similar to: Asterisk, IAX and NAT issue

Hello Everyone, I am trying to change our /etc/sudoers (using visudo) to allow 2 commands to be run as root without a password, but it isn't working. Here is the part of the sudoers file that is in question. # User alias specification User_Alias FULLACCESS = doug, scott # members of the FULLACCESS User_Alias may run chown and chmod without a password FULLACCESS ALL = (root) NOPASSWD:

Hi everyone, I am trying to connect 2 asterisk servers via IAX, but it just fails to do so.. I'm using SIP to connect the IP phones on the LAN at the 2 different physical locations where each server resides and I'm able to communicate on my LAN via SIP without any issues. The problem is that I'm unable to make Asterisk servers talk with each other via IAX.. Here is my issue.

hello i finally have success with winbind!!!! :) thanks everybody so, i use this acl's in squid external_acl_typqe NT_global/-group %LOGIN /usr/local/squid/libexec/wb_group acl FullAccess external NT_global_group "domain group" httpd allow FullAccess httpd deny all with this configuration all is DENID by squid. what???????? it's possible to write acl's specifics for

Hi, I'm running samba-3.0.14a_1,1 under FreeBSD 5.4-STABLE. I migrated all my data from samba 2 to samba 3, and I'm having a problem with some users. For example, user john can login perfectly to the network. It?s ok. But user smith not, him get the error ?The network name cannot be found?. But all configurations of john and smith are the same. I found this on Samba Official Hanbook:

Carlos and Stefan (and other who have helped): I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling Asterisk is unrealistic in my position but I wonder if I can build the one module. Here's what I do have: apbx:~ $ locate *res_timing_timerfd* /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo

I have setup an Ast 11.6 host and I want to login via AJAM. I setup manager.conf, http.conf described in the docs. When I login via the AMI it works fine (see below), but when I login via AJAM the same credentials fail (see further down) Can someone tell me how to fix this? ----------- Connection closed by foreign host. root at pbx:/tmp# telnet localhost 5038 Trying 127.0.0.1... Connected to

Dear All, I bought one IP PHONE from Integrated Networks which was showed to wiki too: http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks I have problems with the Asterisk authentication. It does't want to LOG IN to Asterisk; it always says "LOG ON FAILED". I'm using the IAX2 protocol and all paramters seems to be correct. Does somebody use the same IP PHONE with

Hi, When I try to run the command wmix to mix two WAV files recorded by the Monitor application I get the following warning in the console and the macro exit at that point. Running the command from a standard system console it works. More, even from this macro it works and produce a valid mixed file, but still get that error and the macro cannot continue. Why? I have tried even with a simple

Hi all, i;m obviously a newbie, its been 2 days that im trying to figure out a way to deny a specific extension (300) from calling another specific extensions (03) except if the caller punch a specified password.. sorry if im not explaining myself well.. heres an example: i called my pstn line(with 300 as its sip account), an attendant answers and asks me to punch in an extension number right

Hi, I got my * working fine with FWD at office with 2 extensions, i receive calls and i can make calls thru FWD. I got also my * at home, and i connected it using auth=rsa. From my home, i can make calls using my office iax, but if i try to redirect incomming calls from FWD to my * at home, it rejects the call. I created the pub/key pairs for rsa and its working ok and i just pasted the

Hi all. Thought you would all be interested to know that i, like many others, am experiencing issues with Samba and Photoshop 7.0. The problem is specific to Photoshop 7.0 when saving files to a Samba share via a W2k/XP OS. Users experience a disk full error when saving from Photoshop but have no issues copying/moving etc files to/from the share outside of PS. Well, after some testing, the

Hello, I am experiencing weird problem in Asterisk 16.2, possibly a bug. Same thing works fine in Asterisk 11. Here is the situation: I have 2 extensions on 2 phones. 4 extensions in total. phone 1: 8882 8382 phone 2: 8884 8384 And I have 2 SIP trunks for outgoing calls. I want to call via SIP1 when called via 8882 or 8884, and SIP2 when called via 8382 or 8384. And one last detail. SIP1

Hi all, I'm a newbie and I have a configuration problem with Asterisk. Seems that I'm not able to call an outbound number. I'm quite sure that it is a configuration problem, but I'm not able to find out where is the mistake, even reading several docs to www.voip-info.org. I do not have a good knowledge of Asterisk, I'm not very familiar with its configuration and I've a

Hi all, I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028. This IP phone is registered in an Asterisk PBX and I've a problem when I dialing internal number. If I dial an internal number, like for example 102, the IP phone takes 35 seconds to send the number to Asterisk; here below the debug output 192.168.0.75: first 192.168.0.75: 1 192.168.0.75: 10 192.168.0.75: 102

Ok, to review, I'm trying to get Avaya 9608G to come up in a pure Asterisk environment-- no Avaya SBC or gateway or any other Avaya gear in sight. I have the phone working to the point where it boots up properly, then displays a Username and Password prompt, and says its extension is 123 and the time is 4:57p, which is wrong. But please don't tell me the only way to program up each

Hi, I have a strange problem trying to update Asterik on one of my boxes. I have done the following: - delete all the old source files - download the new file using: cvs checkout zaptel zapata libpri asterisk - compiling the new source file using: make clean ; make install Everything is compiled fine, except.... when I stop and start Asterisk server, it is still the old CVS version:

On 1/14/19 4:02 PM, Duncan Turnbull wrote: > > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> wrote: > >> Duncan: >> >> You may have it right—I took one phone and set the ring time to 60 >> seconds. I now get about 4 rings on that one. >> >> I wonder how I

Hi, I am using Asterisk 1.4.25. I have one Broadvoice account. I Integrated this broadvoice account with Asterisk Server. I am Able to Make calls but cannot recieve calls. In Incoming calls, call lands to SIP extension, as I attend the call....It gets hungup......... If i dont transfer this call to extension or I play any file then it works OK. But as I transfer it to SIP Extension it get

A call comes in and goes into the queue, the queue dials a sip channel using a macro. The macro plays a set of options to the callee and if the callee presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends. For some reason the caller goes back into the queue rather than continueing on in the dial plan. Why is this, i could have sworn in 1.2 if i set MACRO_RESULT=CONTINUE that the

Hello: Is it possible to run SAMBA over IPX/NetBeui? (I will thank any suggestion) I hava an small LAN of 3 computers, connected to the Internet through a Cable-modem with DHCP (the cablemodem and the computers are all connected to a hub). The CableModem assigns every computer a random public IP adress, but usually from diferent range (ie: 212.78.x.x, 62.57.x.x, etc), so I cannot get SAMBA or