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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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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8 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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16 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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18 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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1answer
17 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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15 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
21 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
53 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
125 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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1answer
51 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] ...
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92 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 ...
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2answers
52 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 ...
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241 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 ...
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1answer
20 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. ...
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1answer
99 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 ...
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48 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 ...
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20 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 ...
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1answer
154 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. ...
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1answer
76 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 ...
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15 views

SIP UAC crash during a call

If a SIP UAC crashes in the middle of a SIP call, what is the expected behavior of the UAS and the remote UAC? Assume that a single UAS connects both UACs in the call. Is this scenario discussed in a ...
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2answers
69 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 ...
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2answers
128 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 ...
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349 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 ...
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80 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 ...
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273 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. ...
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33 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 ...
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105 views

Netgear N300 and VOIP phone

I work with a not-for-profit organisation who have recently purchased an internet connection plan that comes with a Netgear N300 wireless router. Unfortunately, there is not an option at this time to ...
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60 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 ...
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1answer
54 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 ...
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58 views

Freeswitch Database Hangup Causes

I am looking at our Freeswitch database for the VoIP service my company provides and trying to investigate an issue with a particular router. I see from a CDR (Call Detail Record) in the Freeswitch ...
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1answer
1k 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 ...
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1answer
84 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 ...
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169 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 ...
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1answer
94 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 ...
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29 views

Report on how many received calls on Cisco unified CM

I've been struggling with this a couple of days now and I feel like I've seen every webpage about Cisco Unity CM but just can't figure it out... The request I got sounded fairly simple: We want to ...
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45 views

Firewall Ports for Toshiba CIX server

Other then SIP ports: 5060 and 5061 what other ports need to be opened to have a phone set up in a satellite office with via external IP address. Phone Server CIX100 Firewall ASA 5510
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2k 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 ...
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120 views

Routing all network traffic through a VPN for VOIP purposes

We have recently moved offices and as we are in a rural area there is no broadband, but there is a strong 4G signal. Problem I have is that our VOIP phones are not allowed to pass through the 4G ...
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142 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 ...
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100 views

Combining Elastix And Zultys PABX

have anyone integrated Zultys MX250 with Elastix? I got some problem with it. Before, my PABX is Elastix, now i want to upgrade using MX250 but keep Elastix as gateway for MX250. But i can't call out ...
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1answer
1k 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 ...
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61 views

can't connect E1 digium te420 to freeswitch on CentOS

i'm experiencing an unknown problem for me, while i'm trying to connect digium te420 to freeswitch running on CentOS, digium card gives an alarm light while i'm connecting it. any ideas, can IRQ ...
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1answer
275 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 ...
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2answers
451 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 ...
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87 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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322 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 ...
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431 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 - ...
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1answer
1k 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) ...
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106 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 ...
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1answer
411 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 ...