search for: atopo

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, 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, 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, I'm playing around with the queue features, and I'm looking for a solution to indicate the agent status (logged on / logged off) on the phone (Grandstream GPX2000 or SNOM 190) It would be nice to use one of the busy lamps... thanks -- Alexander Topolanek Intelligente EDV- und Telekommunikationsl?sungen Montage von Sicherheitstechnik Probst Peitlstr. 85 2103 Langenzersdorf +43

Hi, I'm using Grandstreams as the agents phone of a queue. Attended transfers in a normal situation (direct call to the extension) work fine, but when the agent has a queue call and tries to transfer it to another sip extension the called party is hung up. Transfer is to pick another line on the Grandstream, dial the other extension, wait for a pickup, and then press the transfer button +

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 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

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, 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:

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