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

1
vote
2answers
850 views

Asterisk/TrixBox and Predictive Dialing/Outbound Call Center

Is anyone out there running predictive dialers or Call Center type products on Asterisk/Trixbox. I'm looking for vicidial type functionality. The best way to characterize it would be political ...
1
vote
0answers
23 views

Asterisk - very high Load Average with no reason

There is a problem with Asterisk server showing high Load Average I can not explain. Ny setup is: Asterisk 13.1.0 Linux 3.13.0-24 (Ubuntu Server) Dual socket (Xeon E5-2620) server, HT enabled - 24 ...
1
vote
0answers
48 views

sip register success on one host but fail on another host in same local network

I acquired antisip account and setup asterisk (lets say PBX-A) to register. It worked fine. However, I attempted to do so with another asterisk (PBX-B). No luck this time. PBX-A and PBX-B are on the ...
1
vote
0answers
17 views

Log to file in Asterisk: append with newline

(I am testing on Asterisk 11.7.0~dfsg-1ubuntu1) I am using the following format to append to a logfile, according to the documentation: same => n,Set(FILE(/tmp/mylog.txt,,,a)=my-log-message) ...
1
vote
1answer
32 views

Asterisk Event PauseQueueMember - AMI Calls

We use Asterisk-AMI and we have a little JS server which listens for Events from the AMI to gather & display statistics (Calls, Queues etc). I initialy thought that when a queue member is paused, ...
1
vote
1answer
200 views

Asterisk can not send fax via T.38

I am a newer on asterisk. when I tried to send fax, I can not get success. After doing a lot of reseach, I decide to ask my question at here. I hope someone could give me some idea to correct my ...
1
vote
3answers
216 views

Asterisk ring out even when the client is offline

I need to configure an Asterisk box to go to voicemail but only after X ringouts. exten => 1234,1,Dial(SIP/ivan, 30) exten => 1234,2,VoiceMail(777@mb_tutorial) exten => ...
1
vote
0answers
531 views

Vicidial in Autodial does not connect correctly dongle phone call to agent conference

I want to make work Vicidial with chan_dongle Asterisk module to use Huawei USB modems as GSM gateways (trunks/carriers) for asterisk. In manual mode everything works. Problem appears only if ...
1
vote
1answer
650 views

Asterisk AGI randomly returning 4 and failing to execute PHP AGI

I've got a PHP script that I execute 2-3 times throughout a single calls life time, if it is inbound it will execute once for the queue, once for start of call and a last time for end of call. For ...
1
vote
2answers
1k views

Asterisk IVR setup: disallow direct dial for some extensions?

We've got PBX to configure. It's based on Elastix 2.4.0, freePBX 2.8.1 and Asterisk 1.8.20.0. Currently we have a setup for inbound calls like that: Inbound call > Announcement (30 sec) > IVR ...
1
vote
1answer
43 views

SecAst: Configuring Email Alerts for Google Apps Account

Requesting assistance with configuring SecAst email alerts to work with Google Apps hosted account (posting here as per request by Generation D)... We have SecAst installed on a system with a new ...
1
vote
1answer
161 views

Asterisk Penalty for dynamic Agents

Is there a way to use penalty with dynamic agents to order agents call distribution for received calls in a queue? We are using linear ring strategy and this only order calls to dynamic agents by ...
1
vote
0answers
149 views

Werbrtc2sip crashing intermittently. Aborted (core dumped)

I am having issues with making webrtc2sip actually work. Webrtc2sip is now getting crashed intermittently during a call. I have seen this even when only one call is active in a queue. By the way, my ...
1
vote
0answers
130 views

Connect Cisco 7975 VOIP phone to two separate Asterisk PBX's

I have a Cisco 7975 registered on an Asterisk SIP server, and I want to register a second line on a second (different IP) Asterisk SIP server. I tried to do this with firmware version 8.5.3 but it ...
1
vote
1answer
233 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 ...
1
vote
0answers
248 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 ...
1
vote
0answers
216 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 ...
1
vote
1answer
815 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
0answers
634 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
102 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 ...
1
vote
0answers
127 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?
1
vote
1answer
2k 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
3answers
452 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
0answers
354 views

UDP Range Nat Translation port forwarding RPT Traffic to inside host PBX Cisco ASA 5505 8.2a

I have an Asterisk based PBX that I'm providing VoIP services with, My gateway is a Cisco ASA 5505 running 8.2 and I only have one public IP Available along with other services I provide with that IP. ...
1
vote
1answer
1k views

Accept calls on external cell phones - FreePBX Distro

We installed FreePBX distro last week and configured it so that when someone calls in, it goes to all the phones so someone here can answer the call. After trying all the internal extensions, it ...
1
vote
1answer
460 views

Asterisk outbound calls hanging up imediatellly on Zap trunks

The inbound calls work fine. Outbound calls fail as follows: Dial("SIP/xxxxxxxxx","ZAP/g0/14442223333|300|") Couldn't call g0/14442223333 Hungup 'Zap/2-1' Everyone is busy/congested at this time ...
1
vote
1answer
412 views

How to establish SIP connection, when SIP-proxy is required?

I have Asterisk/1.8.13.1 Asterisk GUI-version : SVN--r Yes, quite old one, but I can not update it since this is installed on my Synology NAS. NAS is connected to internet thru router Asus RT-N16. ...
1
vote
0answers
484 views

Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic ...
1
vote
2answers
637 views

How do I detect call forwarding in Asterisk?

Basically, I want to do the same thing that Google Voice does. I forward my calls to a DID number that rings my Asterisk server via IAX2, which, if it detects the call has been forwarded, sends it to ...
1
vote
2answers
816 views

FreePBX How to play an announcement for misdialled calls

Our Asterisk-based PBX (FreePBX "Tango" distro) does not require any prefix for outgoing calls. The system is able to recognise external numbers and route them appropriately. Some of our users who ...
1
vote
0answers
327 views

Asterisk 1.8, Dedicated SDSL but SIP Provider Inbound Channel heavily delayed?

We have a dedicated Asterisk 1.8 server running as a front end SIP server for authenticated and internal users only, using SDSL 2MB/2MB, in the UK. The provider of our internet also is the provider of ...
1
vote
1answer
290 views

asterisk call file not deleted

I'm placing outgoing calls using Asterisk call files on Asterisk 1.8. My call files are very simple and have no MaxRetries or RetryTime specified. If a call succeeds, the corresponding callfile is ...
1
vote
1answer
601 views

Asterisk Call Center Report - Inbound and Outbound Calls

We currently use Asterisk to run our call center which handles incoming calls (both direct dial to extension and queues) as well as a bunch of outbound calls. Up until now, we've used a custom report ...
1
vote
1answer
70 views

Asterisk (Trixbox CE) sound to low in message2email

Hy in Asterisk (Trixbox CE) latest version, we receive message to our email in WAV. But the sound it's too low. Where can I fix that ? Thakns
1
vote
3answers
2k views

How many concurrent calls can a 0.9Mbit connection handle with G711a?

We've been using G729 but the call quality was terrible. I swapped over to G711a and everything was perfect. We've got about 0.9mbit upstream. What's the maximum theoretical number of calls, and how ...
1
vote
0answers
150 views

asterisk Dial-plan?

I want to make a dial plan for asterisk to do this: for incoming-call check the caller-id and if caller id is equal with specific number (for example 666) then hangup that call.(this dial-plan also ...
1
vote
0answers
2k views

Configuring Hylafax to work with Asterisk over a single SIP provider

I'm setting up an Asterisk 1.8 based PBX, and as part of the requirements, I need to make provisions for sending and receiving faxes. The 'big picture' goal is installing a fax printer at each Windows ...
1
vote
1answer
376 views

Problems in Asterisk when configuring x-lite

I setup AsteriskNOW 1.6 with freePbx. I assigned it an address 192.168.164.128, and I made a team on a virtual machine (vmware). There are 2 clients: Windows XP the AsteriskNOW server I logged ...
1
vote
2answers
2k views

How to enable CDR on AsteriskNow 1.5

I have upgraded PBX to Asterisk 1.6.2.7 and now CDR files are not created. It looks that such logging is disabled: Connected to Asterisk 1.6.2.7 currently running on pbx2 (pid = 5824) Verbosity is at ...
1
vote
2answers
1k views

DNIS Number In Asterisk Dialplan

I am trying to get the phone number customers dialed when calling in. I have tried all of the CALLERID options available but to no avail. Any ides?
1
vote
0answers
41 views

add presence for a remote user to a legacy telephone system?

we have a small call center that uses an old nortel phone system with analog lines. one of our sales people works from home so her calls do not go through the phone system. this creates a problem at ...
1
vote
0answers
290 views

How to send T.38 from a mac?

I'm trying to set up a fax-server on a macintosh. I have Hylafax, and we're going to use an internet FOIP fax provider (Haven't decided who yet, that may be another question). The problem is how to ...
1
vote
1answer
551 views

Asterisk call failure timeout

I am using Asterisk and there seems to be a problem with the delay that the server answers a call. What actually happens is that the caller hears a busy tone and then hangs up, but a channel is opened ...
1
vote
1answer
829 views

Calls cut off in Asterisk 1.6.2

We are offering hosted VoIP. The Asterisk servers are running in our data center and customers connect their Aastra phones remotely to it. However, I have one customers connected to his own PBX and he ...
1
vote
1answer
381 views

Where can I get some real life CDR/SMDR/AMA logs from asterisk or ther IP based switch?

In order to develop some generic call accounting software I need some sample logs from IP based PBX. Where can I get some? - I need more than 100 records, preferably using both SIP and telephony ...
1
vote
1answer
1k views

How to configure Gizmo5 on Trixbox/Asterisk

Does anyone know how to configure Gizmo5 for use on an Asterisk PBX? Specifically Trixbox if possible. For the life of me I just can't get this to work...
1
vote
2answers
666 views

elastix cdr stop working

CDR was working before 19 march. Unfortunately i dont remember what kind of changes i made to configuration, but this exactly not changes to CDR config. elastix 2.4.0 asterisk 11.7.0 mysql 5.0.95 ...
0
votes
1answer
794 views

Permission Denied trying to execute a .conf file.

I'm using a virtual machine in Windows 7 and I have installed Linux on it and Asterisk software. I want to configure the sip file to create sip user but when I type ./sip.conf in Linux it replies ...
0
votes
2answers
817 views

Timing Source on Asterisk [closed]

I have tried to look for simple language of explanation what is timing source for in asterisk? and why it has significant impact on the asterisk Anyone can give me insight will be appreciated
0
votes
2answers
226 views

How do you set up speed dial on Polycom IP6000 phones?

Basically I would like speed dial on this so I can press 1 and it will dial 101 AND I can press 2 and it will dial 555-555-5555. Is there a way to do this?