similar to: Queues without members

Hello! One of our guys created a short rap sequence (7 MB): http://www.nativeinstruments.de/tmp/vmrap.wav Andi -- -> Andreas Roedl -> Senior IT Manager -> NATIVE INSTRUMENTS GmbH -> andreas.roedl@native-instruments.de -> Schlesische Strasse 28 -> http://www.native-instruments.de/ -> D-10997 Berlin -> Tel. +49-30-61 10 35-430 -> Germany

I love my Polycom IP600s. However, I'm not clean on how the status setting on the phone impacts the behaviour of *. Anyone here have the details? When the phone is set to online and I'm on a call then * routes a second incomming call to voicemail playing the busy message. If I simply leave my desk and the phone remains online then voicemail plays the unavailable message. I would have

After emerging some updates this morning asterisk 1.0.3 fails to start I get the following errors: ..Jan 6 00:39:24 WARNING[28998]: chan_zap.c:765 zt_open: Unable to specify channel 1: No such device or address Jan 6 00:39:24 ERROR[28998]: chan_zap.c:6197 mkintf: Unable to open channel 1: No such device or address here = 0, tmp->channel = 1, channel = 1 Jan 6 00:39:24 ERROR[28998]:

When I do a 'asterisk -vvvvvc' I get following, but asterisk does NOT stay up: == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found

Hello! Asterisk "CVS-HEAD-08/06/04-14:55:13" won't start on two of three different Gentoo machines. This is the output of gdb: ultra asterisk # gdb /usr/sbin/asterisk GNU gdb 6.0 Copyright 2003 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions.

Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue "myqueue" In extension file: Queue(myqueue|t|||120) And my agents are joining in following

Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? In particular I am interested to hear how many calls could be put through 128Kbs satelite link simultaneously? Ta SJ

Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Regards Bilal ____________________________________________________________________________________ Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545469

I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:1234@213.137.73.178/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -----Original Message-----

Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

In United Kingdom, we have time based dialling pricing from most of Telco's based on time the call is placed! It is called PEAK (08.00- 18.00 Mon-Fri), OFF PEAK(18.00-08.00 Mon-Fri) and WEEKEND (all other times! Could someone from any of other countries let me know if time based charging exists in your country? Also, what numbers (up to 999) are commonly used for emergency, police or other

Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark

Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Our telecom provider (your communications) gives us monthly itemized bills that list

Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with "1" in order To successfully make a call to other USA destinations? ---- I have not been to USA (yet) :) Ta SJ

Hi, all This is certainly very good news! http://www.neowin.net/comments.php?id=21119&category=main

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News -

I enabled the logs on the mysql database and ran : realtime load queues name cou0002-test in the mysql log I can see that the proper select statement is being executed: 2018-12-04T16:29:27.253094Z 229 Query SET SESSION TRANSACTION ISOLATION LEVEL READ COMMITTED 2018-12-04T16:29:27.254384Z 229 Prepare SELECT * FROM queues WHERE name = ? 2018-12-04T16:29:27.254902Z 229

hi all anyone that've heard of any working IAX hardphones yet? roy

Hi, I have enabled ztdummy in order to have * compile it. I can modprobe ztdummy with no problems. The sole reason, i need ztdummy is to heve musiconhold and meetme working. However when I start *, it says this and does not start. ---------------------------------------------------------------------------- ---------------------- == Parsing '/etc/asterisk/zapata.conf': Found