search for: luki

...pass different variables for each extension (SIP_CODEC and ALERT_INFO) and this works quite well -- but I don't think this is the cleanest way of doing it. I could certainly tweak it to have different timeouts per extension, but I'm looking for ideas/feedback before I do this. Thanks... --Luki

Guys. Anybody know why sometimes on some servers Asterisk shows more than once while doing a ps? [root@server2 akrall]# ps -ax|grep asterisk 20555 ? S 0:00 /bin/sh /usr/sbin/safe_asterisk 20557 ? S 0:00 asterisk -vvvg -c 20558 ? S 0:00 asterisk -vvvg -c 20560 ? S 0:00 asterisk -vvvg -c 20561 ? S 0:00 asterisk -vvvg -c 20562 ? S

A BETA firmware upgrade toasted my ATA286. It now has limited operations. It will get an IP address via DHCP and register to the last configured SIP server, but the web interface is gone as is the voice config menu. Apart from registration, there doesn't appear to be any other SIP functionality. An Ethereal dump does not show the device trying to grab a new firmware via tftp on bootup, so

Anyone know how to override the 99 message limit in voicemail? (yeah, we have a public VM that gets that many a day). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050812/1bff12e4/attachment.htm

I have many remote locations that dial into a central server to retrieve voicemail via IAX. Outbound calls are handled as SIP calls from a Snom to a local (to them) Asterisk server that dials the main server thru IAX. I have trained them to check their voicemail via the emailed WAV file, however some of them are, how shall I put it, idiots*, and insist that they *have* to have the MWI indicator

...ilto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mohammed Firdosh Nasim Sent: Tuesday, March 15, 2005 11:08 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to * On Sat, 2005-03-12 at 07:42, Luki wrote: > Firdosh, > > there were couple typos on my last email, but that's essentially what > I said. There are two ways of doing it -- but neither will work given > you current setup. > > 1) Phone A talks directly to B. > 2) Both Phone A and B talk to a common point C...

Hi All, I have two SIP softphones(Windows Messenger) running on different subnet (Phone-1 on IP XXX.XXX.25.ABC & Phone-2 on IP XXX.XXX.15.XYZ) and my Asterisk Server is running on IP XXX.XXX.25.PQR.Because of some security issues both the subnets are completely isolated ( U cant even PING from one to other) and I want to connect Phone-1 & Phone-2 to the *. How can I proceed? Please

Hi all, I've just bought some SPA922. First time with this hardware for me. I see no LAN tab in its web GUI where I can setup NAT for PC conected to its LAN ethernet port. However, when I connect a PC to that port, SPA922 works as bridge. Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist such LAN tab for setting up parameters as port forwarding? (by the way,

...as both accounts use IP:5060 so Asterisk cannot tell them apart during the initial peer matching stage. Of course the source port the Cisco selects is different with every dialog, so that doesn't help either. Any input would be appreciated before I throw that phone out of the window. Thanks, Luki

Hi everybody, sometimes we have an Asterisk-crash, but no clue why this is happening, so I'm trying to make a coredump to analyse it. I compiled Asterisk 1.4.20.1 on CentOS 5.4 i386 with "DEBUG_THREADS" and "DONT_OPTIMIZE", then I start it with: # /bin/bash /usr/sbin/safe_asterisk This should do an "ulimit -c unlimited", but I entered it in the terminal again.

PLEASE CONFIGURE YOUR AUTORESPONSERS TO NOT SEND MESSAGES TO PEOPLE WHO POST IN MAILING LISTS YOU SUBSCRIBE TO. This is an extremely rude thing to allow, and is becoming increasingly common, especially with users of the Asterisk-Users list. Daryl _____ From: s.speckenheuer@posservice.de [mailto:s.speckenheuer@posservice.de] Sent: Friday, February 04, 2005 6:41 PM To: Daryl G.

...der. That reasonable fee doesn't actually include the thing working more than 50% of the time. -A. ANDREW - Thanks. I'd be happy to consider any US provider that can transfer / retain my POTS phone numbers. Any recommendations> RE Message: 7 Date: Sat, 7 May 2005 20:28:57 -0700 From: Luki <lugosoft@gmail.com> Subject: Re: [Asterisk-Users] At home Asterisk via Broadvoice? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> My thoughts: 1) I would not run asterisk on a laptop, or on Windows (if you can get it to turn properly via e...

On Feb 2, 2008 10:16 AM, Luki <looky at poczta.fm> wrote: > Hello Carlos > > could you please help me to solve issue with Mustek 1000VA UPS under > Freebsd ? > I am not able to connect with it : > > >Network UPS Tools - UPS driver controller > >Network UPS Tools - Megatec protocol driver 1...

Yeah, I'd like to hear you guys' opinion instead of CleverNickName's! -----Original Message----- From: Luki [mailto:lugosoft@gmail.com] Sent: Tuesday, February 15, 2005 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Comments on Vonage SIP port blocking complai nts?? You can always visit Slashdot for countless (useless, well, not always) comments: h...

Hi, Is it possible to put some kind of bridge which will do traffic shaping/prioritising between my 6 external IP addresses and my PPPoA modem interface? My other option is to put some kind of device at the edge of all my networks to shape the traffic in/out. I'd rather do it in one box if possible? thanks Mike

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone Due to security reasons I want to run asterisk as a non root. I normaly installed asterisk, created an * user, moved the binaries to /usr/bin and chowned all the files and directories mentiont in the * manual (handbook-draft.pdf) Now I can start * but I get the following warning (which I don't get if I run it as a root): Feb 14

...h then PCI bus/hard rive could do :) Asterisk can be directed to save files to tmp and them you can move the files to remote server with least possible priority. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Luki Sent: Monday, 10 April 2006 18:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: update - 512 Simultaneous Calls with DigitalRecording > Has anyone seen these solid state "Drives" from gigabyte yet? - > http://www.pcper.com/article.php...

We're investigating audio quality issues in our system; maybe someone can help. We're using Asterisk as a basic PBX, with a single PRI on one side and SIP phones on the other: Sipura SPA-841's. We're experiencing several audio effects which seem to commonly correspond to network failures (packet loss, high jitter, etc manifested as "robot voice", dropouts, periodic

Hi, Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one

...tached (Learn from this Vonage!). > > > >-----Original Message----- > >From: asterisk-users-bounces@lists.digium.com > >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > >Disgruntled Asterisk Luser > >Sent: Sunday, May 08, 2005 9:06 PM > >To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion > >Subject: Re: [Asterisk-Users] RE: Asterisk at home with Broadvoice? > > > >Don't worry about these subtle details. Broadvoice has been off the > >air for > > > >almost a solid week, with no real ex...