similar to: How can I get support about Dialogic hardware

Please rest assure that I have been following the * development for a while and understand the value the Digium hardware gives me vs any other vendor. Most of the people on this list probably know whats good for everyone else, but I like to find out for myself (I am not a CNN junky). Now the * site mentions Dialogic as supported hardware at: http://www.asterisk.org/index.php?menu=hardware It

We've changed E1 providers and I'm trying to reconfigure an E400P to make it work with the new lines. They're supposedly "standard" EuroISDN lines (in the UK). I'm initially just trying to get a single line up. I have the following in /etc/zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk The LED on the back

Hi people. The latest version of my Ethereal plugin for IAX2 is now available here: - http://almaw.com/ethereal-iax2-plugin-0.3.zip A screenshot showing what you're missing is here: - http://almaw.com/ethereal.png The new version adds the following features/bugfixes: - Decomposes the CODEC fields for supported CODECs, complete with nice English descriptions. This gives you a

Hello, I've search through all of the lists and cannot find any descriptions of live monitoring (monitoring a phone call going on between an extension and a zaptel channel live from another extension while the monitoring phone is muted). I am aware of the monitor function which is actually a call recorder, but I'm looking for live monitoring from a muted extension. is this easily

Hi. I'm trying to do some custom call logging, and I want to call an AGI script from a hangup handler to log call durations and things. Although the script executes, it isn't retrieving variables from the AGI interface. Looking closer, I realised the variables are actually getting unset before the h extension is reached. [foo] s,1,SetVar,foo=bar s,2,Play(audio/a-long-prompt)

Lets say I have an * at my business, with 7960 SIP phones. All the sip phones are registered using their extension number (like 305), but I would also like to put my SIP URI on my business card and in a name format, not an extension number (like lee.goodman), so that the SIP URI would read lee.goodman@asterisk.company.com. How would I set this up in extensions.conf? I got

Hi, I have * set up as a PSTN->VoIP gateway (with an E1 with multiple numbers pointing to it). I'd really like to be able to dial out to a SIP server like so: exten => _X.,1,Dial(SIP/${DNID}@hostname) I.e. the remote SIP server receives a SIP INVITE with a "To:" header containing the dialed number (e.g. 02085555555@computer.company.com). This is equivalent to having a

hi see below's conversation. it seems the e100p card doesn't work with BT. Any idea how this'll work against Telenor (norway)? roy <RoyK> does anyone know if I can trust the E100P to do full PRI stuff in .no? <cypromis> dunno about no <cypromis> I cannot use it in UK <cypromis> cause the framer has problems with system-x switches at bt

As mentioned on the devel list earlier today, I'm interested in writing an IAX2 plugin for Ethereal to make debugging IAX protocol implementation and simultaneous calls on normal networks easier. Anyway, I started work on it this evening, so it's not complete yet, but it's starting to look quite sensible: - http://raq626.uk2net.com/~al/ethereal.png A couple of people have

Beginner: How can a script be called after a calling user hangup? What's wrong with this: [incoming] exten => s,1,Playback,welcome exten => s,2,Record,msgfile:gsm exten => h,1,Goto(callscript,1,1) [callscript] exten => 1,1,Wait,5 exten => 1,2,System("SomeScript") Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL:

I can't seem to hide and/or set my caller ID from *. I'm using a quite recent (three weeks or so) CVS with an E400P card. I have pridialplan=unknown in zapata.conf and I'm based in the UK. The relevant bit of pri debug looks like this (reformatted to fit 80 char width): > Calling Number (len= 4) [ Ext: 0 > TON: Unknown Number Type (0) >

Hi everybody I've just installed a new Redhat 8.0 and configured it with Asterisk, zaptel and libpri. Afterwards I installed a TE410P and configured this as well. But when starting Asterisk I get the following error message: ------------------------------------------------------- -- Registered channel 1, PRI Signalling signalling ..... -- Registered channel 15, PRI Signalling

Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,

Hi all, I am trying to write a program that sends SIP requests to asterisk. My aim is to make asterisk record as many voicemails it can at a time. The design of the program is like this: There are two processes: One main process and a child process (No flames pls. I have very little idea about pthreads and dl modules) The main program asks the user to input the number of test instances. When

Hi, Is there IAX client in Applet JAVA which can be embeded in a web page ? Best regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031208/c388ef61/attachment.htm

Hi all, I'm about to start setting up my first asterisk/cti system in our test lab. I've read through all the documentation I can find and relevant posts in the list archives but can't seem to find anything explaining how to go about initiating an http request upon an incoming call. I basically want asterisk to request an uri on our intranet, which will pass call details to our

Does anyone know of a Java based SIP client and if so have has anyone used it. I found JAIN at https://sip-communicator.dev.java.net/ but have not tried it yet. Rgds, Stuart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030818/ea1e2717/attachment.htm

Hi, Just need to check somthing.. Am I correct in saying that conferencing does not work on a system that does not have a Digium board installed?? Thanks.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hello All, I wonder is there a way where I reload asterisk on CLI without disconnect any call that is currently taken place. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030813/41f0a4ca/attachment.htm

Hi everybody. We have a problem with the configuration of the card, the cards work and we receive incoming calls but asterisk don't receive dnid. We have 5 servers with 1 E400P with the same problem and the telco told us that we need to configure the card to request it, how we can do this? Can you help me to solve the problem. Best regards, Carlos Fernández Puente carlos.fernandez@alisys.net