) that now has libwx-gtk-xrc-2.4.a linked in. Please let me know how it works for you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm Update available [Ringtones, Intercom, UI improvements]
On Sun, 30 Nov 2003 19:09:00 -0500 (EST), [EMAIL PROTECTED] wrote: Steve Kahn and I were having this very discussion the other day on the iaxclient-devel list. I know that Steve is now aware of it and I believe he's going to pass the the same suggestion to Mike VanDoselaar. I can't speak for either Steve or Mike but I think you will probably be seeing it in the future. AJ I'll fix the Makefile and recompile Monday. The box I was using for linux builds is down right now (long story). On Mon, 1 Dec 2003, Brancaleoni Matteo wrote: Hi. Isn't possible to have a statically linked version for linux? [EMAIL PROTECTED] iaxcomm]$ ./iaxcomm ./iaxcomm: error while loading shared libraries: libwx_gtk_xrc-2.4.so: cannot open shared object file: No such file or directory [EMAIL PROTECTED] iaxcomm]$ :( isn't very useful under linux if on every box I must install the wxwindows gtk libs ... Matteo Il sab, 2003-11-29 alle 23:11, Michael Van Donselaar ha scritto: iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X systems. Sources included in the iaxclient library: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Precompiled binaries at: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Features: * Register with multiple servers (ie enterprise server and iaxtel). * Multiple call appearances. * User selectable audio devices. * User defined ringtones. * Autoanswer intercom calls (with password protection). * Registration with an asterisk server prior to dialing is no longer required. * Simplified Directory dialogs. * Incoming callers automatically added to phone book. Recent changes include: * Registration with an asterisk server prior to dialing is no longer required. * Simplified Directory dialogs. * No longer crashes on unexpected screen sizes. ___ 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
[Asterisk-Users] iaxComm Update available [Ringtones, Intercom, UI improvements]
iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X systems. Sources included in the iaxclient library: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Precompiled binaries at: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Features: * Register with multiple servers (ie enterprise server and iaxtel). * Multiple call appearances. * User selectable audio devices. * User defined ringtones. * Autoanswer intercom calls (with password protection). * Registration with an asterisk server prior to dialing is no longer required. * Simplified Directory dialogs. * Incoming callers automatically added to phone book. Recent changes include: * Registration with an asterisk server prior to dialing is no longer required. * Simplified Directory dialogs. * No longer crashes on unexpected screen sizes. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 without iaxtel.com
Hi, Ricky On Sat, 22 Nov 2003 03:15:27 -0800, Asterisk [EMAIL PROTECTED] wrote: Greetings everyone. Could anyone tell me how to setup an IAX call using iaxcomm from a remote (PC) user without going throug iaxtel.com? If you want to call PC-toPC, just type 192.168.0.1/s just above the Dial key. No need to register with iaxtel.com I would like users to register to my server directly instead of looking up in iaxtel directory. Please provide an example of iax.conf commands and extensions.conf. My laptop registers with my home asterisk server, vangate, as extension 309 extensions.conf: exten = 309,1,Dial(IAX2/309) iax.conf: [309] type=friend host=dynamic secret=oops_I_forgot _to_change_this context=from-iax callerid=Michael PC 309 Hope this helps. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm new version installation problem
On Thu, 20 Nov 2003 04:49:12 -0800 (PST), CM wrote: hi, i am trying to install iaxcomm-win-20031117.zip in my windows xp machine. i am really messed up with the wxwindows and the xrc support thing. can anyone give me links to which i need to download for this new version to work? i installed he older version succesfully... when i installed new version it just disappears.. i am missing something. i did download a 12 M wxWin file installed it and again installed the ne iaxComm.. but i don't see anyting.. its running in the processes of the task manager in xp. Sorry about that. The 20031117 version contains a bug in the layout code that only causes crashes on some machines. Try this: http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-current.zip for an updated binary that also does a better job of looking for the rc files. I'll update the web page this afternoon. That link will always point to the latest public build, and will point to a newer version this afternoon. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm new version installation problem
On Thu, 20 Nov 2003 09:06:40 -0800 (PST), you wrote: thx. it solved my problem. why not put the working app in the website so that ppl won't get my kind of problem I was doing the linux build when I responded to your mail. By the time you read this they should both be up there, along with an updated web page. I don't have a Mac, so that binary will have to wait. The newest build that you'll see has a minor change and includes a quickstart doc. Please let me know what you think. cm --- Dan [EMAIL PROTECTED] wrote: Hi, - Original Message - From: C M [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 20, 2003 3:26 PM Subject: Re: [Asterisk-Users] iaxComm new version installation problem sorry dan, i downloaded the file from http://iaxclient.sourceforge.net/iaxcomm/index.htmland file iaxcomm-win-20031117.zip but it does not seem to work... it just disappears in the background.. i can see it running in the task manager thing. and my computer gets really slow. This is the buggy one. Try this link (from 18nov): http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-current.zip This one works. Good luck, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ 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] new iaxComm build available
I tracked down the problem with crashes at startup (and with no window appearing). I was trying to make the last column of the call list as wide as possible when resizing the window. With some display settings, the reported width of the client window is off by 1 pixel. Instead of automagically adding a scrollbar, wxWindows starts running around in circles. Sooner or later (sometimes much later) it crashes. I've left a couple of pixels' buffer, and redone the dialling logic. The most frequent critique I've gottn is that it is cumbersome to require server configuration to dial a number. Now you don't. If the dialstring in the text box contains a username, hostname and extension, then it is dialled as-is. If there's a hostname and extension, we try to lookup username and password from a registered server with that name. If there's no server in our config with that name, dial it as-is (assume no username needed) If there's just an extension, then use the default server that you've (hopefully) configured. Windows : http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-current.zip Linux: http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-lin-current.tar.gz Next features to add are optional autoanswer for intercom functionality and streamlining the phonebook (only storing dialstrings) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new iaxComm build available
On Thu, 20 Nov 2003 16:07:46 -0800, TC [EMAIL PROTECTED] wrote: Could you make it so that the Number Line Appearances are a configurable Options/Preferances Do you mean the number _of_ line appearances? Sure can, I'll add it in real soon. (Probably tomorrow). I like to use these softphones for testing as well and its nice to load up the calls this way ___ 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] Updated iaxComm binaries available for WinXP, Red Hat 9.0
On Wed, 19 Nov 2003 09:18:44 +0100, Peer Oliver schmidt [EMAIL PROTECTED] wrote: Michael, I have the same problem with running iaxcomm. Did the following: * Extract to c:\cd * Open command prompt * c: * cd \cd * iaxcomm What happens: A cursor change for a couple of seconds from an arrow to an arrow with a clock. iaxcomm never shows up. I think that I've found the problem. I installed on about every Win32 box I could get my hands on, and could only get two to crash. I think th problem has to do with different display metrics that uncover a wxWindows bug. I've posted a revised binary at http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-current.zip Please let me know if this version works for you. BUT, iaxcomm makes the computer nearly unusable. Killing iaxcomm in the task managers process list helps to make the computer usable again. Where are the configuration information stored? Might it be, that some old configuration information is being used which is no longer of use? It's in the registry in \HKEY_CURRENT_USER\Software\iaxComm All of the talk of USB keys has convinced me to (at least optionally) move it to a config file. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
On Tue, 18 Nov 2003 17:02:42 +0200, Dan [EMAIL PROTECTED] wrote:
Hi,
Tried on WinXP Pro and it loads, but in the background (no window).
There is something needed from the wxWindows package to just run the
executable?
Nothing needed from the wxWindows package. I think it's because it can't find
the rc directory.
I'm sorry that I didn't put this in the README. Bad coder. No donut.
You must run iaxComm from the installation directory beacuse it looks for rc
files in ${cwd}/rc.
Steve put an error dialog on failure in the CVS sources, but I'm working on a
better solution.
Please let me know if this solves it, or if the problem lies elsewhere.
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[Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
iaxComm is a cross-platform IAX2 softphone available for Win32 and Linux. Win32 and Linux binaries as well as the LGPL source are available at: http://iaxclient.sourceforge.net Recent improvements are a less cluttered user interface, audible ringback and audible outgoing ring, and of course IAX2 protocol support. iaxComm is based upon the wxWindow GUI framework and compiles on Microsoft Windows, Linux, and OS X. It runs on Windows XP, Windows 2000, and Red Hat 9.0. It doesn't (yet) run on OS X. The digium S100U is supported for handset audio only (no hook state detection, no DTMF decoding, no ringing) with the IPO-11 audio drivers. I've been trying to get a Windows SDK, but no luck so far. Please send me bug reports, critiques, and feature requests. I would like to get a 0.99 release out by 1 DEC 03. If anyone is interested, there is a mailing list for the iaxclient library upon which iaxComm is written. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
On Tue, 18 Nov 2003 11:35:01 +1100, Peter Brown [EMAIL PROTECTED] wrote: At 17:45 17/11/03 -0600, you wrote: iaxComm is a cross-platform IAX2 softphone available for Win32 and Linux. Win32 snip Doesn't run on Win NT4? It probably does. I don't have an NT4 box to test it on. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
On Tue, 18 Nov 2003 11:35:01 +1100, Peter Brown [EMAIL PROTECTED] wrote: Doesn't run on Win NT4? I just verified that it works under Win98 on a Thinkpad 560. I got it to launch on Windows NT4 Workstation. That box doesn't have a sound card, so I couldn't verify operation. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax configuration
On Fri, 14 Nov 2003 05:36:25 + (GMT), mukta vasudeva
[EMAIL PROTECTED] wrote:
Hi,
I have configured 3 users in my iax.conf, i am using iaxcomm phones. Iaxcomm has
excellent voice quality although there is no ringing tones(either ring back or
ringing tone),but i can live without right now.
Check the website this weekend. We've just added ringback and ringtone.
I find that for each user i want registered i have to add his name and his ip
address.I have been using host = dynamic.Isnt there any way that i can define a
dialmap such as _7XXX and all users can then be registered with the server and get
allocated the their individual numbers by the server.(till now i define the numbers
with callerid field). i need to do this so that i can add 15+ users without having to
add each individually.
what would be the entry inthe iax.conf for this?
something like:
[_7XX]
type=friend
host=dynamic
auth=plaintext
secret=bla /*all phoens would have the same usrname passwrd
I don't _think_ you can use wildcards in iax.conf like that. I think you'll
have to make all fifteen entries.
and in extensions.conf, i would just have to use something like the following:
exten = _7XX,1,Dial(IAX/_7XX/s,100,r)
You want to use ${EXTEN} in the dial string. I use
exten = _2XX,1,Playback(loligo/pls-wait-connect-call)
exten = _2XX,2,Dial(IAX2/${IAXINFO}/${EXTEN})
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Re: [Asterisk-Users] IAX2 based software client ..pls help
On Thu, 13 Nov 2003 13:10:52 +0200, Dan [EMAIL PROTECTED] wrote: Hi, I am very closed to implement the IAX2 version in DIAX, but still some issues which I don't know how to handle, maybe someone from this list can help me. Trying to register with the * server as in version 1, I get the following in the * console: NOTICE[1150495040]: File chan_iax2.c, Line 2919 (register_verify): Inappropriate authentication received and in the client: Registration rejected Since you are using the iaxclient_lib core, your should really just need to change IAXVER=1 to IAXVER=2 in the iaxclient/lib/Makefile Note that the .remote element of the iaxc_ev_call_state struct only holds callerid number when you switch to IAX2, rather than the name and number. (I added a remote_name element, because I use that in iaxComm) There is something to be changed in iax.conf file regarding the user definition? there is an iax2.conf file too? I ask this because in the iax.conf file there is a line in the general section: port=5036 which is specific to IAX1 I've always run with the port line commented out. You'll notice that there is a console error associated with this. I always took it to mean that IAX2 would be disabled. Thanks, Dan ___ 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] IBM to Run VoIP On Linux
On Fri, 7 Nov 2003 23:50:06 -0800, Darren Martz [EMAIL PROTECTED] wrote: If I'm out of place in the following suggestions, I'm sure others will tell me grin - Create a clean SDK of the wonderful IAX2 protocol for Win32 and Mac to gain exposure everywhere - Push, entice, bribe IP phone designers to support the IAX2 protocol based on the clean and easy to use SDK - Someone once suggested an Asterisk logo program, excellent idea Darren, Take a look at iaxclient.sourceforge.net The current CVS version supports IAX or IAX2, and works on Win32, ia386Linux and Macs. There are also a few working crossplatform softphones there. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
On Sat, 8 Nov 2003 21:59:43 +0100, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: Take a look at iaxclient.sourceforge.net The current CVS version supports IAX or IAX2, and works on Win32, ia386Linux and Macs. There are also a few working crossplatform softphones there. ...and iaxclient is probably not one of them. Working softphones for me includes stability, intuitive user interface at first. AFAIK, iaxclient lacks both Could you offer some constructive feedback? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USB handsets/headsets??
On Thu, 06 Nov 2003 12:22:27 +, you wrote: Anyone got any pointers on where to find USB handsets or headsets that can be used as the audio device on a softphone? The S100U works with iaxComm as a headset. I use a cheap VTech 900 MHz phone with it. 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
[Asterisk-Users] LGPL IAX2 software phone (for WIndows/Linux platforms)
I don't want to rain on Dan's parade, but I'd like to call everyone's attention to an existing project. Steve Kann has developed a crossplatform IAX/IAX2 library, and there are a few clients available for it. I have written iaxComm using the wx toolkit. It compiles and runs under Windows XP using the MinGW tool chain. A precompiled IAX only version is available at iaxclient.sourceforge.net. iaxComm _ought_ to recompile and run on linux, and OS-X as well with minimal tweaking. There is another earlier client called iaxPhone that does run on linux and OS-X. Steve has just updated the library, and iaxComm now uses IAX2 rather than IAX -- just check out the current CVS. (We just fixed a minor registration bug today). I'm no c++ coder, and I built the UI to fit my own needs, but I think that it would be an excellent starting point for anyone who wants an open source, cross platform IAX2 client. NB that iaxComm uses wxWindows' xrc system for the UI, so you can change the appearance of the app even after compiling. If anyone has any critiques, please send me a note directly, or better yet, subscribe to the iaxclient development mailing list. I can already see a few things that I would have done differently if I were developing for general consumption rather than myself. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm - IAX client for Win32
On Wed, 17 Sep 2003 15:14:27 -0500, Josh Roberson [EMAIL PROTECTED] wrote: The copy I downloaded from the website never did register with *. It would make authenticated calls, but wouldn't actually register with the server. Even checked the IAX peers, and nope, wasn't registered. Do you see anything with iax debug? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Van Donselaar Sent: Wednesday, September 17, 2003 1:00 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxComm - IAX client for Win32 On Wed, 17 Sep 2003 11:27:25 +0200, Florian Overkamp [EMAIL PROTECTED] wrote: At 19:55 16-9-2003 -0500, you wrote: iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome. Well, this looks like a big improvement, but I cant seem to find the option to register at the asterisk server. Is it impossible, or am I missing it ? Would be a hefty requirement for real use, I think... It automatically registers with all asterisk servers that have been configured in the Options|Directory dialog. I dial out and register from two different servers. I previously had an auto register checkbox, but changed to registering all servers when I moved the servers list from a listcontrol to a combobox. I'm thinking that you would want to register with any server through which you may want to make outbound calls. When the servers are read from the registry, they are read in alphabetical order, and registration is attempted in that order. (The order may be different on other platforms). You should see Registration accepted in the status bar after the last server is registered. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ 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] iaxComm - IAX client for Win32
On Wed, 17 Sep 2003 21:01:02 -0400, Uriel Carrasquilla [EMAIL PROTECTED] wrote: If possible, I'd like to get the source code (don't need Linux or Mac) for Windows, please. http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz gets the source code. There are instructions in iaxclient/simpleclient/wx/README on how to instal/prepare mingw and wxwindows. Also, which C++ compiler should I be using to compile. I have had success with the DOS/prompt version. I used mingw, but I think you ought to be able to use Borland if you tweak the makefile. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Florian Overkamp Sent: Wednesday, September 17, 2003 5:27 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxComm - IAX client for Win32 At 19:55 16-9-2003 -0500, you wrote: iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome. Well, this looks like a big improvement, but I cant seem to find the option to register at the asterisk server. Is it impossible, or am I missing it ? Would be a hefty requirement for real use, I think... Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ 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
[Asterisk-Users] iaxComm - IAX client for Win32
iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iLBC, Speex and X-Lite
I've been trying out the newest X-Lite (Build 1012) with iLBC and speex codecs. If I enable only iLBC _or_ SPX on X-Lite and call the echo-test on my asterisk server, the call connects, but I get no sound. If I enable only iLBC _and_ SPX, X-Lite indicated that it has connected with iLBC, and I hear a weird squawking. My sip.conf contains: allow=iLBC allow=SPEEX allow=gsm I've heard that asterisk and kphone will work together using iLBC. Does anyone else have any interoperability experience? Any ideas on what I should do to debug this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
