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

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20 views

Asterisk PBX Manager not saving changes

We are implementing Asterisk 11.4 and using the native PostgreSQL database to store call data. So far, we have not been able to change the admin password through the PBX manager. The password gets ...
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45 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?
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109 views

Polycom IP 650 Custom display

I've been googling all over the place and haven't found a good way to do this. I'm hoping that someone out there has done something like this. I have Polycom 650's and 331 deployed to my users via a ...
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201 views

Asterisk ${CALLERID(name)

I am working with asterisk (Asterisk 1.8.11.0), freepbx (2.x) and was building a dialplan (extension_custom.conf). I wanted to get caller Name from database which i have stored for example. CLI> ...
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312 views

How to forward a call to SIP server

I have a SIP server(Asterisk) I have configured a DID for outbound and inbound. My asterisk server has freepbx installed to manage it. My requirement here is, when anyone calls to this DID call should ...
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54 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 ...
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86 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 ...
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105 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 ...
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109 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 ...
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203 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! ...
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56 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 ...
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78 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?
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272 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. ...
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427 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 ...
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297 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 ...
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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 ...
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131 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 ...
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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 ...
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276 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 ...
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10 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 ...
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29 views

Switch from E1 to Analog PSTN line on IP-PBX failure

I have two Asterisk servers, E1(PRI) Gateway Asterisk 1.4 with E1(PRI) card IP-PBX Asterisk 1.8 on VM Server When my e1 gateway is failed (ex. shutdown, no power...), PSTN automatically switch to ...
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45 views

Iperf UDP listening on used port

I get WARNING[9157] chan_sip.c: Retransmission timeout reached on transmission error on asterisk and trying to troubleshoot the problem. Now asterisk is listening on 5060 port: udp 0 0 ...
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11 views

Adding milliseconds to Asterisk v1.4 log

I'd like to add milliseconds to the timestamp for an Asterisk v1.4 instance. I currently have the following entry in /etc/asterisk/logger.conf: dateformat=%B %d %H:%M:%S I'd like to add ...
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24 views

Asterisk SendText application

I am trying to send text message to the callee using the below config in extensions.conf exten => 5000,1,Answer() exten => 5000,2,SendText('hello world') exten => 5000,n,Hangup() And I am ...
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38 views

Asterisk virtual machine stops responding to network connections when I connect with Phoner Lite

I've installed Asterisk Now with PreePBX from bootable CD to a VirtualBox machine. Then I updated through FreePBX all the modules available for update and noticed a strange behaviour: After I ...
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472 views

exit: 1 [FATAL] Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting

I am struggling with this error after an Elastix upgrade: exit: 1 [FATAL] Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting From what I can see the manager is ...
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201 views

Asterisk Real time does not perform register with SIP TRUNK provider

I am using astersik real time (dynamic). I have entered my SIP trunk details into the sippeers tables. However, the sip trunk does not perform a register with the SIP TRUNK providers servers as it ...
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22 views

check if CCSS callback is available

I've just discovered Call Completion Supplementary Services (CCSS) Is there a way for the dialplan to know if callback is available? If a call terminates with BUSY or NOANSWER, I'd like to offer ...
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50 views

Connect Cisco IP phone on two Asterisk sip servers

I have a cisco ip phone with sip firmware v9.3.1. I want to connect two lines on two seperate Asterisk sip servers. In section i can only define USECALLMANAGER (in node) and not the IP of an ...
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136 views

Soft phone not getting register to my asterisk server

I have config in my local lan asterisk for test. which is working fine. With the same OS version and packages on "AWS Cloud" I am setting as following. But it don't get register "error = sip 408 - ...
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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 ...
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120 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 ...
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247 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 ...
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14 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 ...
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81 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: ...
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52 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 ...
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62 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) ...
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17 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.
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62 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 ...
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152 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 ...
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156 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 ...
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342 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 ...
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147 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 ...
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38 views

Append DID identifier to Caller ID?

Our company has two branches - a sales branch and an engineering branch. The sales branch has a different number and a different branded name than our company, and our receptionist needs to know ...
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479 views

Not able to register via SIPclient on asterisk server?

I'm unable to register a SIP client on my Asterisk server. What changes do I need to make to my Asterisk server configuration to fix this? Here are my configuration files for Asterisk. sip.conf ...
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371 views

LYNC 2010 Dial-In in Meeting DTMF issue

We are facing an issue in the LYNC2010 dial-in to a meeting. We redirect an Asterisk number to LYNC, whitch connects successfully in the dial-in plan of LYNC. After calling from external network to ...
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56 views

Cannot recover extension in Trixbox

I've picked up an old Trixbox installation that was left half completed and I've got pretty much everything working properly. However, no matter what I do, I can't get extension 101 to behave. When ...
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266 views

Troubleshoot telnet connection from Windows 7 to UNIX

I am trying to connect to an Asterisk server in USA. I am using telnet < IP Address > 5038 from India to USA. The person in USA is able to telnet to the IP address and port from USA , but I am ...
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219 views

Is it possible to split Asterisk SIP signaling and voice traffic?

How do I configure Asterisk to send SIP signaling to one IP address and the RTSP stream, voice traffic, to another address?
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165 views

Connecting 2 asterisk channels together

Does anyone know if it is possible to do the following with asterisk... there are 2 users, John and Betty John calls -> asterisk accepts the call asterisk dials Betty and asks for DTMF ...