[asterisk-users] Mediatrix 1124 setup
I recently purchased a Mediatrix 1124 from an auction of a company that went out of business. It came with nothing other than the unit itself. In digging thru the Mediatrix web site, and various google searches, it looks like it only supports SNMP setup, and only with their software (or the correct MIB). However, Mediatrix doesn't appear to let you download said software or MIB from their web site. Does anyone know where I can get the setup software or MIB needed to program this thing? I *think* I need the correct one for its firmware version, but I can't find out how to tell what version firmware it has. There is what appears to be the remains of a sticker marked Rev 4 on the bottom if that is any help. I have been able to default the unit and it properly gets a DHCP lease, but doesn't appear to respond to anything other than pings, a port scan reveals no open TCP ports on it. That is as far as I have gone with it so far. I figured I'd ask around for the setup software before I struggled too much more with it. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Zaptel module compile woes
That seemed to work well. Thank you! On a side note, now that I have the zaptel module installed, how can I uninstall it? I've been googling for over an hour and have had no such luck. If all else fails I'll try the IRC channel in the AM. Thanks again for your guidance and help, Phil On Sun, Dec 10, 2006 at 11:12:32AM -0500, Phil Finkler wrote: I'm running Debian 3.1 with the 2.6 kernel. I've also got the kernel source installed and compiled from source so I'm assuming I don't need the kernel headers. Can you elaborate on what m-a a-I zaptel refers to? Are those flags for running make linux26? I haven't come across that in any of the documentation I've been reading. m-a -k /path/to/kernel/source/dir a-i zaptel I generally prefer, though: m-a -t -i -k /path/to/kernel/source/dir a-i zaptel as the module assistant user interface is not very useful. I also tend to build debs as a user: m-a -u . -t -i -k /path/to/kernel/source/dir a-i zaptel And for more information: man m-a -- Tzafrir Cohen icq#16849755jabber:tzafrir at jabber.org http://lists.digium.com/mailman/listinfo/asterisk-users +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://lists.digium.com/mailman/listinfo/asterisk-users http://www.xorcom.com iax:guest at local.xorcom.com http://lists.digium.com/mailman/listinfo/asterisk-users /tzafrir ___ --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] Jabber Client
Look for Asterisk-IM (http://www.jivesoftware.org/asterisk-im/) It'll work with your jabber server. I don't know if it 'works' with Exodus, but it will show if you're on the phone. On 12/9/06, Shinji Kawamoto [EMAIL PROTECTED] wrote: Hello, I would like to connect jabber client (Exodus) to Asterisk 1.4. Asterisk 1.4 already supported jabber (client/component), didn't it? Can I connect Exodus to Asterisk directly? And if yes, please tell me how to connect? Kawamoto ___ --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
[asterisk-users]: Some warnings occur
Hello all I have installed asterisk1.2.12.1 on enterprise linux and i am using soft phone XLite , the manual calls connected properly but when I logging on VICIDIAL than it gives thi error Dec 11 10:26:05 WARNING[7215]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64) Dec 11 10:26:05 WARNING[7215]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64) Dec 11 10:26:05 WARNING[7215]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64) Dec 11 10:26:05 WARNING[7215]: chan_sip.c:2570 sip_write: Asked to transmit fram e type 64, while native formats is 256 (read/write = 64/64) If any one has solution please tell me i really appreciate for the help SHOBHIT NIRALA CONT NO. 9871476403 - Everyone is raving about the all-new Yahoo! Mail beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] X100P clone dial problems.
I'm not sure if I have a configuration problem or not. I am unable to dial out. When I try to dial in I can hear the phone ring on the dialling phone but Asterisk does not register anything. In zaptel.conf I have loadzone = au defaultzone=au fxsks=1 In zapata.conf language=au context=from-pstn When I do: zap show channels I get: Chan Extension Context Language MusicOnHold pseudo from-pstn au running ztcfg -vv shows Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. when I try to make a call I get: -- Executing Dial(SIP/201-08c237c0, Zap/1/1800456456|70) in new stack Dec 9 13:52:54 NOTICE[4697]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) Please help. ___ --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] Asterisk 1.4b3 Realtime Voicemail
On 12/11/06, David Thomas [EMAIL PROTECTED] wrote: I only say two options for voivemail staorge when compiling 1.4, IMAP and ODBC. Are you using one of these? Are they configured? Does anyone know if Version 1.4 still does filesystem based storage of voicemail or if you must use IMAP or ODBC? David This seems to be fixed in the svn trunk r77. Seems to work now when I did a checkout from SVN. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk 1.4b3 Realtime Voicemail
On 2006-12-10 04:13:02 -0800, RR [EMAIL PROTECTED] said: Hello, does anyone else have a problem with Asterisk crashing right after a valid password/PIN is entered when trying to access voicemail in the 1.4b3 version? Not sure if this is anything to do with realtime per se but I keep getting the asterisk process bail on me as soon as a valid PIN is entered. Sometimes if there is a message in a format that voicemail doesn't like, it crashes like that. Make sure the voicemail box is empty and try again... I have seen it crash like that with audio data it didn't like going back to before 1.2. Marty ___ --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] Re: Asterisk 1.4b3 Realtime Voicemail
On 12/11/06, Martin Joseph [EMAIL PROTECTED] wrote: Sometimes if there is a message in a format that voicemail doesn't like, it crashes like that. Make sure the voicemail box is empty and try again... I have seen it crash like that with audio data it didn't like going back to before 1.2. Marty Yeah I thought it could be that but this was brand spanking new DB with no messages in it. It was only when I turned on verbose logging on (*) console that I actually saw what it crapped out on. It was do with non-definition of the odbc_request_obj function. So the bottom line is that you can't use voicemail realtime with ODBC with the stock 1.4b3 release unless you update all relevant files that are affected or just checkout the entire SVN trunk. A fair bit seems to have changed in the voicemail code alone since the 1.4b3 release ___ --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] X100P clone dial problems.
On Mon, Dec 11, 2006 at 04:18:20PM +1100, Klaverstyn, David C wrote: I'm not sure if I have a configuration problem or not. I am unable to dial out. When I try to dial in I can hear the phone ring on the dialling phone but Asterisk does not register anything. In zaptel.conf I have loadzone = au defaultzone=au fxsks=1 In zapata.conf language=au context=from-pstn Those need to be in the section [channels] and be followed by a channel = 1 to actually have any effect. You also must set signaling (signalling = fxs_ks; in your case). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Re: Zaptel module compile woes
On Sun, Dec 10, 2006 at 11:28:51PM -0500, Phil Finkler wrote: That seemed to work well. Thank you! On a side note, now that I have the zaptel module installed, how can I uninstall it? I've been googling for over an hour and have had no such luck. If all else fails I'll try the IRC channel in the AM. dpkg --remove zaptel-modules-VERSION_NAME e.g: dpkg --remove zaptel-modules-`uname -r` Or use your favourite package management tool. IT is a package like any other. Why would you want to remove it? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New installation CentOS 4 x86 or X86_64
Hi list! I have to do a new bare metal installation of a box running Asterisk with bristuff or vzaphfc. The box will be used as a really lightly loaded file server and pbx. Any advise on which architecture I should use? The cpu is a 64 bit capable AMD (the box is running x86_64 now) but is still suffering from echo on the BRI lines. Should I go with the normal x86 or the 64 bit x86_64 arch.? Thanks!! ___ --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] popups, queue agents
Todd- Asterisk a écrit : Hi everyone - I have a nicely working system to which I'd like to add popups for incoming calls. Calls go into a queue, then all extensions ring. I'd like the agent that answers to call to get the popup on screen. I'm currently using Flash Operator Panel to get a popup (other suggestions welcome). Currently, all users get a popup when the call first goes into the queue which obviously isn't that great Where in the dial-plan do I put the code for the popup and specify only the agent to whom the call is connected? Use the URL option of the Queue application, and a softphone that supports URL: MozPhone is my favorite :) Jean-Denis ___ --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] X100P clone dial problems.
I have since added fxs_ks=1 and channel = 1 This has not fixed the problem. I do notice a warning on the reload of asterisk. WARNING[4296]: chan_zap.c:10874 setup_zap: Ignoring signalling -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, 11 December 2006 4:47 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] X100P clone dial problems. On Mon, Dec 11, 2006 at 04:18:20PM +1100, Klaverstyn, David C wrote: I'm not sure if I have a configuration problem or not. I am unable to dial out. When I try to dial in I can hear the phone ring on the dialling phone but Asterisk does not register anything. In zaptel.conf I have loadzone = au defaultzone=au fxsks=1 In zapata.conf language=au context=from-pstn Those need to be in the section [channels] and be followed by a channel = 1 to actually have any effect. You also must set signaling (signalling = fxs_ks; in your case). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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
