similar to: One way audio

The configuration I'm building replaces an existing PBX with Asterisk. There are 8 existing modems that people use to call in from the outside to connect to PC(s) on the inside to transfer data, etc. Callers access these modems by calling the main number and then dialing an extension for the modem they want to talk to. What are my options for supporting these modems with Asterisk? Here are

Does anyone know how (if possible) to do three way calling on the UIP-200? There doesn't seem to be much info about this phone, but all the feature lists I've read says it can do conference calls. I can't seem to do it, though. Any help would be appreciated. -- Seth "et lux in tenebris lucet" Mattinen sethm@rollernet.us

I've been trying to get fax reception to work using an SPA-2000 to ring the fax machine or modem that's taking fax calls. I was curious if anyone else had tried something similar, and if so, had any luck getting it to work reliably. I've been able to get it to work, but it isn't reliable. (Pages/lines of black dots result more frequently than not.) The incoming lines are FXO

Hi Hekuran I have installed asterisk-1.0-RC2.tar.gz, asterisk-prepaid-0.3.1 and postgresql. When I tried to call from any IAX client to another IAX client and also sip client to sip client it worked fine. And also the cdr table filled properly. Now I tried to configure asterisk-prepaid-0.3.1 with asterisk. I have compiled asterisk-prepaid-0.3.1 and also copy the configure file. I

Hi: Is DynExtebDB module still working?? -- JO

I've come across an odd dialing problem with my * setup. After * has been running for a while, if I try to dial out on any of my zap channels, (both are X100P cards) it picks up the line but never sends the DTMF. Has anyone heard of or seen this problem before? Right now I'm looking at 19 hours of uptime on * itself. If I restart it, everything works fine for a little while, then the

I can't seem to make the "Hold" button function on the GS BudgeTone-100. I'm trying a procedure like this: 1) On a call 2) Press "Hold" button 3) Hang up phone What I expect is for the call to go on hold until I pick up the receiver again. (Like my SPA-2000's, except it's a flash, but I can hang up the phone and the call waits there for me to pick it up

I''ve got an order model that stores order data. One piece of data is a credit card type, which is a digit 1,2 or 3. I have a cardType model that has an id, shortName and LongName for the credit card merchant (visa, mastercard, amex). I want to be able to say: order.cardType.shortName, but can''t seem to get has_one working. It works with has_many and a finder_sql statement on

Hello, I'm trying to setup a queue where call goes from agent to agent in strictly set order. I have queue (roundrobin): Agent1 penalty 1 Agent2 penalty 2 Agent3 penalty 3 When I call to this queue Agent1 rings. If this agent does not take the call, after set timeout same Agent1 is dialed again. The call never goes to Agent2 (only when Agent1

I''ve been confused by this question for near two days ...my puppet master version is 2.7.9-1.el6 and client versiong is 2.6.16-2.el5. This is what my command lines shows: *[root@agent1 ~]# puppet agent --server=edward --test --waitforce 30* info: Creating a new SSL key for agent1 warning: peer certificate won''t be verified in this SSL session info: Caching certificate for ca

Hi, I am able to install my RPM via this puppet code ... [root@agent1 ~]# puppet apply -v install_named_conf.pp info: Loading facts in /opt/puppet/share/puppet/modules/stdlib/lib/facter/facter_dot_d.rb info: Loading facts in /opt/puppet/share/puppet/modules/stdlib/lib/facter/puppet_vardir.rb info: Loading facts in /opt/puppet/share/puppet/modules/stdlib/lib/facter/root_home.rb info: Loading

I want use iLBC and have following in mind, please help me is it possible ? ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM 1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC because a lack of codec). 2. SIP incoming codec should be iLBC (snom is ALAW). 3. SIP outgoing codec should be iLBC /snom

Hi, I'm really lost on the directory functionality of the asterisk. I am currently setting a ACD using asteriskathome. I want to use the # feature where in caller can access the companies directory. My issue is how to setup the entry on the directory. Sorry for my silly question Can somebody guide me please. Regards, Leonimar Cape __________________________________________________ Do You

Because I'm tired of seeing Debian users look like idiots for using such horribly out of date versions just because they run "stable", I've set up a page with .deb packages of my 1.1.x rebuilds for use under lenny. http://www.rollernet.us/opensource/ Standard disclaimer applies: use at your own risk, no warranty, no support, if it eats your first born or erases your computer

I have both the PJSIP add and the chan_sip way of adding SIP headers in there. The Verbose is showing the variable value is there. The INVITE to PJSIP/Agent1 does not include either X-My-DNID or X-My-DNID2 headers. exten => 1234,1,Verbose(X-My-DNID:${MY_DNID}) same => n,Set(X-My-DNID=${MY_DNID}) same => n,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID}) same => n,Dial(PJSIP/Agent1)

Hi, I successfully installed postgres and prepaid application in my asterisk box but after I digited the code I receive this error: ERROR: Function asterisk_authenticate("unknown", "unknown") does not exist Unable to identify a function that satisfies the given argument types You may need to add explicit typecasts -- Playing 'prepaid-no-aaa'

hi I've written, or upgraded a little more, a plugin for asterisk/nagios, just in case someone should be interested. it uses the manager interface to connect and checks staus. it's a dirty hack, but it works. see https://sourceforge.net/tracker/? func=detail&aid=746083&group_id=29880&atid=541465 for more info roy

Hello, i want to use asterisk as a gateway for H323-Phones. But i cant get it work. I'm using a gatekeeper on another computer. My IP-phone is registered there. Does anybody can sent me an oh323.conf and extension.conf as examples? Thanks in advance Erik Bastian -- NEU : GMX Internet.FreeDSL Ab sofort DSL-Tarif ohne Grundgebühr: http://www.gmx.net/dsl

I am trying to install the Modified Prepaid App. I have installed PostgeSQL, created the tables, etc. Make Install runs ok. The when I try to launch asterisk (asterisk -vvvvvgc), it fails to run. I get the following errors, 1st error: [app_prepaid.so] => (Prepaid Application) == Parsing '/etc/asterisk/prepaid.conf': Found Jun 16 14:27:27 ERROR[-1085267840]: app_prepaid.c:127

Hello, Here in our office we are testing Asterisk. My collage Igor created to Asterisk PrePaid application with Postgresql. It is not in Perl. We would like to release it to the group.... as soon as it will work ok. It will have authentication, different rates for users, different rates for destinations and so on. Is there anybody who would like to improve it ? -- Bart -------------- next part