search for: landi

Displaying 20 results from an estimated 29 matches for "landi".

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2009 Dec 01
6
Question about g729
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
2005 Jun 10
11
/etc/network/interfaces
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
2004 Aug 12
10
H323 problems
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' --
2009 Oct 08
4
No sound on voicemail from analog line
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
2010 Jan 02
4
Help getting info from caller
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 =>
2009 Nov 26
1
Unable to open sound file error
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.
2010 Jul 12
3
deliver and root user
...$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
2010 Feb 16
6
Asterisk listens on all NICs
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
2010 Jun 15
2
a2billing for residential voip usage
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,
2008 Dec 10
1
plot Geneland result in a map
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]]
2005 Jun 14
2
Two questions
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
2009 Oct 22
2
ivr menu not hanging up call
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 =>
2009 Nov 16
1
can't call through voip provider
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
2009 Dec 12
1
how to randomly use provider?
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.
2010 Mar 26
1
no voicemail on pstn line
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
2010 Jul 28
1
app_swift.c:338 engine: Failed to set voice
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:
2010 Aug 03
1
chinaroby fxo card - never heard of them
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.
2009 Nov 12
1
Can't connect to voip provider over NAT
Hello. I'm trying to test an Asterisk server by using a VOIP provider for international calls but, I'm having problems trying to get my server communicate with theirs. I don't know if I'm having all these issues becuase I'm behind NAT or what. I have the following in my server's sip.conf: [provider] type=peer host=<theprovider's server> username=<username>
2009 Dec 13
1
Unable to open file...
Hi List. Don't know if I already posted about this problem but, if I have I apologize for the double post. I am trying to test a time of day extension dialing 80, all I'm trying to test is if is morning I would like asterisk to say "Good Morning" but, when I run the test I get the following error message saying that the file doesn't exist and it does: Night..............
2007 Jan 28
4
Compiz .36 => No Borders
hi all i'm running debian sid..and i tried to compile the las version of compiz because the old version of compiz did not supported compiz settings.. btw here is what i did: 1) followed the official compiling guide => http://go-compiz.org/index.php?title=Compiling 2) try to run by launching compiz...but the only results was to not view the windows' borders.. so i