0
votes
2answers
25 views

Asterisk inbound trunk unauthorised without allowguest=yes

I am trying to configure an Asterisk (Elastix) box to receive SIP calls from a provider without requiring allowguest=yes to be enabled in sip.conf. Basically the SIP trunk provider uses multiple IPs ...
0
votes
0answers
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 ...
0
votes
1answer
82 views

SIP INVITE packet has WAN address rather than call manager LAN IP

I am using SIP between two subnets (192.168.3.0/24 and 192.168.30.0/24) each connected via VPN. I have a call server on 192.168.3.100, and two phones 192.168.30.118 (Ext. 3128) and 192.168.30.119 ...
1
vote
0answers
52 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
84 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 ...
0
votes
0answers
131 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 - ...
1
vote
1answer
81 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
3answers
100 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 ...
0
votes
1answer
38 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
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.
2
votes
0answers
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?
1
vote
1answer
138 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 ...
1
vote
3answers
301 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 ...
0
votes
1answer
195 views

Asterisk / Voip phone redirects to last called number

I experience a very strange problem with our PBX (Asterisk 1.8, FreePBX, Grandstream GXP1200 phones) If I call an internal number (1020 in this case) the asterisk server redirects me to the last ...
2
votes
5answers
192 views

VOIP Power outage strategy

In the past 2 years, we have had 4 instances where we have been without power for a period of 4-6 hours - due to construction crews cutting power lines, car accidents involving downed power lines, ...
5
votes
2answers
93 views

Trans-atlantic VOIP

We're a Canadian business on the east coast and will soon be opening up a new call centre in Australia. This new call centre will be handling our graveyard shift and will be overlapping with the ...
0
votes
1answer
93 views

Variable to store Asterisk extension dialed by user

I am using Asterisk for voice calling. I am using below context If user call back- [from-pstn] exten => _X.,1,Playback(demo-thanks) suppose my number is 74900 on which user can call back. ...
1
vote
0answers
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. ...
1
vote
1answer
346 views

Asterisk Connection not working

I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something ...
0
votes
1answer
545 views

asterisk peer with SIP provider through proxy

I'm trying to make a asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but it doesn't seem to cover the case ...
1
vote
1answer
655 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
321 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. ...
0
votes
1answer
248 views

Asterisk dialplan context and label clarifications

I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the ...
1
vote
1answer
375 views

Configuration of a Cisco IP Phone 7916 and 7975g

Hi everyone, I would appreciate some help configuring my Cisco IP Phone 7975. In fact the phone works well with the SIP firmware (version SIP75.8-4-1ES2) and an Asterisk Server. Now I would like to ...
14
votes
2answers
924 views

Can anybody explain why to use Asterisk PBX rather than standard PBX from Matrix or Siemens [closed]

Can anybody explain why open-source Asterisk is favourable in comparison with a traditional PBX system from a vendor such as NEC/Syntel/Matrix or Siemens.
2
votes
1answer
423 views

Elastix and two ISDN lines

Since I am not an expert on communications, I will describe everything from the user perspective. We had a PBX with elastix embedded and two ISDN cards in it (we have two isdn lines). The mobo of the ...
1
vote
3answers
676 views

Elastix and extensions.conf

My VOIP provider required me to add this into /etc/asterisk/extensions.conf: exten => h,1,Hangup exten => _00X.,1, Answer exten => _00X.,n, Set(CALLERID(num)=96354205144239) ; exten => ...
0
votes
2answers
688 views

Skype Connect and Elastix for incoming and outgoing calls

I ordered Skype Connect, And i want to integrate skype connect with my Elastix server to handle incoming and outgoing calls. I created new SIP Trunk through GUI with the following info : ...
0
votes
1answer
650 views

Elastix call-center with computer telephony crm integration

The title isn't explanatory enough but I can't explain it better. I want to setup a call-center using elastix which would have the following feature: Suppose bob(client) calls the center and ...
0
votes
2answers
446 views

How do I configure additional phone lines asterisk/trixbox?

I have a 4 port Digium card in there, and have 4 lines running smoothly. Now, we added ANOTHER 4 port card 1AEX410PELF and have 4 more analog lines coming into the Trixbox server. It still runs the 4 ...
4
votes
2answers
95 views

Using an extension to block a caller

I have a couple of SIP phones and use callcentric. I get a lot of junk calls. I'd like to implement the following feature and would like some suggestions on how to do this: Once I get a junk call, I ...
0
votes
2answers
115 views

Can a “4 line” Grandstream GXP 2000 handle 8 lines?

We have a whole bunch of GXP 2000 phones here but now we're growing too big for just 4 incoming lines. We use Trixbox (asterisk) for the phone system with 4 analog lines coming into a Dell server. ...
1
vote
0answers
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 ...
0
votes
1answer
555 views

Asterisk / Elastix Address Book and Call Recording

I am trying to build a small asterisk based PBX using Elastix. It'll be having 4 FXO (2 nos.no ISDN, normal analog POTS, 2 nos. GSM connections using a GSM Terminal) and 4 FXS (2 IP Phones and 2 ...
3
votes
4answers
8k views

How to stop registration attempts on Asterisk

The main question: My Asterisk logs are littered with messages like these: [2012-05-29 15:53:49] NOTICE[5578] chan_sip.c: Registration from '<sip:912@xx.xx.xx.xx>' failed for '37.75.210.177' - ...
2
votes
3answers
391 views

How to add more incoming/outgoing lines capability to Asterisk?

We have 4 people call us from around the country at once and it causes our phones to be busy. We want to be able to have at least 8 people call us at once without anyone getting the busy signal. ...
0
votes
1answer
202 views

Get Asterisk/Freepbx to ignore digits that were unnecessarily dialed (area code for local calls)

This is the second part of this question, worded better. Remove local area code on outbound calls, and remove "dial 1" for long distance in Trixbox/Asterisk I have removed the NEED to ...
1
vote
1answer
155 views

How to add a user who is out of state as if they were just another extension AT our office? [closed]

We want to add a person who is in another state to our phone system as though they were simply in another cubicle here at the office. So, they can dial "302" and get the office assistant's extension ...
0
votes
1answer
403 views

Extension not found error on calling an extension

I have setup a SIP server and connected two polycom soundstation duo devices to it. They both register line with the SIP server fine but when I try to place a call from one device to the other, ...
0
votes
1answer
334 views

Voip, connect to asterisk servers over WAN with A580 base station

I have LAN infrastructure with asterisk and SIP phones(Gigaset A580) in it, I have done port forwarding on my public static ip address to my asterisk server in my LAN network, I also have trunk which ...
1
vote
1answer
366 views

IM and File Transfers Integrated With VoIP

I have an Asterisk box serving an office of people. I'd like to provide instant messaging and file sharing capabilities alongside the voice and video capabilities provided by Asterisk (ala Windows ...
0
votes
2answers
175 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?
1
vote
2answers
1k views

Remove local area code on outbound calls, and remove “dial 1” for long distance in Trixbox/Asterisk

We have Trixbox CE 2.8 and Asterisk 1.6. I want to remove the 933 area code so you can just dial 555-5555 instead of 933-555-5555 for local calls. I also want to remove the +1 for long distance so ...
1
vote
2answers
503 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
votes
1answer
86 views

Is a PBX responsible for routing calls, or is that handled by the provider?

I am setting up a PBX that external users will connect to and I was wondering how important PBX placement was. Are calls routed through the PBX itself, or do they go directly through the provider ...
0
votes
1answer
139 views

Automatically log into SIP phone when user logs into Windows

We run an asterisk-based PBX, with FreePBX front end, configured in 'device and user mode' to support hot desking users. That is, a user arrives and sits at any desk, keys *11 into the phone (Cisco ...
1
vote
1answer
862 views

GUI for Asterisk 1.8.8 on Debian 6

Can someone point me towards a tutorial for installing either FreePBX, or a Webmin GUI for Asterisk 1.8.8 on Debian 6? Asterisk Thirdlane Multi Tenant PBX Manager and Asterisk Thirdlane PBX Manager ...
0
votes
3answers
3k views

Asterisk + SIP 404 not found

I want to make a small Asterisk server in my house. I installed asterisk on my Ubuntu and I use 2 computers, in order to connect to one another. when I connect I can see in Wireshark that registrar ...
1
vote
0answers
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 ...
9
votes
2answers
2k views

Can you please explain the difference between PBX and soft switch in less technical terms?

Asterisk is an open source PBX and FreeSWITCH is an open source soft switch. Source: http://www.freeswitch.org/node/117 Can you please explain the difference between PBX and soft switch in less ...