similar to: Is the CVS server working??

hi all it seems like whatever I do, I can't use more than 1 BRI on my AVM C2 with chan_capi. Both channels seem to work, but not at the same time. And - yes - they're connected to different NT boxes :) Any ideas? kapejod? -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open

Hi all, I'm currently evaluating ideas for a phone system for our new office and I have a copy of Asterisk along with a couple of H.323 phones setup in the lab here. I'm very impressed - thanks for the great software! (I was especially impressed that the IAX connection to Digium worked out of the box like that - someone at Digium no doubt has a voicemail from me this morning!) ;-) What

hi Can I use Asterisk with OS X? roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows.

hi all when typing 'exit', asterisk complains and tells me to use STOP NOW, but not with CTRL+D! CTRL+D stops Asterisk immediately, also when calls are in progress (when asterisk is started with -c). Any chance to change this behaviour? roy *CLI> exit The QUIT and EXIT commands may no longer be used to shutdown the PBX. Please use STOP NOW instead, if you wish to shutdown the PBX.

hi all the help output for DISA ends like below, with the half-sentence 'Note that in the case' what's the rest of that sentence? The file that contains the passcodes (if used) allows specification of either just a passcode (defaulting to the "disa" context, or passcode|context on each line of the file. The file may contain blank lines, or comments starting with

hi Just installed astping in Nagios (former netsaint), and it works fine - for some time. After several hours, the Asterisk manager module just stopped responding, and I had to restart to make it work Any idea what might cause this? Sorry - I don't have the log now. Is it possible to have asterisk log console output somewhere? roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS -

hi all is it hard/possible to move the following from chan_capi_pvt.h into a setting (preferably global) in capi.conf? #define AST_CAPI_NATIONAL_PREF "0" #define AST_CAPI_INTERNAT_PREF "00" and ... Is it hard to move or copy the txgain and rxgain to [global], either as a given 'default' if nothing's set in the interfaces, or as a overall

hi all as with the standard 'shutdown' command, it'd be nice to have a 'canceller' to 'die when convenient'. is this a heavy task to add? roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows.

hi all seems there's a memory leak in asterisk somewhere. probably in chan_mgcp. It seems like 2048kB is allocated but not released each time I lift the handset. This is, however, never released. So... oslpbx:~/debug/20030801112939# ps axfv|grep asterisk | tail -1 3206 pts/0 S 0:00 0 410 35601 6024 1.1 \_ /usr/sbin/asterisk -gvvvvcp (lift handset)

hi all I'm trying to use the manager interface to do some nagios (http://nagios.org/) integration, and I find some parts of it not really optimal. What I'd like to change, is to make \r\n\r\n an actual terminator, something it isn't today, AFACS. Below is the Status output - it shows Response, Message, \r\n, Status post, \r\n, Status post etc etc. Without a parsable terminator, I

hi all when I (using my D-Link DPH-100M MGCP phone) press #, I get told 'transfer'. I dial the new number and after that, * just tells me 'meep meep meep' [hangup] any ideas why? -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows.

In my test box I've installed chan_h323 and I've been testing it with Micro$oft netmeeting and openphone with success. I alos have in my installation a Cisco 1700 series router with an FXS card on it. On the router I places the g711-ulaw codec and it worked but I experienced one bad thing. When I made up more than three calls, in the first three calls I was able to transmit and

Hi, I just made a fresh install on a new box and at the end I got this message: make: warning: Clock skew detected. Your build may be incomplete. I had all the various libs added to a default install of RH 9. Though its possible that I'm short on developer tools. Any clues anyone? -- Steve ______________________________________ This sig is pending approval

ðÏÞÅÍÕ ÂÙ É ÎÅÔ? ÷ÏÐÒÏÓ ÔÏÌØËÏ × ÔÏÍ, ËÁË Õ ÔÅÂÑ ÂÕÄÅÔ ÓÏÅÄÉÎÑÔØÓÑ ÐÁÎÁÓÏÎÉË Ó ÁÓÔÅÒÉÓËÏÍ. Date: Wed, 30 Jul 2003 20:06:17 +0400 From: Pavel Zheltouhov <pavlo@comlink.ru> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk,ata186 and Panasonic TD1232 Reply-To: asterisk-users@lists.digium.com I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk.

IAX2 now has support for a "trunk" mode ("trunk=yes" in the appropriate friend section). Trunk mode allows IAX2 to use bandwidth extremely effectively. The original impetice (and strategy) was a result of a mistake in which it was claimed that Asterisk with a T100P could send 96 simultaneous calls over a single T1 using VoIP. Thanks to a suggestion by the customer, combined

hello, sometimes my capi_channel stop works - e.g. when i try to call number which does not exist ( typo error ) and i must restart asterisk. following lines appears in the log files : ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free channel on controller 1! will continue searching. ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1!

Hi, I'm trying to get H323 support using asterisk 0.4.0 Unfortunately the pwlib and openh323 versions mentioned in the asterisk-oh323 readme file are no more available, and I had to use newer ones. Now I installed all libraries, but got a segemntion fault when starting asterisk while reading the chan_oh323.conf file. When I declare more than 9 gwprefix I get first a error "out of

Hi all, There is something special I must configure in order to get the voice mssage by mail? In voicemail.conf I have: serveremail=asterisk@mydomain.ro attach=yes [default] 301 => 6535,Home Mailbox,dtoma@fx.ro I have tried to let a message for 301, but this message is not forwarded by mail. I am missing something? Thanks, Dan

hi all when starting kphone, it tries to register with asterisk but fails after a while. The SIP entry in * for this user is below. This is identical to the other SIP entries. The other SIP clients are MSN messenger plus one snom. these work fine. See SIP debug output attached as 'screen-exchange' thanks roy [roy] type=friend ;insecure=yes username=roy ;secret=password host=dynamic

Sheesh. I only joined here a few days ago and already there's a flame war. Look, to remove your name from the list is easy. It tells you where to go to manage your subscription down there at the bottom. If you want another mailing list, why not go to yahoo!! or topica and set one up, or set one up yourself. It ain't rocket science with mailman. Even an idiot like me has managed it.