search for: landy

Hello. I am currently testing an asterisk server using the default codecs, I have allow=all, and noticed everytime I test it in a wireless lan the latency rockets off the roof to over 1000ms. I would like to test g729 since it uses less bandwidth but, read somewhere I have to buy a license per every channel I have. Does this means if I have my server connected with 10 sip clients I need to buy a

...ateway for the rest of the lan? Itself? My "net" NIC''s address? Something else? My lan isn''t getting internet access using the default Shorewall config file (edited per http://www.shorewall.net/two-interface.htm) and I think it may be a interfaces setup problem. -- Landy J. Bible The University of Tulsa Computer Science Student IS Computer Help Desk Tech. Java Programmer Lighting Designer Train Nut

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound. What can cause that problem? Thanks in

Hello. Happy New Year to everyone. I have a small WISP and would like to have customers to call our number to check their balance. I am planning on writing an AGI with php so it can get the customer info from the customer database. I don't know how to interact with the caller while in the agi script so this is what I have in mind: [test-agi] exten => 33,1,Answer() exten =>

Hello. I have a question regarind sound files in asterisk 1.6. I have a sound package in ulaw format and I would like to know if I have a sip extension with allow=alaw would asterisk convert that file to the codec the user is allowed to? I am having a problem playing a file that exist in /var/lib/asterisk/sounds/es/good.ulaw but asterisk is telling me it doesn't. Here's what I get when

On RHEL5, dovecot 1.0.7, I have set up sendmail to use `deliver` for my local mda. It keeps giving me this error for the root user though: Jul 12 12:51:29 mail sendmail[4105]: o699225f001348: to=<root at localhost.localdomain>, ctladdr=<root at localhost.localdomain> (0/0), delay=3+08:49:26, xdelay=00:00:00, mailer=local, pri=7502879, dsn=4.0.0, stat=Deferred: local mailer

Hello List. I am puzzled and how asterisk listens to calls or connections from clients. When I do a netstat -nat I don't see asterisk listening on port 5060. Now, I'm testing a server with three network interfaces: two to the internet doing load balancing and the other to our LAN. I would like asterisk to only accept connections coming from our LAN but, can't find where to configure

Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for "?VoIP residential services"? if yes, how? if no,

Hello, I want to import a txt table into R but the software give me this message. I have windows vista. Errore in file(file, "r") : cannot open this connection Besides: Warning message: In file(file, "r") : cannot open file 'C:/Users/Vincenzo/Desktop/prova/prova.txt': No such file or directory thank you **********************************************************

Hello, anybody know the procedure to plot the geneland result in a map? thank you ********************************************************** Vincenzo Landi Post Doctorate student Animal genomic and breeding cell:0039/3395388713 Fax. 075-5857122 [[alternative HTML version deleted]]

Dear R-help folks, Can someone guide me to a source where I can learn more about the / operator in model formulae? I found a reference to it in Venables and Ripley's MASS, p. 142, where it says, in reference to ANCOVA: "Terms of the form a/x, where a is a factor, are best thought of as 'separate regression models of type 1 + x within the levels of a.'..." This seems

Dear statistics and R experts, I am a new R-user and my statistics is probably more than a bit rusty. So forgive me if the following question is relatively simple. I would like to plot the predicted values from a quantile regression analysis (quantreg v.3.31; R v.1.7.1) so that I can evaluate the fit. My response variable is tree growth (continuous), and my predictor variables are height

...all traffic two and from an internal ip address on a specific port? (ie, no port 80 outbound from 192.168.6.50, and block all returning packets to that same ip) Restrict all traffic out to an external ip address on a speicific port? (ie, no port 80 outbound to www.example.com [or its ip]) -- Landy J. Bible The University of Tulsa Computer Science Student IS Computer Help Desk Tech. Java Programmer Lighting Designer Train Nut

I am testing an ivr but I'm having problems. The call keeps looping and it doesn't hangup the call after passing three times through the menu. Here's my conf: exten => s,n,NoOp("Here's Count") exten => s,n,NoOp(${COUNT}) ;123,n,Set(COUNT=$[${COUNT} - 1]) exten => s,n,GotoIf($[${COUNT} = 4]?33,1:44,1 ) exten => 1,1,goto(tech-support,s,1) exten =>

Hello. Sorry to repost this message but, I don't have the original message in my inbox nor in my sent box. Well, last week I posted a problem I am having trying to use an asterisk server use a voip provider and a pstn. Pstn works fine but, I cant even connect to my provider's server. I don't know what I'm doing wrong. I tried using a soft phone and I'm able to register and

Hello List. I would like to know how I can use two or more service providers with asterisk to be used randomly for ei, if an user tries to make a call I would like to randomly use a provider. It doesn't matter where the call is destined to. Thanks.

Hello List. I am having problems retreiving voicemails on my system. I noticed when someone leaves a message through the pstn line I can't hear anything. I tested leaving a message from one of the extensions and that can be heard. I don't know if is the type of card I'm using for analog ( cheap X100p modem ) calls but, can't hear any message coming in from that line. Any

Hello. I'm trying to set TTS with Cepstral and Swift but can't get it to work. I get this error when testing it: -- <SIP/101-00000000> Playing 'welcome.gsm' (language 'es') -- Executing [702 at local-calls:3] Swift("SIP/101-00000000", "Hello this is ceptral") in new stack [Jul 28 18:29:16] NOTICE[5191]: app_swift.c:304 engine: Text to

Hello. I'm looking to buy a FXO card to do some testing with two phone lines I have at home and was looking in ebay some and found some cheap ones but, the I've never heard of the brand or manufacturer: chinaroby. They run for about $99 plus shipping. Have any one used these? or please recommend one... Money IS an issue. Thanks.