similar to: asterisk freepbx difference or solutions..

Is there a way to monitor a call mid-stream? I did look on the Wiki and found that AstGUI can do it, but it's a bit of an overkill. What I want is for a customer service rep, sitting in front of a Cisco 7960, to be able to hit a button (either on their phone, or maybe a specific webpage) that will start recording the call from that point on. I'm thinking the services button on the

Hi, what is the preferd way to configure/start mongrel? [ ] with Mongrel::Configurator => HttpServer [ ] @var = HttpServer.new(...) @var.run I ask because the only debug methode i have found was in Mongrel::Configurator ;-) @Zed: are you also subscribed on nitro list?! regards Alex

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

Hello, I am trying to create a simple IVR for testing.. What i did is to create a voice file from asterisk-gui. And i saw it created that under /var/lib/asterisk/sounds/record/ as deneme.gsm Then i tried to make a IVR menu and play that file. I tried exten=s,4,Playback(/record/deneme.gsm) exten=s,4,Playback(record/deneme.gsm) exten=s,4,Playback(deneme.gsm)

Hi, Previously i was using asterisk 1.4 with freepbx installation. To try the 1.6 version i installd anc configured everything.. Just one thing didnt work so far.. I am using grandstream 2000 and it has a line busy indicator for chef secretary phones. But now, this feature does not work. I can see the line is online..with a green steady light.. But when the line is busy or DND, it wont change to

Hi, I am using asterisk 1.6.0.10 For debugging i set verbosity to 10 with asterisk -vvvvvvvvvvr.. now i am trying to set it lower but.. when i type asterisk -r it starts with Connected to Asterisk 1.6.0.10 currently running on asterisk1 (pid = 2408) Verbosity is at least 10 when i try set verobisty 1 or similar commands.. i think this command is obselete in 1.6 .. set verbose 1 No such command

I have Cent OS 5.1 I also have http://phpsysinfo.sourceforge.net/ I have asterisk running . Now when I look at System Information , I see that "Physical Memory" keep increasing and at one point it reaches 96%. Then my sever get hang and then I have to restart it. I have 4 GB RAM. Processors 2 Model Intel(R) Core(TM)2 Duo CPU E7200 @ 2.53GHz CPU Speed 2.53 GHz Cache Size 3.00 MB

Hello, I want to make an agi script to match incoming DIDs with usernames. I tried to do such entry in incoming trunk. [DID_diddw] include = from-didww [from-didww] exten = 3130XXXXXXX,1,AGI("did.php") exten = 3130XXXXXXX,n,DIAL(SIP/${yup_no},20) but when i run the rule it says chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to extension

Hi all, I was using dahdi 1.6.2.0.9 version for a long time. We decided to upgrade to 1.6.2.22 a few days ago. After that we started to have some problems with dahdi channels. PS:DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2 We have 2 PRIs between Ericsson pbx and asterisk and a sip trunk for outside calls. At begining everything works fine but in a few hours, calls from asterisk to ericsson

Hello, I have asterisk 1.6.2.9-2 I tried to install fax utility as it is shown on pdf documents on asterisk site. I downloaded Opteron compiled res_fax and res_fax_digium files and copied to /usr/lib/asterisk/modules/ where default modules directory is. I created a free fax license and created license file on asterisk server too. WHen i run asterisk it crashed. I noticed that if res_fax.so

Hello, can anyone using astgui client i have a problem in installation phase everytime i try to create database from MySQL_AST_CREATE_tables.sql it gives error in phone table ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'DBY_server VARCHAR(15), DBY_database VARCHAR(15) default

Odd question here but if I have asterisk running on PC (and mplayer installed). and a video phone calls up the asterisk PC can that video image be played on mplayer? If so how do I do that? How can asterisk pipe the video into mplayer so as to display the video image on screen? Thanks, Jerry

Hello, I want to enable call forwarding for asterisk 1.6.0.6 I couldnt seen any config or option on gui or extensions.conf about it. I found some dialing plans to enable it on web as follows: [apps] ; Unconditional Call Forward exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4}) exten => _*21*X.,2,Hangup exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) exten => #21#,2,Hangup ;

Hello, There is a problem that i can not figure out how to solve. I got users with 5 digit usernames for sip. Some users has a callerid for outside calls. I have such problems When a user activates (for ex) call forwarding, System creates that entry on database as CFIM/callerid not the username, So this rule works only if a call is made from outside to the callerid. Not the local calls made

Hi, i managed to connect to Ericsson MD110 with PRI at last. And made a successful call thru asterisk to ericsson. But when i try to call from ericsson to asterisk i got an error on asterisk side. And i couldnt figure out why. Here's my extensions.conf about incoming calls. [DID_span_1] include = DID_span_1_timeinterval_all,${timeinterval_all} DID_span_1_timeinterval_all] exten =

Hi, i know i am asking a lot of questions lately in this forum..sorry about that first of all. :) Ok, here is the deal.. I am trying to make a hybrid system with an ericsson MD110 and asterisk. Internally we have 4 digit phone extensions on ericsson.. and so in asterisk. So, what i want to do is to call pbx side without adding 9 or etc to the begining of the number from asterisk clients.. For

Hello, I was testing my asterisk for a while with 1.6 without much problem. Now i am trying to install a new system with asterisk 1.4 but now i am using a dual pri card instead of single pri.(TE220P) What i want is to use both PRI ports as group. Now i have zaptel.conf file created as follows -------------------------zaptel.conf-------------------------- # Span 1: TE2/0/1 "T2XXP (PCI)

Hello, I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in registering users. As i see from debug it successfully reads from users.conf but later,when a user tries to logon it say peer not found.... And there were an error msg about mysql about the username field..Smthing changed in mysql tables??? Now i downgraded to 1.6.0.9 again and everything is working..

Hello, Our asterisk is connected to an ericsson pbx by PRI. What i want is the asterisk clients should call operator numbers by dialing 0 But, when a call is made to ericsson via number 0, it assumes that the call is made from outside, so it doesnt allow to be dialed. There are 3 real operator extensions which is grouped by ericsson for operators. Lets assume 1111 1112 1113. What i want to know

Hello, I am using 1.6.2.9-1+b1 asterisk.with cdr_mysql. Everything seems workging correctly except cdr logs. It fills up all data when a call established except src and clid Wht can cause this and where should i check??