search for: portaone

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 support any VSA at all (i.e. cpprad doesn't support VSA, etc. etc.) Now, according to the wiki, PortaOne's RADIUS client implementation uses C...

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 /usr/lib/...

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! http:...

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) ___...

...compiled and installed) Now When I am trying to get asterisk communicate with a Radius (in my case: it's the VoiceMaster Radius) I was able to do the following: After installing all recommended to download and install radius client for asterisk (http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth ), but without applying the patches because I wasn't able to apply them on version 1.0.5 I run ast-rad-acct.pl in the background (successfully) I run the agi script from within asterisk contexts I was able to send an Access-Request and to get authenticated by obtaining a reply from...

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

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 confidence to implement it in a production environment... What do you people recommend me? Whic...

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 these two variables ar...

...bK0cb77ea3;rport=5070 From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614 To: <sip:0100000000 at 52.22.22.22:5160> Contact: <sip:+27888888888 at 41.11.11.11:5070> Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11 CSeq: 102 INVITE User-Agent: PortaOne Max-Forwards: 69 Date: Sat, 19 Sep 2020 21:42:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-LOCATION: X-StartID: SDvb9r601-d482cc2b5e0b32417957ea02aaade464-a04aba0 Content-Type: application/sdp Content-Length: 283 v=0 o=root 6009 6009 IN IP4 41.11....

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:

...reebsd. looking at the current cvs copy of the ports though, the zaptel drivers are at version 0.7, and the Makefile for the port is marked as "requires FreeBSD 5.x to build", though the Asterisk FreeBSD wiki at http://www.voip-info.org/wiki-FreeBSD+zaptel and the Changelog at http://www.portaone.com/~gonzo/ChangeLog say that it can be built on FreeBSD 4.x i guess my primary question is, can the ported zaptel drivers work on 4.10 (or 4-STABLE cvsupped a couple of weeks ago) ? i plan on having asterisk running in a small Pentium III machine with a Wildcard TDM400P, specifically the TDM40B...

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 ha...

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.-

I have a telco providing DTMF inband, they say they can't provide it any other way. This is creating headaches for me. What is the common method for SIP DTMF? Kpml, or 2833 or inband? My handsets don't support inband so I'm tying up some expensive resources to convert the inband DTMF to out-of-band DTMF... Can you recommend a vendor in US that provides SIP with DTMF in RFC

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

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

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph