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...tel for about a week, and every few days, I find myself sitting at the DID menu clicking the link that reads "Click here to get a random toll free number". I have three toll-free numbers now, and I don't know if it will stop... Is there any hope for me? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/

I know this has been covered more times than to mention and this is where I got most of my info from... But I am having issues with this. I can't seem to get the phone to register with *. This is being tested on a internal network right now. Here is the setup - sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

We have a test server that runs a single PIII 500MHz(256MB RAM) under Slackware, and we can get 12 SIP -> Zap calls running on it just fine. Over that and we have seen intermittant errors like call quality and very high load spikes. You should be able to get at least 24 SIP -> Zap on that setup. Post on the list when you do max it out. It's always good to see capacity specs. MATT---

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

I just booted Xen unstable (2.6.16 kernel) and Domain0 is running fine, all on Debian 3.1 Sarge unstable on amd64. I can''t do much else: Error: Error connecting to xend: Connection refused. Is xend running? What''s wrong and how do I fix it? -- Neal Probert (nealzilla@probestar.com) aka Niall Lundkvist

Can someone tell me what this setting is supposed to be? peer->nat = globalnat; It looks like it's inheriting a parameter, but I'm curious, is globalnat an option that we're supposed to set(or let default) in sip.conf? ----- Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise.

--On Thursday, February 03, 2005 2:20 PM -0500 Andrew Thompson <asteriskuser@aktzero.com> wrote: > What you are seeing with these bargain providers is they have a clause in > their contract that says they own the number, not you. It is a lock, and > it ought to be illegal, but sadly, it's probably not. If you choose one > of these companies that doesn't allo...

...act number off their website for a while, I guess hoping I wouldn't be able to contact them. They never once responded to any of the telephone calls I made to them. I just signed in and see that they didn't ever refund my payment and close my account, either. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/

...kallsyms matches Binary file usr/gen_init_cpio matches On google, I found a reference to a possible gcc version incompatibility, and a couple of buzilla reports about linux-2.6.11. For now, I''ll be recompiling gcc and trying again. Any pointers appreciated. -- Andrew Thompson http://aktzero.com/ _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

...gui(Cpanel/WHM) only allows the following options for entry type: A6 AAAA CNAME MX NS PTR TXT WRK Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? ----- Andrew Thompson http://aktzero.com/

...well. I would prefer that there be no other intermediary involved in call setup. I have a SPA-2000 at home, but I couldn't find anything in the setup for doing this. If anyone believes that the SPA can do this, please provide me with configuration info. Thanks. -- Andrew Thompson http://aktzero.com/

Does any one know what happened with voice conduits? I have been trying to reach them for nearly three weeks now. Their voice mail boxes are full and writing email to them does not get any returns. Thoughts or sightings are appreciated. -- R.J.

Hi all, Thanks to Steven Sokol great work, the IAX2 bug in DIAX is now solved. For the interested people, you can download the new DLL (just the IAX2 version) from the following location: http://www.laser.com/dante/diax/wiax2.zip Replace the wiax2.dll file in the app directory with the new one and this is all. Please test it and send me your feedback. I intend to release a new DIAX version this

All: I am trying to use Voicepulse as my incoming line and want the caller to simply dial the extension of the party they want to reach. Here is my problem: - the first time they dial it works fine and I see the following on my console Aug 24 23:14:31 DEBUG[-1126876240]: chan_sip.c:4408 build_route: build_route: Contact hop: <sip:6035057098@66.234.228.137> --

Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>

Dear all, I'd like to know if there is a way for multiple asterisk servers to share a common SIP and/or IAX registry. The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com

Hi all, A new version (0.9.6) of DIAX is available for download at: http://www.laser.com/dante or http://www.geocities.com/tdanro There are no new functions, but some bugs fixed: What's new in version 0.9.6: - add Default_user locales as new language. The program language can be automatically selected based on default user locales on your system. You still can manually select the language,

Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing

Is it possible to disable one of the (real) Ethernet interfaces from Xen so that Domain 0 doesn''t see it? I.e. when the Domain 0 XenLinux boots up it only has eth0 although there are actually two NICs in the machine. Would it be possible to do this using the "nics=" parameter? I know this can be done for doms>0 but get the impression Domain 0 is different. Alternatively