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
2answers
136 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 ...
0
votes
2answers
20 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 ...
0
votes
0answers
4 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 ...
0
votes
2answers
943 views

Lync Server 2010 with Hosted VoIP PBX

We just deployed Lync Server 2010 in our organization and it is working great so far. The next step for us is to enable enterprise voice so that we can replace our telephones with service that is ...
0
votes
0answers
6 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 ...
1
vote
2answers
1k views

Asterisk DID incoming rules =s

So I have a 10 way SIP trunk from a UK provider. Calls in and out work fine, but as I have multiple numbers for different purposes I set up the inbound rules for each number but they never work, only ...
0
votes
0answers
23 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 ...
1
vote
1answer
1k 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 ...
0
votes
2answers
2k views

Limiting Bandwidth per port Cisco router

Let's say a businessses purchases a sufficently fast internet connection (such as 60Mbps cable connection) and VOIP quality is pretty good for the most part. The only thing I would be worried about ...
0
votes
2answers
147 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. ...
1
vote
2answers
493 views

Skype performance in IPSEC VPN

I've been challenged to "improve Skype performance" for calls within my organisation. Having read the Skype IT Administrators Guide I am wondering whether we might have a performance issue where the ...
0
votes
2answers
992 views

Connecting VoIP Phones to a Laptop

I'm trying to connect two VoIP phones to a laptop using the internal ethernet adapter and a usb-to-ethernet adapter. My configurations are as follows: Laptop eth0 - 121.1.2.1/24 Gateway - 121.1.2.4 ...
2
votes
1answer
1k views

asterisk/freeswitch in nat/no-nat setup

my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party ...
0
votes
0answers
18 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 ...
2
votes
1answer
2k 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 ...
0
votes
0answers
10 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 ...
0
votes
0answers
53 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 ...
0
votes
0answers
31 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 ...
0
votes
1answer
25 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 ...
0
votes
1answer
17 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 ...
-4
votes
1answer
56 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?
0
votes
1answer
49 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 ...
-1
votes
1answer
79 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 ...
1
vote
2answers
1k 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 ...
0
votes
0answers
87 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 ...
1
vote
0answers
197 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
1answer
91 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 ...
0
votes
0answers
37 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 ...
1
vote
1answer
2k views

How can I limit Asterisk's Chanspy feature to certain channels?

I'm using a standard config of Asterisk 1.4 with FreePBX using the chanspy feature code. I want to restrict what channels it is possible to spy on, ideally doing this in a .conf file. How would I do ...
16
votes
5answers
370 views

What alternatives do I have for telephony if my LEC says the Central Office has an issue?

In our New York office we've been waiting for almost a month for Verizon Business to repair a problem at our local CO. They keep saying that the damage from Hurricane Sandy was so extensive that they ...
1
vote
1answer
61 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
66 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 ...
2
votes
2answers
11k views

Asterisk SIP digest authentication username mismatch

I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue ...
0
votes
1answer
41 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 ...
2
votes
5answers
365 views

Intercom over IP?

This is an odd request, but I'm hoping some of you have had to field it before: We have an office in New York and remote employees in Oregon and Florida. Communication is extremely important to us, ...
0
votes
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) ...
0
votes
2answers
197 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 ...
0
votes
1answer
47 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
1answer
26 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 ...
2
votes
1answer
583 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 -> ...
1
vote
1answer
121 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
234 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 ...
0
votes
2answers
192 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
633 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 ...
0
votes
3answers
186 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
30 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. ...
1
vote
1answer
350 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. ...
0
votes
1answer
213 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 ...
2
votes
2answers
378 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 ...
5
votes
2answers
1k views

How should I structure a VoIP phone system?

I'm going to be setting up VoIP for a small business that is expected to grow rapidly. I don't really understand much about VoIP other than it's a telephone system that works over a regular network ...