similar to: Asterisk with PortaOne Radius client- problem in accounting script with OH323

Dear all, I've searched the web about Asterisk with Radius integration for user authentication, and got a bit confused... I see that there have been some work around it, there is PortaOne's Radius client patch, an still open branch of Digium Issue Tracker "SIP peer authentication on an external database (RADIUS - LDAP)", etc. Although, none of these seems to give me the

hello i am trying to work with radiusclient form portaone. but i have some problems in installation. when i am trying to use example from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth error sip debug Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0

Hi Freinds, I am trying to do the following. 1. Accept the call from call ifle. 2, Answer it 3. Extract the dial number and variables from the call file request. 4. Pass that parameters to voiceglue 5. Catch the parameters (dialnumber and cli) in voiceglue.conf 6. Point to the voice.xml file dynamically by matching number I want to make the line in voiceglue.conf as <DEST_NUMBER>

Hello, I am trying to diagnose an intermittent error when a call comes in over our PRI lines. The problem appears random, however I have feeling it has something to do with the call volume, as the frequency increases with more calls on the system. I am not an expert when it comes to reading the PRI Span Debug statements but here is a call that had a problem and I bolded, italicized, and

hello i am trying to follow http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth can any one tell how to install this 2. Install Asterisk::AGI and Asterisk::Manager (unfortunately it is not on CPAN yet!) thanks in advance Kamran __________________________________ Discover Yahoo! Use Yahoo! to plan a weekend, have fun online and more. Check it out!

hello i m trying to use radius with asterisk http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth how to fix this patch 8. Make sure that your Asterisk includes all related bug fixes and patches, namely: - SIPGetHeaders for chan_sip (derived from chan_sip2 ) i m using asterisk-1.0.9 extensions.conf exten => _X.,1,SIPGetHeader(SIP_Authorization=Proxy-Authorization)

Hi all, I'm trying to get Asterisk do AAA against a Cisco VSA compatible RADIUS server/billing platform. I decided to give a try to appradius (since it was written in C and didn't require all the modifications to Asterisk that PortaOne's client does) played with it the whole day, after lots of modifications got it working only to find out that it'll need a lot more work to

I'm trying to pass a 65K DATA call in one channel on my Digium TE411P to another channel on a different span. Any idea what could keep this call from going through? -- Accepting call from '' to '5444' on channel 0/1, span 1 -- Executing Goto("Zap/1-1", "sendto-definity|5444|1") in new stack -- Goto (sendto-definity,5444,1) -- Executing

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Please can you provided me by a radius module name for asterisk,or how to authorize user and get cdr from radius server. Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip

hello how to solve these errors /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10 use Asterisk::AGI; vi /etc/asterisk/extensions.conf exten => _X.,1,agi,agi-rad-auth.pl|Routing=SIP&AuthorizeBy=SIP vi /etc/asterisk/modules.conf load => res_agi.so <---------------errors------------------------> *CLI> Can't locate Asterisk/AGI.pm in @INC (@INC contains:

Hi, How do I listen to .sln audio file or convert it to some format that can listened to? Sox does not seems to support .sln as an input file. Thanks. - Andrew

Hello, I would like to know, if in Asterisk version 1.6.0-beta9 is it possible to use Radius as an external authentication server. Thank you for your help Ciao Salvo

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

hey asterisk folk, i'm new to asterisk, but not to freebsd as i've been using it and developing on it for quite a few years. we're planning on deploying asterisk as part of a voip provider's service platform and naturally i'd rather be using freebsd instead of a linux based distro. as such, i am very, very excited that there is work going on towards porting the zaptel drivers

Does anyone have any experience with asterisk and this radius module? http://appradius.minitelecom.org/ If not, what radius module is recommened, for tracking SIP phone calls for things like billing per phone?

hello i am using ast-rad-acc.pl from portaone connected with asterisk manager. my (%cdr) = @_; $cdr{'CALLERID'}, $cdr{'DNID'}, these are empty why these two variables are not working on hangup any comments thanks Kamran Ahamd __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

I need to integrate Asterisk with radius server, because I need auth and signaling from radius (i.e start and stop messages in realtime) (or something that do that). I can't find an implementation of this with enough documentation to install and run.-