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.

learn more… | top users | synonyms

0
votes
1answer
272 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 ...
-4
votes
1answer
155 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?
-1
votes
1answer
430 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 ...
0
votes
1answer
197 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 ...
1
vote
1answer
104 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 ...
0
votes
1answer
196 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 ...
0
votes
1answer
194 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 ...
0
votes
1answer
30 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 ...
0
votes
1answer
77 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 ...
0
votes
2answers
531 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 ...
2
votes
1answer
2k 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 -> PC/...
1
vote
1answer
307 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] ...
0
votes
1answer
695 views

FreePBX UK Dial Patterns Route Through Specific Trunk

I've read a ton of links and tutorials and for some reason cannot get my dial patterns to work correctly. I've got Freepbx installed and working properly - no issue there. I want to create a dial ...
1
vote
2answers
753 views

SIP Trunk for VOIP behind a SonicWALL

I appreciate this question is quite "out-there" but - has anyone had any success with VOIP over SIP behind a SonicWALL. We are having an issue with one-way audio inbound. Essentially, the person ...
0
votes
1answer
2k views

Can't dial through SIP trunk: FreePBX/Asterisk

I set up a SIP TRUNK in FreePBX/Asterisk that works perfectly for incoming calls. Here' s the relevant configuration: type=friend host=201.217.31.10 callerid=mynumber username=595XXYYZZZZZZ@prepago....
0
votes
1answer
48 views

Use VOIP through small switch

Probably a silly question but I have to ask. I don't know much about VOIP. We have a location with a large room now being used as an office for 2 people. There is one network cable ran to this office. ...
0
votes
1answer
491 views

Asterisk Sip Server and “Screen Sharing” function

I'am newbie in asterisk, i followed the tutorials on the asterisk wiki, and installed the latest version (13) of the asterisk server. I was able to setup the voip now i wonder, there is a function im ...
0
votes
0answers
143 views

Is it possible to virtualize a Genband/Broadsoft M6 voip telephony system?

I work with a company that runs a hosted VOIP platform originally known as the Genband M6 acquired by Broadsoft a few years back. This system is going end of life March 2016 and will no longer be ...
0
votes
0answers
35 views

How to automate updating of a setting on Cisco phones remotely?

Normally, to make a change one would get the IP address of a Cisco VOIP phone, access its settings via a web browser and make any changes to the settings they need to using the web interface. Is it ...
1
vote
1answer
975 views

Asterisk behind NAT sets wrong Contact Header

I'm using SIP with asterisk 13.1.0 behind a statically configured NAT. The servers private_ip differs from the public_ip, where I can reach it. I've already set these options in the sip.conf file. ...
4
votes
1answer
125 views

Ping time variation correlates with IP phones not working

We have an IP Phone system (most models are snom 320). Today someone's phone has decided to stop working. It has an IP address assigned to it by DHCP, and I can sometimes go to it's IP address to view ...
0
votes
2answers
239 views

routing for iax-protocol does not work on pfsense

I have problems with a pfsense-box routing iax-traffic (UDP port 4569). Our company has a central office and eight branch-offices. At the branch-offices we use pfsense as firewall and VPN endpoints (...
2
votes
2answers
1k views

Do I need VLAN for VOIP phone if my network switches supports IEEE P802.1p QoS?

I have 30 workstation and 30 IP Phones all under one single network 192.168.9.0. The network switches is unmanaged switch but supports IEEE 802.1p QoS (e.g.: DLink DGS 1008D and DGS 1016D). I ...
0
votes
1answer
1k views

Call duration limit & least cost routing

We have Elastix version 2.4 setup and we have some little questions which if you could please answer to solve via web based panel of Elastix, as agents are making calls via Elastix there are little ...
0
votes
1answer
457 views

Change vlan settings to Yealink phones at once

I have about 90 yealink ip phones. My target is to change vlan settings to all of them massively. For example, when it's going to reboot all of them, I run the following http:///cgi-bin/ConfigManApp....
0
votes
1answer
1k views

One-way audio problems with SIP over AT&T

I have a customer who has an VOIP PBX connected to a Level 3 fiber connection. He has offices all across the country using different ISPs. Two of those offices use AT&T, both in different states. ...
0
votes
0answers
74 views

Using a Barracuda F400 Firewall Router Without a Modem

I'm looking into a problem where VoIP phones don't ring for some of the incoming calls that I believe might be due to the network configuration. Do you need a modem if you are using a Barracuda F400 ...
0
votes
0answers
121 views

Cisco 7921G IP Phone not working

Got my hands on a second hand 7921G wireless IP phone. It should be configurable via USB; but need drivers to do that (http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7921g/5_0_1/english/...
1
vote
1answer
100 views

Configuring VOIP server without SIP

So I want to create a VOIP server with that I can connect with OpenVPN so I can call only a number of people that are in my network. I'll be using CSIPSIMPLE for that purpose. I'll be using only ...
3
votes
1answer
4k views

QoS settings for Wi-Fi calling on pfSense firewall/gateway?

The new version of iOS 8 as well as Android supports Wi-Fi Calling with multiple cellular providers including T-Mobile in the US. I am currently running pfSense as the primary Internet gateway for ...
2
votes
1answer
157 views

Preventing IP Phones from eating up DHCP addresses

We just installed a new IP phone system. The phones get their configuration via DHCP, which includes an option to set the phone to a different VLAN. The phone then requests an address via DHCP on that ...
0
votes
1answer
384 views

Asterisk relay between multiple subnets

I wonder what's the best way to go when you have phones on multiple networks which are not directly reachable. I have 3 networks 10.3.x.x 10.6.x.x 10.17.x.x My asterisk server resides on the 10.3.0....
0
votes
1answer
211 views

How can I add user to Gnu Gatekeeper?

I've installed Gnu Gatekeeper from Debian wheezy repository. And I can't figure out how I can add user account to Gnu Gatekeeper server to use it in Ekiga or NetMeeting? What I should write in /etc/...
-2
votes
1answer
3k views

How configure 2 Lan cards in Windows 7/8 pc one to connect to Internet and other to Local Network [closed]

    I am about to install a dedicated VOIP server in our office. It is a 3CX pbx system on Windows 7/8 machine. The environment currently is a Windows SBS 2011 with 8 client machines. I want ...
4
votes
0answers
258 views

RTP analysis - Discerning ptime (packetization time) for a given VoIP packet capture

I would like some help on the subject of an automated way of discerning the average packetization time (ptime) of a VoIP call's packet capture. The reason I am not depending on the value in the SDP ...
-1
votes
1answer
3k views

route incoming asterisk sip calls - fake auth rejected

we are trying to handle incoming sip_calls on our asterisk server, but somehow we always end up getting either 403 or 603 (which should be the same?) what should happen if it works: user calls ...
1
vote
1answer
472 views

qos with tc - high latency voip

I'm having following setup: local pbx doing calls for remote vpn clients. And have trouble with high latency a and quality of voip calls in remote office. Remote office is 10/10mbps link while local 6/...
0
votes
2answers
969 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
108 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
609 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
1answer
811 views

Freeswitch : Audio handshake failure 1 error while making call using Sipml5

I just installed FreeSwitch and successfully connected to server with user 1001. Details -> OS - Ubuntu 12.04 LTS 64 bits FS - 1.5.13b+git~20140614T114905Z~fc7a74905b~64bit OpenSSL - 1.0.1 chrome - ...
0
votes
1answer
2k views

Voice VLAN Configuration on Cisco SG300

We currently have a single Cisco SG300 switch on our data network. We are introducing a hosted VoIP system. As such we will be installing a second Cisco SG300 (to account for growth and new phones) ...
1
vote
0answers
192 views

Cisco VoIP telephony - Can't save Night Service Bell Configuration

When trying to save "Night Service Bell Configuration" settings under Cisco Unified Communications Manager Express the following error message pops up. Configuration change failed. Please report the ...
1
vote
1answer
907 views

QoS Degrading VOIP

I have the privilege of trying to optimize our network at work to help improve the quality of the voip system we have. The general setup is: Bandwidth is 18 Mbps down, 3 Mbps up. ~10 people in ...
-6
votes
1answer
131 views

Is running our workstation network through VoIP handset best practice? [closed]

We are purchasing a Cisco UC VoIP phone system and the handsets come in two flavors: 100mb and 1gb. Both allow network pass-through of another device such as a printer or workstation. Is this an ...
1
vote
0answers
217 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 ...
-5
votes
1answer
101 views

Normal phone number over ethernet, what is this service called? [closed]

Plug in phone to wall with ethernet cable Someone calls a toll free number Your phone rings What is the correct, vendor-neutral term for this service? Some terms out there are: Comcast calls the ...
0
votes
1answer
2k views

DHCP For Voice And Data VLANs With Router-on-a-Stick

Background: I have 2 Powerconnect 62xx switches stacked with a Watchguard XTM-26 between them and the outside world. Currently all devices connected to the switch are in the same vlan and the ...
1
vote
0answers
464 views

freeswitch forbidden gateway error

i've stuck into a strange problem, i have Ubuntu 12.04 with static IP, installed Freeswitch with FusionPBX with easy install script from here http://wiki.fusionpbx.com/index.php?title=Easy_FusionPBX ...
1
vote
0answers
260 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 ...