Asterisk is software that enables a server to act as an IP PBX system, VoIP gateway, conference server, and more.

learn more… | top users | synonyms

0
votes
0answers
9 views

integrating asterisk with Ravenclaw dialog system

I want to integrate Asterisk with AudioServer in Ravenclaw dialog management system, which is running on Galaxy Architecture. The purpose is to integrate voice calls with the dialog management ...
0
votes
1answer
119 views

Asterisk, How can i play an audio file [closed]

Here is the dial plan [testInComingCalls] exten => s,1,Answer exten => 30953025,1,Dial(SIP/20000,20) I would like to play an audio file as soon as somebody answered the call (Welcome Message)..
0
votes
0answers
147 views

Elastix Tranfser config files

I have installed Asterisk PBX to a new machine. What I want is to transfer configuration files from my running Asterisk PBX to the new one. I do not prefer the solution to backup my running Asterisk ...
0
votes
0answers
279 views

Asterisk: Retransmitting again and again

I have following sip settings Name : external Description : Realtime peer: No Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : public ...
0
votes
1answer
93 views

Empty Asterisk 12 CDRs in SQLite3 database

Using: CentOS 6 Asterisk 12 cdr_sqlite3_custom.so cdr_sqlite3_custom.conf (deliberately simplified): [master] table=>cdr columns=>calldate, clid, test ...
1
vote
1answer
107 views

Security, hardening and NAT for RTP and SIP

I decided to create a VOIP server and it has gotten more complicated then I thought it would. So in order to make my server more secure, I keep it behind a pfSense appliance. I use IP filtering to ...
0
votes
0answers
25 views

Asterisk pass-through fax to another Asterisk

I'm having two asterisk machines ("A" with external IP and "B" - local), and one of them (A) is like voip gateway. I have DID (for example 1(234)567-89-00) on asterisk "A" and I want to receive a fax ...
0
votes
1answer
838 views

Asterisk AMI - get detailed extension status

I am using AMI from Python. I guess the same can be achieved from the Asterisk CLI. I want to be able to get details about every extension in the PBX. More specifically, for every extension I want to ...
0
votes
2answers
154 views

asterisk - use a different “SIP CoS mark”

On my asterisk server, when i do a sip reload, i get the message "Using SIP CoS mark 4", followed by a registration time out. I need to make that statement say "Using SIP CoS mark 5". How do I ...
1
vote
0answers
118 views

Asterisk congestion reasons

After moving Asterisk from one server to another, I noticed that ASR dropped by 15-20 %. Also I noticed following warning in logs file [Dec 8 15:52:00] WARNING[4771][C-00000031] pbx.c: Maximum ...
0
votes
3answers
107 views

What is the use of echo canceler in ISDN boards if all my endpoints have echo canceling?

Most digital telephony boards (BRI/PRI) have optional hardware echo canceler presented as alternative to the cpu intensive software echo canceling available in Asterisk and FreeSWITCH. I'm ...
1
vote
0answers
121 views

DTMF Detection in Asterisk PBX

I have an asterisk server as IP PBX. I have created an Auto-Attendant for my company and asking users to enter desired extensions when they call. The problem is that when the message is playing and ...
0
votes
1answer
187 views

Asterisk realtime table for confbridge app

I try to configure asterisk with realtime config (configuration stored in mysql database), I configured my IPBX but I don't find a solution for conference application, I found only the table structure ...
0
votes
1answer
43 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
0answers
92 views

Nagios nrpe commands error out with asterisk

as the nrpe user when Command: /scripts/nagisk.pl -c version Outputs: Asterisk 1.8.11-cert10 Using nrpe Command: /usr/lib/nagios/plugins/check_nrpe -H [HOSTNAME] -c check_asterisk_version Outputs: ...
-1
votes
1answer
105 views

An extension in use receive more than one call with leastrecent strategy

I have a queue with some members. The strategy is leastrecent. Example: Members: 101, 102, 103, 104 A call goes to 102, and 102 answer the call, and then, the ext. is in use. Another call go ...
0
votes
0answers
64 views

Can't listen to port 4445; sqlite3

So I have a dedicated machine running Ubuntu 12.04 server with FreePBX 2.9. Earlier I was attempting to familiarize myself with SQL via sqlite3, but I received this dreaded error: SQLite header and ...
0
votes
0answers
73 views

How to display incoming trunk name at called device end with Asterisk?

We have an Asterisk with FreePBX, and using Grandstream and Panasonic VoIP phones. Now when an incoming call rings on the phones only displays "Line 1" (as it is configured at account 1 on phone) ...
0
votes
1answer
106 views

Asterisk Register username with special character like “@”

I am using a SIP provider that has provided me with a username like: +1122233344@aaa.bb.com (Note this is only the username part) And has a numerical password. My Register string looks something ...
0
votes
0answers
19 views

Alarm when a queue is unattended

Is there a way to make a trigger/alarm when a queue is left with no one to attend it? My setup is freepbx and CDR-stats, but it could be a plain Asterisk solution as well.
2
votes
0answers
48 views

How to set Mitel 5224 ringer volume in configuration?

We have several Mitel 5224 SIP phones, but can't seem to find out to make the ringer volume setting persist after it reboots. Can the ringer volume can be set somewhere in the XML configuration file?
0
votes
0answers
68 views

FreePBX / Asterisk queue leastrecent Breaks on transfer

I am running into some weird issue's with queue's, It almsot seem's as if when you transfer, or go on hold it does not reset the lastcall time Scenario – John is on a call with a customer. Mary is ...
0
votes
0answers
181 views

DRBD show StandAlone on Primary node after secondary node attach with it?

I am using Centos5.4 & DRBD8.2.6 heartbeat2.1.3 & one java program with Mysql & Asterisk1.6.2.6. On DRBD disk i put 2 service Asterisk & Mysql. SO if DRBD Disk not up both services ...
1
vote
1answer
394 views

Limit simultaneous call per user in Asterisk

I want to limit simultaneous calls per extensions in Asterisk for security reasons. For example when a user is on the call no body else would be able to make a call by that extension. How can I ...
1
vote
2answers
353 views

Asterisk issue with ConfBridge

I'm receiving an error when I try to call an extension that leads to ConfBridge() app in my dialplan Here is the extensions.conf exten => 601,1,Noop() same => ...
0
votes
0answers
188 views

Asterisk Call pickup group picking up wrong call

While executing call pickup within a call pickup group via the soft button on our phones the wrong phone call is picked up. Example: phone A calls phone C and phone B tries to pick up the call. Phone ...
0
votes
4answers
779 views

CentOS 6.4 running Asterisk 11.5 with high load, low CPU/memory/disk IO

I have an Asterisk server doing automated calls and I'm noticing an unexplained high load in it. The server is only running Asterisk. Database and other support applications run in a different ...
1
vote
0answers
240 views

Error raised during calls in asterisk server

Today my asterisk could not receive calls for a while. when I checked the asterisk logs, the error below was raising during this period. Purely numeric hostname (1002), and not a peer--rejecting! ...
1
vote
0answers
64 views

Change the caller id from my asterisk server

I successfully set up an asterisk server. When people call my asterisk server via PSTN, the server will place another PSTN call to my phone at 33344455555. When I receive the call, my phone shows ...
0
votes
1answer
307 views

Can't receive phone call on my FreePbx extension

I'm running out of hair to pull here. I have my freepbx installed and configured ok at the beginning. I was testing it with Google Voice account. I can call out and receive call. I'm sure I must did ...
1
vote
0answers
85 views

How can I use the FLASH key of my IP phone to transfer a call (atxfer)

My IP phone has a FLASH key. This key sends the DTMF of Flash [16 in decimal] (http://www.voip-info.org/wiki/view/SIP+DTMF+Signalling). How do I set the atxfer parameter in the features.conf?
0
votes
1answer
262 views

Asterisk behind a firewall

we have a bit a sticky problem on our Asterisk server, which we are struggling to resolve, I'm hoping someone with more knowledge than me can help. We are running Asterisk 1.8.23.0 on Centos 6.4 and ...
1
vote
1answer
75 views

Installing FreePBX or Asiterisk-GUI on working server

We have server with Ububuntu and asterisk 1.8 on it without any web-interfaces. Our managers ask me to make them web-interface for looking over config, users, calls and so on. I've installed ...
1
vote
1answer
954 views

How to configure a 2821 ISR to an asterisk PBX on a PRI line?

I'm trying to move my company's phone system from Cisco's CUCM to an asterisk solution. I have setup a PBXinaflash server, and setup an account with www.vitelity.com. I was able to get it working. ...
1
vote
1answer
146 views

Asterisk ignoring sip_domain part of URI

I am using Asterisk version 11.5. My problem is the following: I literally cannot make any outbound call. That's because, whatever I dial in any client, the sip_domain part gets ignored, it is ...
0
votes
1answer
45 views

Advice respect an architecture PBX [closed]

i'm new on the world of PBX and network infrastructure. I worked in a model and wanted to know how feasible it is to implement it in a production environment. The requirement is to create a solution ...
0
votes
1answer
75 views

How do I set the current working directory of Asterisk?

I need to be able to collect core dumps, but since Asterisk's CWD is / (so claims procfs), it'll never be able to write them. I've confirmed my suspicions by allowing world write to / and SIGABRT the ...
0
votes
0answers
361 views

Avaya-Asterisk call drops immediatly via SIP (sip:anonymous@anonymous.invalid)

I'm trying to call from Avaya to Asterisk (11.5.0) in SIP. Avaya calls to Asterisk, then Asterisk rings my softphone but when I answer the call (in the softphone), call drops immediately. This is the ...
0
votes
3answers
132 views

make web portal visible on two different vlans on same interface

We have recently set up a new Aterisk pbx server based on our previous pbx (also Asterisk) which was installed by an external company. We've kept the configuration of the server the same, all we did ...
1
vote
1answer
181 views

Asterisk PBX on linux (CENTOS 6.2) hangs up calls that are being recorded after 33 seconds

i have an issue with my calls being made via ASTERISK PBX running on linux CENTOS 6.2. The use case is that a call is triggered from /var/spool/asterisk/outbound/ The caller is dialed The dialplan ...
0
votes
2answers
768 views

High Load, low CPU, Memory and Disk IO - Highend Server

This issue has been bugging me for the past several days having spent over 40 hours investigating this issue intensively. Effectively we run asterisk 1.4.42 which I understand is old, however is the ...
0
votes
1answer
1k views

Asterisk and websocket: “rejecting secure audio stream without encription details”

I'm trying this library using the test service: http://tryit.jssip.net/ When I register, using WS, not WSS, peer is registered succesfully When I make a call i got "rejecting secure audio stream ...
2
votes
1answer
66 views

Asterisk console output 'no button pressed' on no DTMF input

I've got an outbound survey program, here is an example of what the survey does from the dialplan. same => n(xtn_age),Read(AGE,rbp_age,2) same => n,Set(CDR(age)=${AGE}) same => ...
0
votes
2answers
2k views

How to install Asterisk with SRTP support?

I'm rather new to Asterisk, and I need my server to support WebRTC. As far as I know, Asterisk version in Asterisk Now is compiled without SRTP support, which is necessary for WebRTC. So, I try to ...
0
votes
0answers
161 views

using iptables to secure asterisk

Following the post at http://www.dslreports.com/forum/r26964868- I created some IP tables for my asterisk install. The server is behind a router and the router only had 5060 and the RTP ports ...
1
vote
3answers
329 views

Asterisk and SIP trunking, is this configuration possible?

I am new to asterisk and before I dive in, I just want to make sure that what I plan to do is possible/correct. My office will run an asterisk server and have both local and remote extensions. We ...
1
vote
2answers
825 views

Interpreting and using the Asterisk “timing test” command

Timing is very important for certain kinds of applications in Asterisk. If DAHDI is the timing source, the dahdi_test command can be used to check the timing provided by the DAHDI kernel module. If ...
1
vote
1answer
166 views

Asterisk: Forcing a sip peer to connect via ipv6?

I've got an asterisk server that connects to an upstream provider over a WAN. The upstream provider supports both IPv4 and IPv6 connectivity, and the asterisk server is behind a NAT. When asterisk ...
0
votes
1answer
505 views

Get details about “Remote UNIX connection disconnected”

I am getting lots of: -- Remote UNIX connection disconnected I would like to get more details about it. How do I get information about who is connecting, ip address, and more details?
0
votes
1answer
176 views

Monit detecting error, but not restarting asterisk

I am running Ubuntu 13.04 with the corresponding versions of monit (5.5-6) and asterisk (1.8.13.1). I have monit set up to watch my asterisk log file for disconnection to my SIP provider and restart ...