a software based private branch exchange (PBX)

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2
votes
1answer
23 views

Asterisk: Make IVR to say numbers during user typing

I want my IVR to say numbers during the Read command. This means, that i dont want to say the numbers after the user type, but during the user type. For example, enter your user code: 124567# ...
0
votes
1answer
14 views

Asterisk IVR call to external IVR in other Asterisk machine via IP

It is possible to write a IVR script to call another IVR in other network by ip? What command should i use? I am trying this: Goto(from-internal-custom,1235,1@other_asterisk_ip) It is not ...
-1
votes
0answers
23 views

Unable to connect to remote asterisk

I am using CentOS 7 (Latest Veriosn). I have installed Asterisk 13 on my CentOS. Now I am trying to start the Asterisk by using this command #asterisk -rvvvvvvvvvvvvvvv but no luck,I am getting ...
3
votes
0answers
21 views

how do I make a SIP phone ring with sipsak?

How can I use sipsak to simulate a call so that there's an actual ring, or possibly even a sound is played? thufir@mordor:~$ thufir@mordor:~$ sudo sipsak -v --usrloc-mode --from sip:demo_alice@...
1
vote
0answers
23 views

How can sipsak simulate a call with usrloc?

Building on a stackoverflow question, how can I actually get the device to ring? thufir@mordor:~$ thufir@mordor:~$ sudo sipsak -vv -s sip:6003@localhost No SRV record: _sip._tcp.localhost No SRV ...
0
votes
0answers
33 views

How can I use sipsak to send a SIP Invite to lrnlookup.telnyx.com with sipsak?

Here's what I have so far: thufir@mordor:~$ sipsak --from sip:thufir@getonsip.com --password flFl24GB5 --invite-mode lrnlookup.telnyx.com --message-mode error: for the USRLOC mode you have to give a ...
0
votes
1answer
28 views

Missing NIC in CentOS 5 - AsteriskNow 1.7.1

So I have two nics, but I can't get the second to load, here is the output from lspci -vvv 04:00.0 Ethernet controller: Intel Corporation 82574L Gigabit Network Connection Subsystem: Super ...
0
votes
1answer
221 views

Asterisk v13 on Kali Linux: No RTP engine was found. Do you have one loaded?

I installed Asterisk and tried to make a call with zoiper but I get an error [call failure 401 forbidden] and Asterisk return this message : Connected to Asterisk UNKNOWN__and_probably_unsupported ...
1
vote
0answers
51 views

Another problem installing Asterisk on kali linux

I am trying to install Asterisk [version 13] on Kali linux. I first install DAHDI, but I won't install libPRI. So when I install asterisk, I use: ./configure and make menuselect Till now ...
0
votes
0answers
183 views

Problem installing Asterisk on Kali linux

I am trying to install Asterisk on Kali linux. I first install DAHDI, but I won't install libPRI. So when I install asterisk, I use: ./configure and make menuselect It's ok! But when I ...
2
votes
2answers
3k views

Unable to install dahdi - /usr/bin/install: cannot stat ‘’: No such file or directory

I am trying to install dahdi in my centos 7 system using this link but having some errors [root@localhost ~]# cd /usr/src/dahdi-linux-complete* [root@localhost dahdi-linux-complete-2.11.0+2.11.0]# ...
0
votes
1answer
146 views

Asterisk: Using extension patterns with call files

I'm trying to make a script that makes call files for Asterisk to execute, however I want the extensions within the call files to match my extensions's patterns. Call file: Channel: trunk/10 Context:...
0
votes
0answers
82 views

How to Configure freepbx and Matrix Eternity on PRI Line?

I need to configure freepbx with matrix eternity on a PRI line and setup 12 seats for making calls outside. Matrix Eternity Device is http://image.slidesharecdn.com/...
1
vote
0answers
985 views

lpc_ich: Resource conflict(s) found affecting gpio_ich

I use Asterisk 11 in my Ubuntu Server 14.04, but I Have some problems with my Dahdi Driver, with conflits Please see the dmesg (null) [ 11.804460] Adding 2052092k swap on /dev/mapper/...
0
votes
1answer
318 views

Asterisk:Registration is UNREACHABLE VOIP chan_sip.c:15171 sip_reg_timeout

I'm trying to register my VOIP but I can't and appear this massege [Jul 24 12:45:58] NOTICE[24430]: chan_sip.c:15171 sip_reg_timeout: -- Registration for 'XXXXX@sip2.tellfree.net' timed out, ...
0
votes
1answer
403 views

How to install Wildcard TDM410P to my Debian?

I want to install a WCTDM/0 "Wildcard TDM410P in my Debian 7 for I use in my Asterisk, but I can't it. And it appear DAHDI_SPANCONFIG failed on span 1: Invalid argument (22). I have to configure my /...
0
votes
1answer
124 views

Asterisk astdb File increases size!

I am using asterisk 1.2.37 with FreePBX 2.6.0 and i am having an issue that I am not quit understanding. The Asterisk database file /var/lib/asterisk/astdb increases his size exponentially at least 1M ...
0
votes
1answer
277 views

asterisk dial hangup direction - Is there a variable I can use to record the direction of hangup?

Is there a variable I can use to record the direction of hangup? Example: Calling ----> Called. Called party hangs up. Info I want --> Release from Called party Example 2: Calling ----> Called. ...
3
votes
2answers
356 views

Script to hang up Asterisk channels

I am trying to create a script for Asterisk. We use a lot of channspy. Sometimes the channspy stays opened and didn't hangup even if the softphone is hangup (I don't understand why). So I thought of ...
0
votes
1answer
289 views

Error: you do not appear to have the sources for the 2.6.32-042stab102.9 kernel installed

I've just bought a new VPS with Cent OS 6.6 installed. I'm attempting to install Asterisk 11 on this VPS via command line remotely. I've used the directions here however I get this error: you do not ...
1
vote
2answers
43 views

iptables for Asterisk

I am running Asterisk with FreePBX. I would like to log all failed attempts and after a while ban them. Ideally after 3 failed attempts in 1 minute, ban for 1 hour. Is it possible to see if this ...
1
vote
1answer
285 views

SIPp “hello world” message to Asterisk

How do I run diagnostics against Asterisk? Asterisk is running on tleilax; and doge is on the same network (My network topology isn't optimal). Specifically, I want to do something like: sipp 345@...
2
votes
1answer
140 views

asterisk network topology — KISS

is this the typical Asterisk topology??? On my home system, I have the Asterisk server, tleilax, and the client, doge, connecting into my router -- so that they're both on the same network. The IP ...
0
votes
1answer
345 views

how to connect a softphone to asterisk

To give context, here's my network: On the tleilax server, there's clearly a result from sipsak on doge: tleilax*CLI> tleilax*CLI> core show version Asterisk 1.8.29.0-vici built by abuild @ ...
0
votes
2answers
1k views

Compiling Asterisk on Debian: Cannot Find `ptlib-config`

My Asterisk ./configure is returning: checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking /...
2
votes
2answers
75 views

Configure PBX to forward calls to my phone only when home

I want to configure a PBX (Yate or Asterisk or SIP Witch) so that when I'm away, it takes the call and tells the caller I'm not home, but when I'm home it forwards the call to my phone, which I ...
1
vote
2answers
538 views

Issue in Asterisk Performance

I am running FreePBX on DigitalOcean VM. I am facing some call dropping issue in Asterisk. When looking at "htop", "top" and FreePBX GUI, I see different results of CPU utilization and can't ...
2
votes
0answers
31 views

Top show high CPU on process but idle on load average [duplicate]

I am running FreePBX on DigitalOcean VM. I am facing some call dropping issue in Asterisk. When looking at "htop", "top" and FreePBX GUI, I see different results of CPU utilization and can't ...
2
votes
1answer
362 views

How to Originate Call from Polycom Intercom via Dialplan

We have polycom phones for use with our Asterisk 1.8 PBX. Consider this: On the phone at extension 2536, I dial 3333. This should originate a call from extension 2500 to play a sound. In other ...
0
votes
3answers
7k views

Why can I not install ncurses development package?

I am trying to configure Asterisk 12.3.2 on Debian 6.0.9 but I am getting an error: "configure: error: *** termcap support no found (on modern systems, this typically means the ncurses development ...
2
votes
0answers
158 views

VoIP dialer for SIP

I have some Java, so am biased in that direction, but not overly so. None of the Java suggestions: http://stackoverflow.com/questions/16471795/sip-client-for-java http://stackoverflow.com/questions/...
0
votes
2answers
242 views

Can Asterisk play sounds over a Marantz sound system?

We are running Asterisk 1.2 (working on an upgrade) and in another building have a Morantz sound system that is connected to the same subnet as Asterisk. Can I use Asterisk to play a wav file over ...
2
votes
2answers
3k views

Creating softlinks to header files that are not in the expected directory

3.12.6-300.fc20.x86_64 #1 SMP Mon Dec 23 16:44:31 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux Fedora 20 I am trying to install asterisks from source and when I run the ./configure script I get the ...
3
votes
1answer
212 views

Asterisk, GSM-FXO gateway example of usage

Can anyone explain where to use GSM-FXO gateway. I understand where to use GSM-FXS - for example to connect analogue telephone to gsm gateway. But do we use GSM-FXO to connect to two telephone ...
0
votes
1answer
323 views

Bad SIP URI in To header field

We are creating Asterisk<->Lync trunk. The Asterisk is under providers control. I am in charge of Lync. I'm slightly familiar with Asterisk. We created a trunk. Everything was fine until provider ...
0
votes
1answer
337 views

Making a tty9 on a openvz VE

I am in a virtual environment. I am installing Asterisk and freepbx. They use this script amportal and it has a requirement to check for tty9 there is alot of mention on folks running this in a VE ...
0
votes
0answers
230 views

PBX randomly loses network connectivity and restores itself

I'm using a QBX appliance, which is just a microATX desktop system with FreePBX connecting to asterisk and some other slapped together stuff running on CentOS 5.4. It's horribly out of date (which I ...
4
votes
1answer
6k views

How to configure sip trunk with different host details in Asterisk

I've read every forum on here, asterisk.org and google about this matter and still can't get it right. Here are the the SIP details. SIP Domain sip.provider.com:5060 Outbound Proxy sip10.provider.com:...
0
votes
1answer
367 views

Unable to find an option to install CentOS 6.4 (64bit) in command line (linux text) for Asterisk

I am trying to learn Asterisk, so I downloaded the latest CentOS 6.4 (64bit) to install on VMWare Workstation 9. When I started the installation it is not giving any screen to start the installation ...
1
vote
1answer
370 views

Can asterisk set-up a three-way call on an analog (classic) line?

I would like to know if it is possible for asterisk to establish a three-way call on an analog line. Typically, I would like asterisk to call 2 numbers, with the same telephonic line, and connect ...
2
votes
1answer
739 views

Machine Check Exceptions with Wildcard TE420 and Asus P5N-E SLI

I have been seeing machine check exceptions when trying to call dahdi_cfg (or, sometimes, even on just loading the kernel module) to configure a Digium Wildcard TE420 Quad-Span T1 PCI-E card. Running ...