similar to: Thank You All

Our basic system is as follows: P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS several weeks ago, working OK for routing, VM, and AA, calls in on separate PSTN lines to Adtran TSU 600, into * server through T100P card. The hardware is not taxed at all with little over 20% proc utilization ever, low mem use, etc. All Phones are SNOM 200's with various firmware revisions

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Question for other snom 200 users: 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragments of audio. Any ideas? 3. Initially we have horrible introduction of background noise into the

This is not specifically related to * but * is the software I'm using so here goes... Does anyone have the correct file set for a 7960?? I've been trying to get the release 6 SIP load on one I have without any luck. The phone keeps getting the same 2 files from the tftp server and starting over. If you have the files - other than the POS30600.bin which I know is licensed - could you

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

I have an installation which is currenly using 14 DID Trunk Lines. I need to be able to use Caller ID information and currently it is not available on these lines. Is there another way to access this information? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/62559e22/attachment.htm

I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented

Thanks we just figure it out a bit ago. It's amazing how simple some things are when you just ask - and then realized that you were making it too hard to begin with!! :-) Lane Hoskins, MCP Network Engineer 540.767.7626 -----Original Message----- From: Jared Smith [mailto:jsmith@drgutah.com] Sent: Tuesday, January 13, 2004 10:59 AM To: asterisk-users@lists.digium.com Subject: Re:

If one of the authors is listening: http://www.asteriskdocs.org/html/apas02.html lists usereqphone 2 times. One of the entries should really be useragent. And the example for usereqphone is wrong. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? ->

-----BEGIN PGP SIGNED MESSAGE----- Just wondered if anyone had any suggestions about syncing up master.passwd files between multiple machines that didn't involve allowing root login remotely? The users need to be able to log in remotely and own files on the different machines. ~~ Andy Harrison ah##@httpsite.com ICQ: 123472 AIM/Y!: AHinMaine [full headers for details] -----BEGIN PGP

I have an X100p card....and it is hard to hear the person on the other end. Should I mess with these values? I have heard both yes and no to this question in the past. If yes, how much louder should I make them? Thanks, MIchael

[this went to me instead of the list; MM, your list maintainer] I am digging in the wrong hole I guess. Where can I find R /S routines for spatial statistics? Also has anyone made an R link to a GIS package? Anyone out there who works in this area? Thanks Richard E. Hoskins WA State Department of Health 1102 Quince Street Olympia, WA 98504-7812 richard.hoskins at doh.wa.gov tel: (360) 236 -

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Here in Sweden, it's supposed to be springtime. A wonderful time of the year, with sunny skies and wonderful weather. Almost summer. Today, it's not. It's winter all over again with rain and only 3 degrees celsius outside. Better to stay inside and write a weekly Asterisk newsletter :-) This week's topics: ------------------- * Looking beyond Asterisk 1.0/1.1 - what's up? *

Excellent. I was using ast-13.1.0 with no luck. I upgraded to 13.2.0 and have made it further, but am having a little difficulty. The outbound-publish object types seems to be working in realtime now. But the asterisk-publication object is only reading from sorcery.conf. I know you said that it *should* work, with no guarantee, which I'm fine with. I just want to make sure I don't

Hello, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. Thanks! Matt Hoskins | NPG Corp | Systems Architect

Matt Hoskins wrote: > Good Morning, > > After further investigation, I found that the res_pjsip_publish_asterisk > module does not use the realtime sorcery wizard, but instead only reads > from the configuration files. I've been able to patch the module, using > the logic from the other modules to learn how to make the sorery > configuration read from the other sorcery

FreeBSD mojo.televoke.net 4.8-STABLE FreeBSD 4.8-STABLE #10: Mon Apr 14 15:48:09 PDT 2003 mike@mojo.televoke.net:/usr/obj/usr/src/sys/MOJO i386 mozilla-1.3_1,2 The open source, standards compliant web browser mozilla-headers-1.3_1,2 Header files for mozilla communicator web-surfboard After stepping up to Mozilla 1.3_1,2 attempting to type in any dialog box (password dialogs in Mozilla

I'm working on an Asterisk system with all of it's PRI ports configured with the LBO setting at 5, like this: span=3,0,5,esf,b8zs As of yet, I am unwilling to change the LBO to 0 to where it probably should be because the system is working and I'm not sure exactly what the LBO does. I'm aware some changes were made to deal with low audio levels. Does anybody know what audio

Hi, I have a data frame with two columns of data, one an indexing column and the other a data column. My issue is, this data frame is incomplete and there are missing lines. I want to know how I can find and add data into these missing lines. See example below ## Example data data <- data.frame(index=c(1:4,6:10), data=c (1.5,4.3,5.6,6.7,7.1,12.5,14.5,16.8,3.4)) index data 1 1