similar to: Dialing Delay

I have a Sipura 2000 working fine, but whenever I dial any extension there is a delay of 5-10 seconds before it starts ringing. However, if I dial the extension and hit the pound key after the number, it goes through right away. Is there any way around this?

Damien, Here's a forwarded bug for you. Cheers, Phil. --[[message/rfc822]] Subject: Bug#52414: ssh-add uses ssh-askpass, but ssh doesn't Reply-To: David Huggins-Daines <dhd at plcom.on.ca>, 52414 at bugs.debian.org Resent-From: David Huggins-Daines <dhd at plcom.on.ca> Resent-To: debian-bugs-dist at lists.debian.org Resent-CC: Philip Hands <phil at hands.com>

Hi I have a simple question but I cannot find the answer. I have a line with 2 different phone numbers I want to redirect each phone number called to a different IP phone Example Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2 Someone calls 5551235 and the call is redirected to IP phone 192.168.0.3 Thanks Patrick

I have some identical windows 7 clients. All connected to Samba 3.5.4-1.1-2382-SUSE-SL11.0. If I'm the windows-owner of the file, swat shows me under "oplock" "exclusive+batch". Other users get: "none". What is the reason for this? ___________________________________________________________ Neu: WEB.DE De-Mail - Einfach wie E-Mail, sicher wie ein Brief! Jetzt

Hi All, I am trying to reproduce this using R instead. [image: Full-size image (38 K)] I tried using the following code *SChla <- read.csv("SM_Chla_data.csv")* *Atlantis <- SChla[16:66,]* *head(Atlantis)* * * Seamount Station Depth Pico Nano Micro Total_Ch dbar Latitude Longitud 16 Atlantis 1217 Surface 0.0639 0.1560 0.0398 0.2597 2.082 -32.71450 57.29733

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I would like to know the blocksize of a particular file. I know the blocksize for a particular file is decided at creation time, in fuction of the write size done and the recordsize property of the dataset. How can I access that information?. Some zdb magic?. - -- Jesus Cea Avion _/_/ _/_/_/ _/_/_/ jcea at

I was wondering if there was a way of setting up the dialplan in a way that if you dial an extension that is NOT in the dialplan then it would play a not-in-service gsm file and then play congestion tones. I would rather like this better than just hearing a busy signal on my phones.. I DID search around on the wiki and using google and could not find anything. Thanks. -- Stephen Rosebush,

Of course I am not a kernel expert, so .. please be patient. I am investigating on my zaptel/zapata problem. As the main error message asterisk quits on mentions <'/dev/zap/channel': No such file or directory> I went peeking over there. [Asterisk Verbose Error Mar 13 20:43:35 WARNING[5779]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or

Hi, I have recently upgraded our web proxy to CentOS 5.6. Before the upgrade, squid was running fine. Since the upgrade, sometimes, squid crashes: > [root at stargate ~]# service squid status > squid dead but pid file exists Looking at log files, I think it could be related to logrotate stuff, > -rw-r----- 1 squid squid 42713088 May 3 04:02 store.log > -rw-r----- 1 squid squid

Hi i need to implement "Inward" SIP usring dialing in my Asterisk IPpbx, So anybody can recah me by dialing my SIP uri. same time my DNS on same server where currently Asterisk running. how ican implement this. Please help me with config details at DNS & Asterisk point of view. anybody can provide me config exmple? I am using Asterisk 1.4.9. Plz help me Regards Amit

Hi - I am trying to find a Linux based solution for one of my clients, I want to bond two adsl lines into one, with redundancy (if one fails the other does all the work). I''ve been looking into ECMP (Equal Cost Multipath), I know BGP would work, but something like ECMP would be much simpler. I have full control of both ends of the connections and I can have both ADSL lines terminate in

I am attempting to install Dahdi on a virtual machine running Centos 5.5 and having various problems. yum install kernel-devel gcc make gcc-c++ libxml2-devel Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirror.optus.net * extras: mirror.optus.net * rpmforge: fr2.rpmfind.net * updates: mirror.optus.net Setting up Install Process Package

deeper still looks like culprit was fts_enforced = "no" which i removed given default anyway .. still get error [root at stargate dovecot.rawlog]# doveadm search -u matt at the-bryants.net mailbox .EXPUNGED/* from noreply at apress.com doveadm(matt at the-bryants.net): Error: fts: Failed to sync mailbox .EXPUNGED: Mailbox doesn't exist: .EXPUNGED 6156a416748c035b5c0b0000ef794dd1

2013/12/9 Fabio Cecamore <ceca_89 at hotmail.com>: > and the UPS don't wake up after the whiteout.. 2013/12/6 Fabio Cecamore <ceca_89 at hotmail.com>: > desc = "AtlantisLand Host Power 851+" Is your UPS an 'Atlantis Land Host Power 851' or an 'Atlantis Land One Power 841+'? The first one should be using a slighty modified version of the

Is there any additional documentation/information around the new stats module. Have added some metrics just to see what they produce ## ## Metrics ### metric imap { ??? event_name = imap_command_finished ??? #source_location = example.c:123 ??? #categories = ??? fields = name args running_usecs bytes_in bytes_out ??? #filter { ??? #??? field_key = wildcard ??? #} } metric sql { ???

Hello! Port security/portaudit reports the following problem: Affected package: FreeBSD-491000 Type of problem: multiple vulnerabilities in the cvs server code. Reference: <http://www.FreeBSD.org/ports/portaudit/d2102505-f03d-11d8-81b0-000347a4fa7d.htm l> Note: To disable this check add the uuid to `portaudit_fixed' in /usr/local/etc/portaudit.conf I have 2 related questions: 1)

Preface, running redhat 8.0 (pretty standard and up2date) [atlantis 158] uname -a Linux atlantis.cs.umd.edu 2.4.18-27.8.0 #1 Fri Mar 14 06:45:49 EST 2003 i686 i686 i386 GNU/Linux [atlantis 159] gcc --version gcc (GCC) 3.2 20020903 (Red Hat Linux 8.0 3.2-7) Copyright (C) 2002 Free Software Foundation, Inc. This is free software; see the source for copying conditions. There is NO warranty;

hi, I've been running a pure sip asterisk box for a while now with no problems, and i've recently added an isdn2e line from bt in the uk. everything is hooked up and i've got it ringing my sip extensions, but the logs don't quite look perfect and i can't see any description of what i should consider to be normal behaviour. would someone be able to look this over and tell me

OK I think you need to read previous posts on this. I?m not looking for any other command. > On May 14, 2019, at 5:10 AM, John Hodrien <J.H.Hodrien at leeds.ac.uk> wrote: > > You misunderstand. su behaves the same when switching to root as to any other account. > > su - > > is probably the command you're looking for. Cheers, Bee

Hi, I have installed two X-Lite phones and they're able to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX.