similar to: SIP URI Dialplan?

I was at VON in Boston today and saw on the Digium stand a Cisco 7960 with a picture of Tux and the Asterisk log on its display. I WANT IT!!!!! Anyone know where I can download this file please? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Okay sorry for the empty message bodies earlier problem with Outlook, I guess is you have digital IDs as I do and they expire, even if you override the signature option it clears all sent text....windows...mmmmph.... Anyway, what I was trying to relay to the group was that ZyXEL has come out with a new Firmware (March 21, 2005) for the Prestige 2000W WiFi SIP Phone. In my opinion it has made

I googled on this for about an hour and the most relevant hit I got was, of course, the first hit: http://www.sowerbutts.com/linux-mac-mini/#support In it, he indicates that the stock Bluetooth module "should work, but untested" - he doesn't qualify the statement with anything. Has anyone tried chan_bluetooth or even the Bluz stack on a Mini or a G5? If so, under Linux or OSX?

Hi, lets say I have a a table, "Linkings" that I am using for polymorphic associations. One of the associations is to the "person" table, and there are several other associations. "Person" has 2 types, "slacker" and "suckup". It is modelled as such: class Linking < ActiveRecord::Base belongs_to :person belongs_to :linkable, :polymorphic

Hi, First of all, apologies if I'm not posting this in the correct place. I'm successfully running Microsoft Visual Foxpro 7 under Wine, however I needed to use the native versions of shlwapi and oleaut32. There is a major problem still, however, and that is that foxpro will freeze as soon as I use the mouse to click in certain controls (the command window, a textbox, an edit box,

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

?for doesn't return anything help.search("for") doesn't return anything- Is the for loop so prevelant in computer programing that the documentation is implicit or is R paradigm to discourage the use of the for loop. I will post data probably tonight, but here is my problem. I have preformed an MDS on a set of data. I have the scores of the four axes that are the optimal

Hi, I can't get dial-out working. I'm trying to call 523936. Is there something wrong with my setup here? Could someone please give me a few pointers? Regards, Norman Zhang [fwd-out] exten => _8.,1,SetCallerID(${FWDUSERID}) exten => _8.,2,SetCIDName(${FWDUSERNAME}) exten => _8.,3,Dial(SIP/${EXTEN}@fwd,70) exten => _8.,4,Macro(fastbusy) exten => _8.,5,Hangup *CLI>

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

Hello all, I have a new Sipura SPA-2000 that I am trying to configure beind a NAT. The SPA is able to register to the asterisk server without a problem and the SPA can make calls to other extension that are not behind a NAT. However, when I try to call the SPA from another extension, the extension connected to the SPA rings, the user at the SPA answers, and there is no audio in either

Hello, I'm pretty new to asterisk (only been fighting with it on and off for about the last month), so please go easy. I've been wrestling with the documentation, forum posts, google, and my lack of telephony and VOIP knowledge, trying to get my setup to work. My current problem has me stumped saying, "There's gotta be a cleaner way to do this." Please show me the light

Hi, following the discussion about acts_as_ferret on the Rails mailinglist, there was an issue about transactions, which could result in beind the database and ferret out of sync. I have taken a different approach from acts_as_ferret trying to resolve the transaction problem. Instead of adding things to the ferret index in the model, I have added it in the controller. I have only the create part

In that example you could make outgoing calls only correct? (since incoming likely needs port forwards) I guess the questions becomes "how often are you going to do that to justify the extra $100 or so you going to pay for a wifi sip phone?" Paul Fielding (paul.fielding@shaw.ca) wrote: > > I think some people are missing the point. You can't throw your cordless >

Hello Asterisk-users, I'm a long time reader, first time caller. I've recently set up an Asterisk network in my house that I eventually want to use to support my small business. I bought a Sipura-3000 and I'm currently running Asterisk on a server I was already using. Everything is working nicely. I can make and receive calls. (important step) I have a basic IVR. The voicemail

Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000 May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian

Hi all, I wanted to let you know that I feel I need to step down as leader of the Virtualisation SIG. When I originally was approached by KB to do this, it was always clear that this would be a temporary thing until the SIG got going. At the time, only Xen and CentOS folks were involved with the SIG. Since then, we added oVirt and Docker maintainers, have a total of 5 maintainers and built good

who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list

At first I thought it was due to running Mangler simultaneously, but recently we've had no sound in World of Warcraft. One night when I was setting up to raid that night, I started WoW, and sound worked fine. I then quit WoW, started Mangler, started WoW, had sound in the first but not the second. When WoW started this dialogue box popped up saying that a change in hardware had been detected

Le 17/11/2019 ? 18:56, Jonathan Billings a ?crit?: > You should never be using ntpdate anymore (which is why the ntp project is deprecating it, http://support.ntp.org/bin/view/Dev/DeprecatingNtpdate <http://support.ntp.org/bin/view/Dev/DeprecatingNtpdate> ). I really only ever suggest ntpd unless you?re running an NTP server that provides NTP service to your network, and needs to

I note with interest that Kerberos support is now available (for the version 1 protocol, at least) in OpenSSH 2.9.9p2. However, it does not build with MIT Kerberos, due to the usual Heimdal/MIT library differences. These look, by and large, like the same problems I encountered when porting Dan Kouril's patch to MIT Kerberos - so I'm having a go at fixing them (my GSSAPI patches need