similar to: FreePbx/asterisk/openser

Hi, i try install FreePbx by tuturial in http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Paso&view_comment_id=13443 but i have this error when i try install freepbx: #pear install db No releases available for package "pear.php.net/db" Cannot initialize 'db' , invalid or missing package files Package "db" is not valid install

Ive installed asterisk and freepbx. Through the interface ive configured 2 extensions, 6000 and 6001. My problem is that when i try to call from extension 6000 to 6001, i hear this msg "Im-sorry&an-error-has-occured" and the call is terminated. As expected if i call to another number i get an error. i thought the problem might been related with the NAT but if checked and changed some

Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound

HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: "Couldn't load variables.txt?aldope=xxxxx " I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? --

Hi, I don't explain very well what my problem, but i can't make calls. i analise my log full and i found two errors Jul 3 19:02:08 ERROR[4670] res_config_mysql.c: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. Jul 3 19:02:08 VERBOSE[4670] logger.c: -- Added extension 'exit-FAILED' priority 1 to macro-vm Jul 3 19:02:08

Hi, i have a doubt about autentication in asterisk. it's possible to integration the asterisk with the other server for autentication, for example kerberos, ou other? i want to implement asterisk in a department of university, but it's necessary autentication by students, login and password for example. thanks. -- Carlos Jer?nimo

Hi there, Any of you know how can i configure two PBX servers with FreePBX with load balancing? Do I need a SIP server like OpenSER? Any of you have some configuration reference? I have installed AsteriskNow in both FreePBX servers. Thank you! Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi i'm student and my final project is related to Voip. I have Asterisk almost fully configured. The next step is to accept login of users, that data is in Universitys database which uses ActiveDirectory and also Ldap and Kerberos. It's possible? I don't want authentications in sip.conf, but in other remote database. The problem is i don't have ideas how to start with. I would

Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: root@hernandezz-laptop:/home/hernandezz# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i

Questions: Does anyone have a really STABLE asterisk system running about one year without need to restart the service or the SERVER ? Does anyone have a production Call Centre saled that don't lockup and is stable for 6 months ? I'm asking this questions because we have choose Asterisk for our call centre solution but, since the bugtracker only grows and people still want to stuck more

I'm taking the liberty to announce this event on the Asterisk mailing list, as Asterisk and OpenSER form a valuable combination in SIP architectures. The second edition of OpenSER Summit will take place in San Jose, USA ,on the 17th of March, 2008, during VonX Spring 2008 pre-conference events. This is the first US edition of the OpenSER Summit - to learn more about the agenda and layout of

Hi all, I try to make a call from my Openser(SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip

Hi All, I am stuck with an issue in the Openser+Asterisk Forking. In this solution we are using Openser as the Registrar. Hence it will store all the contact bindings along with the q values for a given user, say ua1. The current setup is such that the INVITEs are sent to Asterisk by Openser and Asterisk sends out the INVITE. Now if ua1 is registered with two different contacts having

Hello Users openSER is sip proxy and registrar , Asterisk is as PBX, Conference and Voicemail servers, openSER and Asterisk are in the Same N/w Where As the UAC are in Behind the NAT, When Astetrisk is not integrated , UAC are in Behind the NAT is working, openSER is 192.168.2.5 Asterisk is 192.168.2.6 I'm just use rewritehost to asterisk server, UAC ----> openSER - - - ->

Hi! Meanwhile also the location is fixed: it is happening at metalab (http://metalab.at/) - a place for geeks. Thus, we meet there at Thursday, 20.9.2007, 19:00 CEST (=local Vienna time). Metalab is located next to the city hall: http://metalab.at/wiki/Lage Metalab is no pub/restaurant. Thus, don't come hungry! Nevertheless liquid food (drinks) is available. We meet in the library (in

Suggestions on how to use Asterisk to collect CDRs from a OpenSER-based proxy / call routing setup? I need to get simple CDRs; not for detailed settlement/rating, but just for reconciliation with an ultimate TDM carrier just to make sure we only get billed for what we're actually using. I'd use the often-heralded approach of dumping a call from OpenSER into Asterisk and having it

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

Hello Users , I Posted to mailing list, No one is replying My issues, My Issue is No Audio when Openser and Asterik integrated in Same LAN , When UAC are Behind the NAT, With out the Asterisk integration Behind the NAT is working Fine. SIP port and RTP ports are forwarded into router to OpenSER System only. openser.cfg listen=192.168.2.11 alias=sip.hyperion.com # Invite Section if ( method==

Hi, I need some help, getting to work asterisk MWI. I set up Asterisk as voicemail server for Openser as this tutorial shows : http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+OpenSER+1.3 . My voicemail system is working but, I can't get to work the message waiting indicator. It doesn't seems to send the Asterisk any NOTIFY message to the Openser box. How can I

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)