similar to: Indicating agent status on the phone

Hi, perhaps someone has an answer for me: - voicemail isn't sending any mails, even SMTP_SERVER and SMTP_Domain are configured in rc.conf, and the mail address ist configured in voicemail.conf. - On my grandstream GPX2000 VM light flashes when there is a new voicemail, but it doesn't go out after I have deleted the message best regards -- Alexander Topolanek Intelligente EDV- und

Hi, recently I changend a few things in the configuration of the Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that different groups of SIP-Phones are using different trunks to the outside worls, so I moved some of them to a Support context. However, dial out from this phones failes as they're still looking for an extension in the default context, which doesn't

Hi, does anyone know about a plugin that allows dialling a contact from the phonebook of evolution or T-bird? -- Alexander Topolanek http://www.topolanek.at

Hello, I'm running several asterisks in a carrier environment. The asterisks do mainly gateway business between E1 cards and IAX with some routing logic. On one key server I see issues of "Auto-congesting call due to slow response" coming every number of calls. The IAX peer is in the same subnet, the servers are not really loaded. Versions in use are 1.2.2 and 1.4.23-rc3, with rsa

Hi, I'm using a Junghanns Quadbri ISDN card on some lines from the Austrian Telekom. Things are working, the only missing stuff is to add a "0" as a prefix to each incoming call, to make it possible to answer missed call lists. I'm using the 0 as the prefix for outside lines. I've experimented a little with the prefix settings in zapata.conf, but without success:

While using Blindtransfer #Extension everything works fine. But how do i activate announced transfer with an Grandstream GPX2000 ? Greets Markus

Hi, I would like to set up a call center with free seating agents. However I would like to indicate the agent status somehow on the terminal, to tell the agent if she has been logged out due to non-answer. Does anyone has a good idea how this can be achived? best regards -- Alexander Topolanek http://www.topolanek.at

Hi i have two problems with my Grandstream GXP2000 : 1- When i wan pickup a call, that's don't work's (*8EXTEN) and when i test whit Softphone, i have a error too, he say me *8201@MyContext not found .. in features.conf, i have: [general] parkext => 700 parkpos => 701-720 context => parkedcalls

Hi, im migrating existing kvm-images into libvirt domains, by creating a domain in virsh from a xml file. The domain starts, is working fine, but after shutdown it is lost, and I have to recreate it. BTW, can I reset the domain count somewhere? Any ideas? thx and kind regards Alexander virsh # version Compiled against library: libvir 0.4.6 Using library: libvir 0.4.6 Using API: QEMU 0.4.6

Sorry, I know this is not really libvirt related, but I'm searching for a solution sind about an hour and cannot find anything that helps me ... Ok, I used to install a standard debian in a kvm instance, and removed the virtual cd-rom after installation. Now I'm getting this triplet every five minutes in my syslog: Jan 26 19:25:12 srv62 kernel: [ 1324.267675] hdc: task_in_intr:

Hi, Perhaps there's something wrong in my config... I did some tests connecting Asterisk to an Ericsson MD110 PBX by setting up an h323 trunk. When dialling into asterisk I got some problems when the entire number is not in the setup message, i.e. I'm dialling digit by digit on the ericsson phone. Lets say I have 4001 in my extensions, and dial that from the Ericsson PBX, then the

Hi, I had a working solution to display an agent status on the BLF of a Grandstream. Then I made some minor changes to the numbering (changed the extensions from 8X to 6X, and the Agent numbers from 978X to 976X), and out of the blue grandstreams do not register for an Angent status. This is the relevant line in my extensions.conf: exten => 41,hint,Agent/9761 and I see the hints, but no

I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a tool on the GS website to generate

Hi, all-- What do you make of this? Here's my call log--looks like there are a lot of calls going in and out of the server that are not real incoming or outgoing calls. Does anybody have any clue what is happening? 2006-11-14 16:41:00 Local/8183... 8183461773 "8183461773" <8183461773> 8183461773 NO ANSWER 1 47. 2006-11-14 16:40:59 IAX2/Voice... 8183461773

Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ---------------------- Fabrice Tereszkiewicz Sawadka.org

Hello R users, i'm wondering how to trigger variable creation. Whenever a variable is created i want my own function myFun(...) to be started. if (exists("x")) {rm(x)} # after removal of x # any of these calls x<-10 # should call myFun x=10 # should call myFun assign(x,10) # should call myFun etc.

hi all, i want to use a compaq sdlt 320 under centos. the tape drive was already installed and after rebooting i was not able to see it(dmesg | grep tape or dmesg | grep scsi). i add the correct module (modprobe st) but i'm not able to use with mt. the st0 file is missing, can someone help to resolve this issue. thanks ?

ok, sorry, my mistake was the C printf. Thank you for your good answer Regards ----- Message d'origine ---- De : JS Ubei <jsubei at yahoo.fr> ? : jim holtman <jholtman at gmail.com> Cc : r-help at r-project.org Envoy? le : Jeudi, 17 Juillet 2008, 15h25mn 07s Objet : [R] Re : float and double precision with C code thank you for your quick answer, I'm far of the digits

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "user1" calls user "user2" 2. "user1" then presses the feature code "*0" to redirect "user2" to conference room 300 3. "user1" then dials the user "user3" 4.

Author: praks Repository: /hg/zfs-crypto/gate Revision: 4a9dff71a42bacf7232dd2d8dbcec1db3e6c2ccc Log message: 6402576 Sometimes port_getn() never returns indicating that a socket has become writable Files: update: usr/src/uts/common/fs/portfs/port_fd.c