similar to: OT - snom phone question

Displaying 20 results from an estimated 3000 matches similar to: "OT - snom phone question"

2009 Apr 14
2
What means? Correct auth, but based on stale nonce received
Hi masters! I've this Asterisk 1.4.15 running. yesterday I had to change the firewall schema that I had before. I use to have a FW that would be my network FW/Proxy and do the NATs for Asterisk. This FW was receiving too many requests from my LAN and it was making the Asterisk 'cut' the calls or reach very high latency. Yesterday I added a new FW just for this Asterisk. The same
2005 Mar 29
3
help w/ basics
Hello, I am new to Asterisk and new to this list. I got Asterisk setup and running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone to test out. I cannot get the phone to talk to the Asterisk box. On bootup of the phone, it tells me that it cannot contact boot server. Why is that? It gets an IP fine, and I have also tried manually setting the IP of the phone and the Asterisk
2006 Mar 13
1
Shocking news: Yum update repo server runs on Mirosoft IIS?
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> <title></title> </head> <body bgcolor="#ffffff" text="#000000"> Hi everyone,<br> <br> Look what happens when do yum update on one of my
2009 May 26
2
Domains
Hi, I'm trying to understand an issue I'm seeing between two Asterisk servers. I think it has to do with Domain definitions. Server A), has extension 5550 defined. Has a sip client 2000 defined, and has guest-invites enabled. Server B), Dials to server A for any 5550 dialled. Has sip client 2000 and 2001 defined. If I register at server B as client 2001, and dial 5550 then
2009 Feb 07
1
Tatil yanıtı
<p class=EC_MsoNormal style=""><span lang=EN-US style="font-size:12pt"><font color="#000000"><font face="Times New Roman">Dear friend:</font></font></span></p> <p class=EC_MsoNormal style=""><span lang=EN-US style="font-size:12pt"><font face="Times New
2013 Jun 20
1
[LLVMdev] ARM struct byval size > 64 triggers failure
> - "since ABI says the stack pointer needs to be 8 byte aligned at function entry point" (taken from Manman's reply) > What will be considered as entry point here? > Is it place of SP Adjustments "sub sp, sp, #16" > (Or) Is it place of first user instruction(end of prologue) "ldr r2, .LCPI0_0" Eight byte stack alignment is a
2004 Feb 11
1
MCD-Estimator in R
Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable X-Spam-Checker-Version: SpamAssassin 2.63 (2004-01-11) on hypatia.math.ethz.ch X-Spam-Status: No, hits=0.7 required=5.0 tests=BAYES_30,HTML_MESSAGE,MIME_HTML_ONLY autolearn=no version=2.63 X-Spam-Level: <html><style>p {margin: 0px}</style><body bgcolor=3D'#ffffff'
2011 May 14
36
LiveZilla
Hey there I've been using wine for various application and I'm wondering if anybody got enough luck working out livezilla in wine I've tried everything possible core fonts and netframework 2.0 Is it possible to run it under wine at all? what about netframework 3.5?
2008 Jan 22
9
VNIC, non-global zone, dhcp & dns
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1"><font face="Bitstream Vera Sans"><br>     Hello,<br> <br>     I am trying to have dns <b>automatically</b> configured through
2007 Nov 02
1
Jitterbuffer issues
2009 Apr 29
5
What do I need to connect landline calls without telephony hardware?
For some reason, I have been unable to find the answer to this online or in books... I want to have a "click-to-connect" feature on my website where the user enters their phone number and then my asterisk server calls their phone and my phone and connects the two calls to each other. All I have are: 1. A Server 2. A DSL connection 3. A Router and DSL Modem 4. A static IP What do i
2006 Mar 31
1
Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding.
Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again - but only one direction. -David ________________________________ From: asterisk-users-bounces@lists.digium.com
2011 Apr 12
8
GUI Software Raid Monitor Software
2009 Apr 14
0
How are you doing recently?
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etc........Please take some time to have a check, There must have something you'd like to buy. Their contact email: myewell at vip.188.com Hope you have a
2015 Jul 02
2
asterisk email to fax
2001 May 21
1
printing font scale issues
I'm using Wine-20010510 to run Quickbooks. Printing seems to work okay, as long as I use the "default" forms. When I try to modify the defaults, by changing the fonts on parts of forms, the printed result is that the font is tiny. For example, 18-pt Helvetica looks printed out like 4-point Helvetica (or maybe even smaller!) Is there something in the configuration that I
2008 Mar 01
7
ASTCC installation error install: invalid user `apache'
I am attempting a fresh install of ASTCC on Ubuntu. Getting install invalid user as bellow. Has any one seen this? Can some one help out? /usr/src/astcc# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi >/dev/null Detected dry run! ./astcc-admin.cgi >/dev/null DBI
2006 Mar 16
9
Baffling AJAX issue
Ok, I''ve got some link_to_remote stuff working elsewhere just fine, but this time it''s doing something very screwy. Here''s the view code to create a link to remotely toggle a boolean value on a record: <td class="contact_active"> <%= link_to_remote "<div id=''contact_active_#{cm.id}''><span
2010 Oct 30
1
Tormenta 3 (Tor3e) - Driver.
2015 Jun 27
4
Branch based on call volume
Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150627/6774c750/attachment.html>