similar to: How to use * and # as part of number in dial command

Hi All, The wiki seems to be struggling this evening. Anyone else seeing this? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262

Hello, I have setup an asterisk server with two EICON cards, a 4BRI and 2FX. How do I know, which card is the first, so that I can setup capi.conf with the right entries? Thanks for your help, Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93

All Any body know of a generic bluetooth adpater for the universal 2.5mm headset jack on a cell phone that supports the wireless profile *NOT* the headset profile I know jabra has the A210 http://www.jabra.com/JabraCMS/NA/EN/MainMenu/Products/Accessories/JabraA210/ JabraA210 but it only support the headset profile .. I am trying to shoe horn my current braindead cell into DocknTalk

Sorry if this not the right place to post this.... BUT... Since May 31st, ALL of these user list messages have been filtered by "spamassassin" running on my Linux box. - Claim to be listed in "Bayes" as spam. - Have no clue why this is happening. Luckily, "spamassassin" sent the messages to the "probably-spam" folder on the Linux box & I was able to

Folks, I am a newbie to the VOIP world and have a question (might as well sound silly to some). I would like to set up a PC-to-Phone call from my desktop to a regular PSTN number. Does the Asterisk PBX itself act as a VOIP-PSTN gateway or do I have to subscribe to a VOIP provider for this? Are there any free IP-PSTN gateways which I can subscribe to? Do I need any specific hardware to set up such

I've got a one-way audio problem, but I've looked through a few documents on the subject and I'm not sure that it's the same issue. User A calls a local Asterisk user B via a public SIP gateway (voiptalk.org) using (sip:110@siptest.dmclub.net) B is connected to the Asterisk server via VPN B is registered (and has successful bi-directional conversations with other users on the

I've tried to find some details on the wiki, but was unable to get a satisfactory result, so I am asking here: I have a Linux (FC3) box with these specs: vendor_id : AuthenticAMD cpu family : 6 model : 3 model name : AMD Duron(tm) Processor stepping : 1 cpu MHz : 797.388 cache size : 64 KB MEM: currently 256, looking to upgrade to 512/768

I have a brand new IAXy I'm playing with. I do not get a dialtone on the phone, or any response at ll on the phone. No sound, no dialing, no ringing. The phone and wire are tested and known to be good. I think I have it setup correctly. When I give the iaxprov command I get this: #iaxyprov 192.168.1.90 iaxy.conf 02: c0 a8 01 5a 05: 11 d9 03: ff ff ff 00 04:

I am a newbee to asterisk. I recently installed asterisk@home. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a

"Unable to login to the MySQL database 'tiki' on 'localhost' as user 'tiki' Go here to begin the installation process, if you haven't done so already. Lost connection to MySQL server during query" Hi, Everyone: What is the URL of the "list archive"? The 'wiki' seems broken. I want to connect to a single home POTS from an internet

Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? It looks as if the PSTN switch accepts the *31* and then hangs up so you can make the NEXT call with the *31* feature enabled. If so I assume the *31* feature will be enabled for the next call on the ENTIRE SPAN if it is an ISDN trunk group. If

Anybody here having trouble receiving email from the list on Gmail? I havn't received anything since Friday July 29.

Hi all, Does anybody of you have the winbond w6692 working with the mISDN/chan_misdn.so? When loading chan_misdn.so from Asterisk, I get a "No lower Id port:1" error. The /var/log/messages file says: "MISDN free_device: entitylist not empty" I'm using Linux 2.6.11.11 + mISDN-CVS-2005-05-01 + Asterisk 1.0.7 + Zaptel 1.0.7 chan_misdn build from chan_misdn-beta-0.0.3-rc6

> Hi Damon and others, > > Your example is still doing what I tried already, so eventually the > dial command ends like: > Dial(zap/4/*21*) > or > Dial(zap/4/*31*) > I prefer to use Dial(zap/4/*21*<thenumber>) > or Dial(zap/4/*31*<thenumber>) > > But whatever I try, the error message as in my first post shows up and > the line hangs up before the

Hi all, I'm struggling some time now with this problem. Googling and searching on this topic did not deliver the answer yet, so my last hope is this list. Analogue to the things which are possible with modem.conf, where I can configure the MSN's to act on, I would like to have similar functionality. This is the idea: I have 1 ISDN line, it can be reached by 4 different MSN's. I have

Hi, When I fit the mixed-effects model by lme, I got the convergence error code = 1 message = iteration limit reached without convergence (9). Even after I increase the iteration numbers, it still did not work. The only way I found to avoid the error message is to increase the number of iterations for the EM algorithm, which need to be at least 1100. But the weird thing is that the correlation

Hi Folks, After going to the paifull steps of installing AreskiCC and finally being able to access the webinterface, connecting to *, importing rates and setting up accounts I am not being able to make a CALL: No matter what number i try to dial I get the same response: The number you have dialed is currently unavailabel. Please enter thenumber you want to dial starting with 1 for local and

Hi, I've got a (maybe stupid) question: I'd like to install Asterisk on my notebook (just to play around with it). Is my internal phone modem equivalent to an FXO port? Hence, could I make phonecalls to my pots-line with it? thanks philipp

Hi, I've got a (maybe stupid) question: I'd like to install Asterisk on my notebook (just to play around with it). Is my internal phone modem equivalent to an FXO port? Hence, could I make phonecalls to my pots-line with it? thanks philipp

Dear experts, I'm trying to transfer a mixed model developed in SAS to R. This it what it looks like in SAS: proc mixed method=ml; class a b c subj; model y = a|b|c; repeated /subject=subj type=ar(1); I tried something like this in R: mixed <- lme(y ~ a + b + c + a*b + a*c + b*c + a*b*c, random = ~ 1 | subj, correlation = corAR1(form = ~ 1 | subj)