similar to: OS X support?

hi all does anyone have the soundcore lib? I need it for a slackware installation of gnophone ... perhaps it's time to return it to the new cvs? -- 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 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 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, 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 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.

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

see below I tried to call 98013356 from the following phone (from mgcp.conf) [iptlf03] host = 192.168.33.3 context = default inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4

I would like to add the ability for our users to be able to press "0" whenever reaching someone's voicemail box to re-reroute them to the auto-attendant. Here's a sample extensions.conf: [incoming] include => ciscophones exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,BackGround(auto-greeting)

hi all I'm trying to get this Symbol NetVision H.323/802.11b phone (http://www.symbol.com/products/wireless/netvisionphone.html) connected to asterisk, but it doesn't look an easy task - at least without this serial cable.. Anyone who've succeded this? roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air

hi all I'm being spammed by these messages in the console (see below) and sound doesn't work with today's cvs. I rolled back a week, and it works fine. In addition to the sound problems, I had to enable inband dtmf squelch on the dilnk mgcp phones. if not, each pressed key was counted twice NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP ast_dsp_process

I am trying to update my source code but it never completes, the cvs update just stalls indefinately.. I have now deleted the source dir's and tried doing a fresh cvs checkout but it does the same thing.. Anyone else having this problem? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

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.