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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17 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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9 views

Home network - do I need QoS? [migrated]

I'm planning to install a home network with the following hardware: Netgear UTM50-100 ProSecure Netgear ProSafe JGS524PE (24 Port switch, 12 Port PoE) ESXi Server w/ 2 VM's (AWS Storage Gateway ...
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13 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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11 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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19 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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35 views

asterisk, “blonde” transfer and call back

I have Asterisk 11.x and a Yealink T28P IP-phone. One of my user is requesting "blonde" transfer (this is not my invention, this is the developers term, after all, I did my homework and read a lot) ...
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13 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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32 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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35 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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23 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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26 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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24 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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32 views

GOIP-1 does not forward SMS messages to Asterisk

I have a GOIP 1 channel gateway and followed this tutorial: https://github.com/tangaza/Tangaza/wiki/Connecting-a-GoIP-GSM-VoIP-gateway-to-an-Asterisk-AGI For me receiving SMS-es only works if the ...
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1answer
39 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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2answers
47 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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1answer
149 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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1answer
66 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
18 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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141 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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23 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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33 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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72 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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2answers
25 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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29 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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31 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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86 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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93 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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11 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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211 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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2answers
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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18 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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21 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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115 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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153 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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53 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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1answer
25 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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1answer
163 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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1answer
113 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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1answer
310 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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127 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 ...
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1answer
150 views

How to have dynamic extensions in Asterisk

I have recently swapped VoIP providers. So far so good, but I have ran into one little snag (and I'm honestly not really sure how to ask / Google this). With my old provider I could have inside my ...
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78 views

SIP calls not working in kazoo

I have installed Kazoo on my CentOS 6.6. Installation was ok and I didn't get any error. I have created two users in Hosted PBX section and added device to them as well. But when I am trying to ...
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1answer
84 views

VoIP calls dropping

Currently I have a proxy server running pfSense connected to a 40mb fibre lease line. This lease line has 60 external IP addresses (assigned to each room by a virtual IP in pfSense). On the LAN side ...
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1answer
133 views

How to move company voip phones to a different subnet by modifying DHCP of a Cisco Switch?

Our company has replaced its old voip phones with new ones which are the NEC phones. We are done with the configuration part, and the phones are working fine. But I want the phones to be in a ...
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100 views

Make an authenticated SIP call via URI

I'm trying to redirect a phone number to a SIP address that's only accessible when authenticated. Given the following, how would the SIP URI look like (if possible) ? SIP Server: sip.example.com Auth ...
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28 views

voice vs data device authentication

When a device is plugged into a switch (let's say, a Cisco Catalyst switch) how does a switch know whether the device is a VoIP one - and subject to being placed into the voice VLAN - or some other ...
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1answer
67 views

how do I route Asterisk dials from serverA to serverB?

I'm trying to make an Asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but am having a conceptual problem with ...
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2answers
363 views

Asterisk call should be with peer, but gets trapped and stays local

I'm working with Asterisk and Vicidial, trying to place outbound calls through a SIP "trunking" provider. The calls never connect, instead I hear either demo-instruct.gsm or invalid.gsm playing from ...
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1answer
1k views

Cisco Voice VLAN with 802.1X Authentication

I've got a Cisco Catalyst 2960 that I'm trying to configure for a remote office with 802.1X wired authentication. The setup I'm going for is Switch -> VoIP Phone via Internal Switch -> ...
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233 views

why do all softphones fail to authenticate after receiving 401 unauthorized from Asterisk?

When I attempt to connect to the Asterisk server from CSipSimple, over a Cisco router, on the same network, Asterisk logs show the attempted connection: <-------------> [Mar 23 20:01:34] ...