search for: maw

...Putting a 1 in the timing field makes no discernable difference. The guy who tested the lines isn't very clueful - he just plugs in his gear and puts a tick in the box, but he's pretty sure it uses hdb3 coding. CAS framing makes the LED go RED. Anyone give me a clue here? -- Alastair Maw <al.maw@mxtelecom.com> MX Telecom - Systems Analyst http://www.mxtelecom.com

...nsion in sip.conf with a dynamic username of ${DNID}. How does one achieve this? Likewise, it would be nice to be able to use gnophone to simulate calls into the system, by pointing it at the * box and getting the dialed number on that to route things in the same way. Any ideas? -- Alastair Maw <al.maw@mxtelecom.com> MX Telecom - Systems Analyst http://www.mxtelecom.com

...ion, but from a normal one instead. Does anyone have any idea how I might fix or work around this? It's important for us to log call durations (and other things), which obviously needs to be done when the users hangs up. Storing stuff using the cdr isn't really an option. -- Alastair Maw <al.maw@mxtelecom.com> MX Telecom - Systems Analyst http://www.mxtelecom.com

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 list...

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

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

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

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

...eal.png A couple of people have e-mailed me to say that they're in a hurry for such things. If you'd like me to e-mail you a copy, give me a shout. Otherwise, I'll polish it up and give it to the Ethereal guys for inclusion in Ethereal 0.9.17 or something. :) Regards, -- Alastair Maw Systems Analyst http://www.mxtelecom.com

...that I don't want to have to do. Ideally, I'd like to be able to set my callerID to an arbitrary number. If I set pridialplan=national/international I can't work out what format the outbound calls numbers should take and get denied messages back. Anyone have any ideas? -- Alastair Maw MX Telecom - Systems Analyst www.mxtelecom.com

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:

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 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

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

...;s voicemail (based on the incoming DNID). Is any of this possible? Phone cost per handset should be as low as possible, as per usual. :) I have no experience of SIP hardphones, so don't know how much/what information about the call they're capable of displaying. Regards, -- Alastair Maw

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

...ng the hundreds of simultaneous calls required in a server environment, although there'll be no reason not to use it for IAX clients too. Obviously such a library would enable a nice GUI cross-platform IAX(2?) client to be easily created, which would be a nice by-product. -- Alastair Maw MX Telecom http://www.mxtelecom.com

Hi all, forgive the question but is it possible to use PCI modem cards (aka winmodem's) as FXO/FXS ports in * ? what about external modems like the USR Sportsters? Thanks in advance, Bryan. Bryan Nolen Lead Developer http://Arc.Net.AU http://cdonline.com.au