Voice over Internet Protocol (Voice over IP, VoIP) is one of a family of internet technologies, communication protocols, and transmission technologies for delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.

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ISDN BRI vs SIP Trunks over EDIA

I am installing Asterisk in a small office with just 4 lines and 8 Extensions. I have two choices from my local telco (Fairpoint): 1) Old School ISDN BRI lines which I would connect to Asterisk ...
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10 views

FAX on VoIP line does not work

My provider switched me to an IP-based line. Now my analog ISDN (G3) Fax does not work anymore. Is there any way I can use the conventional Fax with a SIP connection? My Router supports T.38 Fax ...
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17 views

Cisco ASA PBR and multiple ISPS for multiple SIP Providers

Have a question about policy based routing on my Cisco ASA 5515 running 9.5 and think despite it being available in this software version that I might be screwed. I have an asterisk PBX behind the ...
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24 views

Getting SIP, RTP and NAT right : who is doing what?

I'm a little confused about who (PBX vs. phones) is doing what in our IP telephone setup. We have our phone system hosted externally. This means that we've got physical IP telephones installed at ...
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10 views

SIP Extension Registration Port 5060 and Port 1024 1026 1028 etc

First time here. I have a problem registering remote extension on a Multi tenant FusionPBX/FreeSWITCH. I have signed up with a FusionPBX/FreeSWITCH EC2 in AWS for training purposes. I have setup ...
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26 views

Asterisk over OpenVPN [on hold]

i have big problem about rtp flow on my network, i have server asterisk, and this server i have installed openVPN access server too. i use directmedia on asterisk to analysis end-to-end client using ...
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1answer
18 views

Redirect Call Back Out To PSTN

I'm wondering if it's possible on a VoIP system (specifically Asterisk) over SIP trunks to redirect a call back out to the PSTN to no longer be routed through the PBX. What I'd like to do is see ...
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14 views

Linphone iOS behind CGNAT sets wrong Contact Header

We are using Linphone client on iOS over Verizon network. Our client sends wrong IP address on Contact header at 200 OK message as a response to INVITE message from Asterisk. SIP trace Our client's ...
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84 views

Hacker bypassing iptables

(moved from SO) I have iptables protecting a sip server. It blocks all IPs except ones I specifically opened, and it seems to work for almost everyone. I have tested from lots of ip addresses that ...
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26 views

Whitelist phones for DISA in asterisk

What would be the best practice to allow only a whitelisted list of numbers to access a DISA system in asterisk? I have a public number that lot of people use but if a certain number calls, it allow ...
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30 views

Phone won't register with Asterisk server

I'm trying to register a Grandstream phone on our asterisk server. We've done this tens of times, have documented how to do this, and recently it doesn't work anymore... I've tried with several phones ...
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49 views

It is possible to use not NAT able VoIP Softswitch behind pfsense/siproxd

We are using diffrent VoIP Softswitches (asterisk/squire) behind a pfsense Firewall and connect to diffrent SIP Endpoints. The local softswitches only have a local IP and the Endpoints are public IPs. ...
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2answers
28 views

Asterisk One Way Audio or No Inbound Calls

I have an Elastix server running in Amazon Web services, but I can't seem to resolve an issue that is causing either one way audio or no inbound calls to ring. If I set my SIP extensions to "Nat = ...
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1answer
23 views

How can I show the DID or Hunt Group on a Mitel 5320e phone?

We have a Mitel 5000 running a building full of 5320e desk phones with PRI from a T1. I recently started setting up custom DIDs for certain projects, but have not yet found a way to let the person ...
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16 views

Screen Share using Opensips and msrp relay

I am using opensips and msrp relay to implement screen sharing between blink clients but getting following errors at msrp relay: debug: session ...
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20 views

How to provide QOS for VOIP in Juniper?

I have 10 computers using browser based VOIP solution and they have problem with the internet connection. they are connected (wire) to Juniper Switches then to Fortigate. How can improve the quality ...
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56 views

DTMF Tone issue with Elastix

I am very new to Elastix and have just gone through the setup of a new system on Elastix 2.5.0-8, I have Elastix in the middle of 2 sip trunks, one is the service provider and the other is a 3rd Party ...
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27 views

Intercom for all phones on Cisco Unified Call Manager

In a Cisco based telephony (VoIP) system (CUCM 9.1.2.11900), I have successfully configured and used the intercom feature. It allows a user (typically a secretary) to push a button and automatically ...
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58 views

using panasonic ip phone kx-nt511abxb over vpn

we are using panasonic pbx system kx-ns500bx. now we started new remote site and planned to use two extensions there through vpn. already we have cisco asa 5505 ipsec vpn for using accounting ...
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45 views

IP Subnetting vs VLAN for VoIP Traffic

I have a fairly small network with scanners, printers, phones and workstations/servers all sharing the same subnet (10.0.0.0). I am having issues with our VoIP phones and sluggish response on the ...
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37 views

Kamailio 4.3 replecing opensips 1.11

We have a customized sip routing script in opensip 1.11 now for good performance and stability we are moving to kamailio 4.3. So here am facing some issues related to kamailio module and funciotns ...
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41 views

Route VoIP traffic over VPN between Routerboards?

Context (the Why): We have a couple of remote offices in shared/coworking spaces across the country. We've had significant issues getting our VoIP phones to work with those internet connections ...
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40 views

How to connect extensions of remote VOIP over remote STUN server?

Image of Architecture I am trying to setup I have configured the Asterisk server in a docker image and it's running, I have also configured the STUN/TURN server and it's also running on the same ...
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52 views

VoIP: No audio, local IP in SIP header

I'm trying to get SIP trunking to work on my Bintec Elmeg hybird 130j with sipgate.de as trunking Provider. So far, call signalisation works fine, so I can call outside phones and also get called ...
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61 views

VOIP T1 vs Fiber

Currently using a T1 (1.544Mbps) for what I think is pricey sum of 500 a month. Found I could get 30/30 fiber for about the same price. Would there be any downside to switching to fiber, I would think ...
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435 views

High availability Asterisk options?

We currently run FreePBX on a single node. I'd like to have a cluster of asterisk instances for sharing the load, but mainly for failover. I'm curious how other people have solved this problem. ...
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112 views

Asterisk with softphone and webphone

I have a problem with Asterisk. I wrote a WebPhone, so I should add to my peers some config details: encryption = yes avpf = yes icesupport = yes dtlsenable =yes dtlsverify = no dtlssetup = actpass ...
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1answer
22 views

Microsoft UC and Skype for Business

I new to the Hosted PBX and VoIP systems. Currently the company I work for has an old T1 phone system they are looking to move away from. We have Skype for Business and Office 365. Basically my ...
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2answers
411 views

SIP: No matching Codecs / 480 Temporarily not available

I'm installing a SIP Phone in a VoIP environment. There are 2 System Phones which work flawlessly (same manufacturer as PBX), and the third phone can be called, but it can't call the other 2 phones. ...
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33 views

VoIP phones not registering on second zone - SonicWALL TZ210

Environment: Hosted VoIP SonicWALL TZ210 running SonicOS Enhanced 5.9.1.1-39o (latest) ADSL WAN connection The SonicWALL has two zones (subnets), X0 and X2, LAN and Voice respectively. Both are ...
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35 views

VoIP Phones not working inside LAN

I have tried to read many documents on running a VoIP system behind a NAT'd router, going out to the Internet, then back through a dedicated connection for the VoIP server. I am not getting anywhere ...
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118 views

Can I set up a VoIP phone to connect to it's gateway over VPN with different subnets?

A little background first: Currently, we have a main office and a remote office. They are connected via site-to-site VPN. Main office uses a Cisco router with DHCP at 10.10.10.0/24 while the remote ...
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29 views

Asterisk Transfer call to next extension if previous INUSE

I am trying to transfer call to next extension if previous is using (INUSE) or call is in progress. I tried to use EXTENSION_STATE(extension[@context]) to find the status as follow: [sales] exten ...
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31 views

What do I need to setup a small scale videoconf system on linux?

I have the task of deploying a small scale (for small groups, up to 15 persons) videoconf system. In principle it shouldn't be that hard, I thought, but after a day of reading I think I miss the main ...
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32 views

EC2 Server Seems to have port issues

I have installed an Elastix server on a m4.large instance,but I cannot get sound on any of my calls. The server is configured as follows: CentOS 5.9 Elastix 2.4 Using the following guide: ...
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112 views

RTPProxy traversal does not work wit Opensips

I installed opensips but there is no sound on clients. This is what I did: apt-get install opensips rtpproxy osipsconfig I selectred Nat Tracversal and generated the .cfg file. The customized ...
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145 views

Does a SIP trunk relay the RTP stream or just set up the call?

Suppose I have a SIP PBX like Asterisk and a bunch of phones registered to it, and outgoing/incoming calls are handled through a SIP trunk. Do the RTP streams go directly between the phones and the ...
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13 views

How to recognize conference calls from individual simultaneous calls When we have Asterisk user agents?

I want to know how I can distinguish between conference call from some independent simultaneous calls when we have Asterisk agents? RFC 4353 says that in conference calls we have a call member with a ...
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263 views

Asterisk SMS with GOIP

I have a GOIP 1 channel gateway and following this guys tuorial: https://github.com/tangaza/Tangaza/wiki/Connecting-a-GoIP-GSM-VoIP-gateway-to-an-Asterisk-AGI For me receiving SMS-es only works if ...
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157 views

Using the VLAN of a bridged switch

In our building, we have 2, 6509 core switches, one for our company, and one for another. We have a vlan configured with a /30 which bridges the 2 switches. Both of us are on separate /16 networks. ...
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21 views

ProCurve 2626 VLAN routing issue

I have a pair of Procurve E26262's that are our Distribution and POE switches. They are configured with a Default VLAN for data with an IP address of 192.168.1.218, and a VLAN 100 for VOIP with an IP ...
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22 views

Need Help Connecting Two VOIP Devices?

I have two Motorola XPR 8400 two way radio repeaters. These repeaters are capable of voice over ip. These two repeaters will be placed at two different sites with different ip addresses. I need to ...
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139 views

DNS Server not forwarding requests

I have been working on setting up some VOIP phones for our provider. The phones connect to their provisioning server in order to download the configuration. If I leave the phone as DHCP, it gets our ...
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224 views

Asterisk SIP Retransmission timeout after a month of server uptime

This Asterisk server runs fine behind NAT with a dozen of SIP registrations and the call volume of a few hundred calls per month. But then, after about a month of uptime, out of the blue the SIP ...
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1answer
74 views

FusionPBX Failed(405) on Cisco 303 IP phone

Since yesterday, my phone started throwing a Failed(405) error when trying to register to a FusionPBX server. I have tried resetting to factory settings and reinstalling the FusionPBX server on ...
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30 views

MCU packet capture

I am trying to do a packet capture of a video conference hosted with CISCO [Codian] MCU. The instructions for capturing SIP packets with Wireshark are available. I also configured MCU to accept SIP ...
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224 views

Dial 0 for operator not working externally

We have a Deployment with Lync and UM enabled with Exchange, both 2013. When we dial the number of a person in a response group and cannot reach him/her we get the Voicemail auto attendant, also we ...
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142 views

Can I use Kamailio Sip Server to handle 500 calling? [closed]

I need a Sip server to build a VIOP server to establish voice call between 500 Android an desktop sip clients. Can Kamailio handle this or I need an Asterisk server too?
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396 views

Auto DHCP IP Phones to a different subnet

I'm standing up a 3CX PBX system in my office and would like to set my DHCP to assign IP phones a 10.10.8.x address, while giving everything else a 10.10.0.x address (our current net). What would be ...
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143 views

Asterisk : SIP/2.0 401 Unauthorized : same phone , different results

First of all I have already taken a look at similar questions such as : http://serverfault.com/questions/392979/asterisk-sip-2-0-401-unauthorized or Asterisk 401 Unauthorized when trying to register ...