[Asterisk-Users] iax-qos-openbsd...
Hi all, We have an asterisk server inside a network using an iax provider. The firewall is based on Openbsd, and we would like to use PF's QOS capabilities to ensure optimum quality. We need to provide good throughput for other applications, so we need to use scheme that borrows bandwith, that is when there is no VOIP communication, the whole upload capability of our link can be use. We have tried all kind of combinations but could not come up with a satisfactory solution. As anyone faced a similar configuration, if so how did you deal with that? Regards, Francois ___ --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] Problems after upgrade...
Thanks Tom, That was it, after upgrading the kernel with Yum, it didn't change the link for the modules. Fixed it manually, recompile everything and we are up again. Best regards, Francois On Nov 12, 2005, at 12:11 AM, Francois Meehan wrote: Hi all, I have upgrade my kernel and asterisk to their latest release on a Centos 4.1 box, now it won't start anymore. Have you rebuilt Zaptel against your new kernel? If you upgrade the kernel, you need to rebuild zaptel. Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems after upgrade...
Hi all, I have upgrade my kernel and asterisk to their latest release on a Centos 4.1 box, now it won't start anymore. I am using a tdm400p card. I have the following error: Nov 11 23:58:00 WARNING[25135]: chan_zap.c:770 zt_open: Unable to open '/dev/zap/channel': Is a directory Nov 11 23:58:00 ERROR[25135]: chan_zap.c:6239 mkintf: Unable to open channel 2: Is a directory here = 0, tmp-channel = 2, channel = 2 Nov 11 23:58:00 ERROR[25135]: chan_zap.c:9191 setup_zap: Unable to register channel '2' Nov 11 23:58:00 WARNING[25135]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Nov 11 23:58:00 WARNING[25135]: loader.c:440 load_modules: Loading module chan_zap.so failed! I use to load the card by doing modprobe wctdm, now to load the card I must do: modprobe zaptel modprobe wcfxs running ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. So all seem fine for the driver. Any ideas? Francois ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I get a dialtone calling from outside...
Hi all, How can I configure, in extension.conf, to call and extension and have a dialtone so I can compose a number to dialout? Basically, I want to be able, when I am out of the office, to call in my asterisk box and then dialout from it. Regards, Francois ___ --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] How can I get a dialtone calling from outside...
Exactly what I need! Thanks a million. Francois Isn't that what you would call Direct Inward System Access? Probably what you'd find at http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DISA Francois Meehan wrote: Hi all, How can I configure, in extension.conf, to call and extension and have a dialtone so I can compose a number to dialout? Basically, I want to be able, when I am out of the office, to call in my asterisk box and then dialout from it. Regards, Francois ___ --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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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] TFTP and DHCP...
Hi Bob, Found the DHCP options but the phone won't use it :( Thanks for the help, Francois On Sunday 18 Sep 2005 15:15, Francois Meehan wrote: Hi all, I have bought an Aastra 480i phone. In order to configure the phone for using a TFTP server, I had to enter the TFTP ip address directly in the phone, and then reboot the phone again. Is it possible to configure a DHCP server so it sends a TFTP server coordinate for the phone to use? Yes, man dhcp-options, but will the phone use it? Random Thought: --- A man without a woman is like a hawk without a falconer. - Louis Bergsagel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TFTP and DHCP...
Hi all, I have bought an Aastra 480i phone. In order to configure the phone for using a TFTP server, I had to enter the TFTP ip address directly in the phone, and then reboot the phone again. Is it possible to configure a DHCP server so it sends a TFTP server coordinate for the phone to use? Best regards, Francois Random Thought: --- A bottle of wine begs to be shared; I have never met a miserly wine lover. - Clifton Fadiman, 1904 - 1999 Random Thought: --- A bottle of wine begs to be shared; I have never met a miserly wine lover. - Clifton Fadiman, 1904 - 1999 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hitachi wip5000
Hi all, Saw on the net the wip5000 SIP wireless phone from Hitachi, a suprising rig. As anyone successfull in making it work with Asterisk? If so, how do you like it? Regards, Francois Random Thought: --- Molecule, n.: The ultimate, indivisible unit of matter. It is distinguished from the corpuscle, also the ultimate, indivisible unit of matter, by a closer resemblance to the atom, also the ultimate, indivisible unit of matter ... The ion differs from the molecule, the corpuscle and the atom in that it is an ion ... -- Ambrose Bierce, The Devil's Dictionary ___ 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] Paging systems from the phone...
Hi all, Is there a model of IP phone, with a built-in speaker, that can be used as part of a Overhead Paging system? Can Asterisk accomodate such a feature? Regards, Francois Random Thought: --- It isn't necessary to have relatives in Kansas City in order to be unhappy. -- Groucho Marx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P (TDM02B) ceased to work...
Hi all, I am runing Asterisk on Centos 4. This morning I have updated the system using yum, a whole bunch of stuff was upgraded. Since, when I try to start zaptel, I have the following error: Waiting for zap to come online ...OK Loading zaptel hardware modules: Running ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6) [FAILED] Did recompile Asterisk with version 1.0.8, reboot a couple of time, still the same error, the card seem dead... Any ideas? Francois Random Thought: --- Nothing is so awesomely unfamiliar as the familiar that discloses itself at the end of a journey. - Cynthia Ozick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P (TDM02B) ceased to work...
Hi Robert, Did recompile, several time actually, upgraded from 1.0.7 to 1.0.8 with same results, there is no light in the back of the card. Regards, Francois Did you recompile and reinstall the zaptel source? I had to do this myself recently on a fedora core 2 update/upgrade Francois Meehan wrote: Hi all, I am runing Asterisk on Centos 4. This morning I have updated the system using yum, a whole bunch of stuff was upgraded. Since, when I try to start zaptel, I have the following error: Waiting for zap to come online ...OK Loading zaptel hardware modules: Running ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6) [FAILED] Did recompile Asterisk with version 1.0.8, reboot a couple of time, still the same error, the card seem dead... Any ideas? Francois Random Thought: --- Nothing is so awesomely unfamiliar as the familiar that discloses itself at the end of a journey. - Cynthia Ozick ___ 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 -- Robert P. McKenzie, CSTA | GammaRay Technical Services Ltd [EMAIL PROTECTED] | [EMAIL PROTECTED] http://www.uk-experience.com | http://www.gammaray-tech.com Ecademy Profile: http://www.ecademy.com/account.php?op=viewid=64014 Random Thought: --- Four things cannot be hidden -- Love, smoke, a pillar of fire, and a man striding across the open bled. -- Fremen Wisdom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P (TDM02B) ceased to work...
Got it, For some strange reasons, neither wctdm nor wcfxs get loaded when starting zaptel (/etc/init.d/zaptel start). By manually modproble wctdm everything works. Have all a nice week. Francois Hi all, I am runing Asterisk on Centos 4. This morning I have updated the system using yum, a whole bunch of stuff was upgraded. Since, when I try to start zaptel, I have the following error: Waiting for zap to come online ...OK Loading zaptel hardware modules: Running ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6) [FAILED] Did recompile Asterisk with version 1.0.8, reboot a couple of time, still the same error, the card seem dead... Any ideas? Francois Random Thought: --- Nothing is so awesomely unfamiliar as the familiar that discloses itself at the end of a journey. - Cynthia Ozick ___ 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 Random Thought: --- I suppose we acquire most of our feelings about our bodies too early, and in ways too complicated, to make them easy to account for. - Charis Wilson ___ 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] iaxy and cvs head...
Hi all, After upgrading to latest CVS head, I have problems using a IAXY device, having slin problems: Jun 15 18:59:31 NOTICE[8197]: channel.c:1475 ast_read: Dropping incompatible voice frame on IAX2/lise-1 of format slin since our native format has changed to ulaw Because of that outside caller can't ear the callee on the IAXY. Found somewhere that disabling transcode in asterisk.conf would fix the problem, so I added, not sure of the syntax the following section in asterisk.conf: [options] transcode_via_sln=no That didn't work, and I am not sure I am using the wright syntax... I have revert back to stable release and everything is ok, but I want to test SCOPSERV-VoIP and it requires version 1.07 or higher... Regards, Francois Random Thought: --- The supreme accomplishment is to blur the line between work and play. - Arnold Joseph Toynbee, 1889 - 1975 ___ 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] Problem with slin
Hi all, After upgrading to lates CVS head, I have problems using a IAXY device, having slin problems: Jun 15 18:59:31 NOTICE[8197]: channel.c:1475 ast_read: Dropping incompatible voice frame on IAX2/lise-1 of format slin since our native format has changed to ulaw Because of that outside caller can't ear the callee on the IAXY. Found somewhere that disabling transcode in asterisk.conf would fix the problem, so I added, not sure of the syntax the following section in asterisk.conf: [options] transcode_via_sln=no That didn't work, and I am not sure I am using the wright syntax... I have revert back to stable release and everything is ok, but would like to give latest release a try... Regards, Francois Random Thought: --- Democracy is more cruel than wars or tyrants. - Seneca, Epistulae morales ad Lucilium ___ 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] upgrade problems...
Hi all, I have done a major upgrade, first I have bought a Digium TDM400P (TDM02B) 2-Port FXO to replace x100p cards, than moved from an old version Asterisk to a new centos 4 box, compiled asterisk from Asterisk CVS-HEAD built on 2005-06-11. Seems that my iaxy is now incompatible with the TDM400P. When I receive a call from the TDM02B, the caller can't hear my voice (I am using the iaxy) and on the console I see the following errors: Dropping incompatible voice frame on IAX2/lise-1 of format slin since our native format has changed to ulaw Change Iaxy to use adpcm now I am getting: Dropping incompatible voice frame on IAX2/lise-2 of format slin since our native format has changed to adpcm. I can call inside from eyebeam to Iaxy no problems. The Eyebeam can received call from outside no problem either. Here is my zapata.conf: [channels] signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived musiconhold=default context=maison channel = 2 context=cedval channel = 1 Here is my zaptel.conf: fxsks=1-2 defaultzone=us loadzone=us Any ideas? Best regards, Francois Random Thought: --- Remember Kruschev: he tried to do too many things too fast, and he was removed in disgrace. If Gorbachev tries to destroy the system or make too many fundamental changes to it, I believe the system will get rid of him. I am not a political scientist, but I understand the system very well. I believe he will have a heart attack or retire or be removed. He is up against a brick wall. If you think they will change everything and become a free, open society, forget it! -- Victor Belenko, MiG-25 fighter pilot who defected in 1976 Defense Electronics, Vol 20, No. 6, pg. 110 ___ 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] 2 incoming lines and Asterisk@home...
Hi all, Is it possible to use 2 incoming fxo lines (one is for my company the other for the family) with [EMAIL PROTECTED] Best regards, Francois Random Thought: --- Errors like straws upon the surface flow: Who would search for pearls must dive below. - John Dryden, 1631 - 1700 ___ 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] 2 incoming lines and Asterisk@home...
Thanks Wiley, I was asking the question because, I don't have my card yet (TDM02B 2-Port FXO) but did install [EMAIL PROTECTED] on a server and when looking at the AMP-setup Incoming calls section, I see only one destination for all incoming calls. I assume then that once the card is installed, there will 2 Incoming calls sources in AMP... Am I right? Regards, Francois This is assuming you have problems with the autoconfig. The latest seems to add the lines just fine. When I started using 0.06, I had to do it manually. W -Original Message- From: Wiley Siler Sent: Thursday, June 02, 2005 9:29 AM To: '[EMAIL PROTECTED]'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 2 incoming lines and [EMAIL PROTECTED] You can support as many as you want. You just need to update your zapata.conf file and change this line... channel=1-8 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francois Meehan Sent: Thursday, June 02, 2005 9:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 2 incoming lines and [EMAIL PROTECTED] Hi all, Is it possible to use 2 incoming fxo lines (one is for my company the other for the family) with [EMAIL PROTECTED] Best regards, Francois Random Thought: --- Errors like straws upon the surface flow: Who would search for pearls must dive below. - John Dryden, 1631 - 1700 ___ 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 Random Thought: --- Fashion is what you adopt when you don't know who you are. - Quentin Crisp ___ 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] Directory config...
Hi all, How do I config Asterisk so when the directory cmd is used, the name of the found entry comes from a pre-record gsm file instead of being spelled letter by letter? Regards, Francois Random Thought: --- All of us failed to match our dreams of perfection. So I rate us on the basis of our splendid failure to do the impossible. - William Faulkner, 1897 - 1962 ___ 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] Directory config...
Right on! Have a good week-end! Francois How do I config Asterisk so when the directory cmd is used, the name of the found entry comes from a pre-record gsm file instead of being spelled letter by letter? If the user as recorded is name, this file will be used. When it's not recorded, * will spell it. Dial to your voicemail and navigate thru the menu to record your name. hth ___ 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 Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: How do I config Asterisk so when the directory cmd is used, the name of the found entry comes from a pre-record gsm file instead of being spelled letter by letter? If the user as recorded is name, this file will be used. When it's not recorded, * will spell it. [...] Content analysis details: (0.6 points, 5.0 required) pts rule name description -- -- 0.5 FROM_ENDS_IN_NUMS From: ends in numbers 0.0 RCVD_BY_IP Received by mail server with no name 0.1 FORGED_RCVD_HELO Received: contains a forged HELO Random Thought: --- I just remembered something about a TOAD! ___ 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] Eyebeam - asterisk - Messenger
Hi Ignacio, Here the info: eyeBeam 1.1 3003x stamp 16296 Asterisk CVS-HEAD-12/05/04-09:27:10 Regards, Francois i did evrything you mentioned, i thing is for my eyebeam version, mine is 3002s what`s yours? On Fri, 28 Jan 2005 23:10:40 -0500 (EST), Francois Meehan [EMAIL PROTECTED] wrote: Thanks Wessel, You really have to know about that little switch on button, I had 2 eyebeam connected with their cameras, no video, 5 min. after I tried the little button and it worked. Must be the effect of the first rhum of the week-end... Tried with (windows) messenger, it would not go. That is why I bought another eyebeam and it's all working now. Thanks to the vpn, video communications from the outside work like a charm. Guess I will stick with eyebeam for now. Also, for the record, in addition to videosupport=yes, the video codecs must be enabled in the sip.conf: allow=h261 allow=h263 and for each phone I have put: canreinvite=no Have a good week-end, Francois Just add a line to your sip.conf: [general] videosupport=yes And to enable video with eyeBeam press the switchon button on the screen :-) Wessel -Original Message- From: Ing. Ignacio Ortega A. [mailto:[EMAIL PROTECTED] Sent: Friday, January 28, 2005 19:33 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Eyebeam - asterisk - Messenger did you find how to configure video with eyebeam using asterisk because i wasn`t able to do it yet as well i want to se messangin with it ThanK You On Fri, 28 Jan 2005 13:23:46 -0500 (EST), Francois Meehan [EMAIL PROTECTED] wrote: Hi all, I would like to connect in sip mode an Eyebeam client to a messenger via Asterisk. I want to use video. Nat is not an issue as vpn connections will be used. Is this a difficult tasks, can someone give me some pointers to get started... Have a good week-end, Francois Random Thought: --- Wanna buy a duck? ___ 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 Random Thought: --- When the eyes say one thing and the tongue another, a practiced man relies on the language of the first. - Ralph Waldo Emerson, 1803 - 1882 ___ 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 Random Thought: --- The important thing is not to stop questioning. - Albert Einstein, 1879 - 1955 ___ 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] Eyebeam - asterisk - Messenger
Hi all, I would like to connect in sip mode an Eyebeam client to a messenger via Asterisk. I want to use video. Nat is not an issue as vpn connections will be used. Is this a difficult tasks, can someone give me some pointers to get started... Have a good week-end, Francois Random Thought: --- Wanna buy a duck? ___ 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] Eyebeam - asterisk - Messenger
Thanks Wessel, You really have to know about that little switch on button, I had 2 eyebeam connected with their cameras, no video, 5 min. after I tried the little button and it worked. Must be the effect of the first rhum of the week-end... Tried with (windows) messenger, it would not go. That is why I bought another eyebeam and it's all working now. Thanks to the vpn, video communications from the outside work like a charm. Guess I will stick with eyebeam for now. Also, for the record, in addition to videosupport=yes, the video codecs must be enabled in the sip.conf: allow=h261 allow=h263 and for each phone I have put: canreinvite=no Have a good week-end, Francois Just add a line to your sip.conf: [general] videosupport=yes And to enable video with eyeBeam press the switchon button on the screen :-) Wessel -Original Message- From: Ing. Ignacio Ortega A. [mailto:[EMAIL PROTECTED] Sent: Friday, January 28, 2005 19:33 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Eyebeam - asterisk - Messenger did you find how to configure video with eyebeam using asterisk because i wasn`t able to do it yet as well i want to se messangin with it ThanK You On Fri, 28 Jan 2005 13:23:46 -0500 (EST), Francois Meehan [EMAIL PROTECTED] wrote: Hi all, I would like to connect in sip mode an Eyebeam client to a messenger via Asterisk. I want to use video. Nat is not an issue as vpn connections will be used. Is this a difficult tasks, can someone give me some pointers to get started... Have a good week-end, Francois Random Thought: --- Wanna buy a duck? ___ 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 Random Thought: --- When the eyes say one thing and the tongue another, a practiced man relies on the language of the first. - Ralph Waldo Emerson, 1803 - 1882 ___ 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] Weir long distance behaviour...
Hi Wilson, I had both features enabled in my zapata.conf file, I will try disabling the callprogress see if it makes a difference, what troubles me is that I have no problems with local calls, what could be the difference with long distance one? I am from Quebec, Ile-Perrot near Montreal. Regards, Francois There is a strange behavior, when we do long distance calls, it keeps ringing on our end, remote callee answers the call but hear nothing. Look up callprogress and busydetect are you in France by any chance? Look here also http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf ___ 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 Random Thought: --- Business will be either better or worse. -- Calvin Coolidge ___ 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
