search for: landis

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

If I''m using eth1 as my lan zone on my router box, it needs a static ip... what do I set the gateway option to in /etc/network/interfaces since this computer is actually the gateway 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

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

...$u') MODIFY_MAILER_FLAGS(`LOCAL', `-f') ... MAILER(smtp) MAILER(procmail) MAILER(local) == END == == dovecot.conf == ... protocol lda { ??postmaster_address = postmaster at mydomain.com ??hostname = mydomain.com } == END == Any ideas, or do you need more information? -- Thanks, Brad Landis

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]]

...s is compared to the lowest level of that main effect (when using contr.treatment). But the alternative model (growth ~spp*lt / ht -1) contains all three levels of the main effect, spp. I'm not sure this is at all clear, but I am happy to try to clarify it. Thanks much in advance, Matt Landis R. Matthew Landis, Ph.D. Dept. Biology Middlebury College Middlebury VT 05753 [[alternative HTML version deleted]]

...te the predicted values without having to manually add up all the relevant coefficients for each level? Or, even better, is predict.rq out there somewhere that I haven't found? The help on rq objects does refer to it, but ?predict.rq doesn't turn up anything. Thanks for any help, Matt Landis R. Matthew Landis, Ph.D. Dept. Biology Middlebury College Middlebury VT 05753 tel. 802/443.3484 fax.802/443.2072 [[alternative HTML version deleted]]

Is is possible to... Restrict 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

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.