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

-6
votes
1answer
55 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
7 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 ...
-1
votes
0answers
21 views

How do I change the default extension that Cisco auto attendant dials when it can't find an extension?

How do I change the default extension that Cisco auto attendant dials when it can't find an extension or 'dial-by-name' name? Our former network engineer has auto-attendant answering calls to x100 ...
-5
votes
1answer
59 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 ...
0
votes
1answer
79 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
21 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
44 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
2answers
59 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 ...
0
votes
0answers
56 views

Soft phone not getting register to my asterisk server

I have config in my local lan asterisk for test. which is working fine. With the same OS version and packages on "AWS Cloud" I am setting as following. But it don't get register "error = sip 408 - ...
0
votes
0answers
18 views

Is it possible to have VOIP and data share a common fail-over connection?

Internally, VOIP and data services are wired separately. They operate on two mutually exclusive networks, with separate broadband connections. Currently, there is no fail-over backup for VOIP ...
0
votes
0answers
43 views

cisco uc560 newly created extensions can't dial out

Our ucs560 is working fine, today I needed to add a few new users, our internal user extensions use the range 820 thru 867 or so. I added the new extensions 880-884. Internal calls back and forth ...
1
vote
1answer
48 views

Security, hardening and NAT for RTP and SIP

I decided to create a VOIP server and it has gotten more complicated then I thought it would. So in order to make my server more secure, I keep it behind a pfSense appliance. I use IP filtering to ...
0
votes
1answer
789 views

Cisco ASA 5505 - NAT or Port Forward for SIP / VoIP ver 8.4

I just had an NEC PBX installed that lets me use SIP trunks for VoIP services, My gateway is a Cisco ASA 5505 running 8.4 and I only have one public/static IP Addresses. So far, my trunks are ...
0
votes
3answers
90 views

How do I calculate the necessary bandwidth for my office with PCs and VoIP phones?

In our office we have 30 PCs, 2 WiFi and 6 VoIP phones connected to the network that consumes bandwidth. Basically 30 PCs are simultaneously using the Internet (like Facebook, surfing the web and ...
0
votes
3answers
89 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
0answers
25 views

Cisco 7921 Authentication troubles

hello I'm working with Call Manager 8.5.1.10000-26 and several Cisco ip phones a 7921 and 7965 I'm having trouble with authorization on the 7921. I wish to invoke a CGI/Execute on the phone . The ...
0
votes
1answer
28 views

Asterisk, GSM-FXO gateway example of usage

Can anyone explain where to use GSM-FXO gateway. I understand where to use GSM-FXS - for example to connect analogue telephone to gsm gateway. But do we use GSM-FXO to connect to two telephone ...
0
votes
0answers
16 views

Alarm when a queue is unattended

Is there a way to make a trigger/alarm when a queue is left with no one to attend it? My setup is freepbx and CDR-stats, but it could be a plain Asterisk solution as well.
2
votes
0answers
40 views

How to set Mitel 5224 ringer volume in configuration?

We have several Mitel 5224 SIP phones, but can't seem to find out to make the ringer volume setting persist after it reboots. Can the ringer volume can be set somewhere in the XML configuration file?
1
vote
2answers
110 views

Setting up VOIP system on separate VLAN / Server on default VLAN

First off, I am not a network engineer so apologies if I get terminology messed up, i realize accuracy is important here. Currently, we have a small(ish) network with around 40 devices and 20 IP ...
1
vote
0answers
133 views

Cisco SPA504G phone CDP/LLDP issue

Good morning everyone, Just recently signed up for the site because I'm looking for some help with an issue that's been plaguing me for about a week now. I hope I'm posting in the right place. If ...
1
vote
1answer
57 views

Which one VoIP SIP audio codec should I choose to high quality calls?

The quality of VoIP calls depends on the codec used for the transmission and on the bandwidth of Internet connection. I want to use best optimized codec for my Internet connection. I have to select ...
1
vote
1answer
114 views

Asterisk ignoring sip_domain part of URI

I am using Asterisk version 11.5. My problem is the following: I literally cannot make any outbound call. That's because, whatever I dial in any client, the sip_domain part gets ignored, it is ...
0
votes
2answers
106 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
0answers
63 views

How do I configure my sonicwall VPN to allow remote Voip phones to communicate to other remote phones?

I have two remote locations and 1 central office that all use a VoiP system located at the central office. We have a TZ210 in the central office and TZ105 at the remote locations. The phone system ...
1
vote
3answers
247 views

Asterisk and SIP trunking, is this configuration possible?

I am new to asterisk and before I dive in, I just want to make sure that what I plan to do is possible/correct. My office will run an asterisk server and have both local and remote extensions. We ...
-4
votes
2answers
148 views

Make own dial-up ISP using broadband connection [closed]

So, I see no reason why this wouldn't be possible. I have a Linux server (a Raspberry Pi to be exact) connected via Ethernet to a broadband ISP. I want to be able to dial a number, have it go ...
1
vote
1answer
290 views

VoIP Network Components: Difference between Signaling Gateway, Media Gateway, Softswitch [closed]

In my degree course of Software Engineering we had a lecture in Computer Networking about VoIP and the Session Initiation Protocol (SIP). There were many terms brought up for some network components ...
1
vote
2answers
79 views

What determines the length of RTP packets?

I have established a SIP session between two clients. I observed the RTP trace between them on wireshark. The 'length' column has a value of 172 for the RTP packets that flow from Client 1 to Client 2 ...
2
votes
2answers
2k views

Factory Reset Cisco SPA504G without admin password

I've been trying to factory reset some SPA500 phones, specifically the 504G but the old provider has locked out everything. I need to reset about 50 phones for use with a new service. I've man in the ...
0
votes
1answer
51 views

Limit Cisco IP phone to only one subnet ranges DHCP

We run stretched layer 2 and we have the e911 service setup to correspond for each location. We run different sub-nets for each location as well. I'm wondering if there is a way to limit a users ...
0
votes
1answer
171 views

What is the best option for SIP service redundancy?

From your practical experience, what is the best option for SIP service redundancy? a) one SRV record pointing to a hostname which has two A records b) two SRV records pointing to two different ...
0
votes
1answer
70 views

Is there a Lync equivalent of Avaya's Telecommuter Mode?

In Avaya Telecommuter mode, the Avaya IP Office has an option to route the voice call to an Avaya hard/soft phone to a traditional phone in the PSTN. While the TCP traffic for call handling goes to ...
0
votes
1answer
40 views

How to use fax with my VoIP

Well the site seems to be Hide Park for networking experts. To be honest I am not any expert at all. I followed my friend have recently got a business plan (for my new business) from Axvoice because ...
0
votes
1answer
174 views

Asterisk / Voip phone redirects to last called number

I experience a very strange problem with our PBX (Asterisk 1.8, FreePBX, Grandstream GXP1200 phones) If I call an internal number (1020 in this case) the asterisk server redirects me to the last ...
2
votes
5answers
181 views

VOIP Power outage strategy

In the past 2 years, we have had 4 instances where we have been without power for a period of 4-6 hours - due to construction crews cutting power lines, car accidents involving downed power lines, ...
0
votes
1answer
97 views

3com VCX - how to get a dump of outgoing phone calls

I am looking (hoping) for a way to dump a list of all outgoing calls from the VCX for a time period? Possible via the web interface, or via the ssh interface. For example, we need the date (time ...
4
votes
1answer
76 views

Trans-atlantic VOIP

We're a Canadian business on the east coast and will soon be opening up a new call centre in Australia. This new call centre will be handling our graveyard shift and will be overlapping with the ...
1
vote
2answers
206 views

VoIP client permamently losing connection after ISP IP reset

I have a SIP client device connected to my ADSL router, so it is behind a symmetric NAT. My ISP resets the client's IP every 24 hours, it looks like this in the router log: Mar 26 12:36:56 daemon ...
0
votes
2answers
112 views

DHCP and ip assignment with VoiP

We just added a Cisco SPA122 ATA with Router to our network, it will be used for our paging system (an amplifier/speakers). We are having a hard time getting it work. Well until we found out the ip ...
0
votes
1answer
374 views

Poor audio quality during RDP with thin client from Windows Multipoint Server 2012

we have problems with audio quality from Windows Multipoint Server 2012 during RDP sessions, we are using Thin Clients for the Remote Desktop the display is good but the audio quality is bad, there ...
0
votes
1answer
286 views

Is there a traceroute command for VXWorks? Specifically a Shoretel Phone?

I can't seem to find if there is a traceroute command for VXworks online or specifically with a Shoretel phone. Once I telnet into the phone, I would like the ability to see whether packets from my ...
0
votes
1answer
135 views

Custom ENUM/NAPTR DNS Server

I need to configure custom DNS server to handle ENUM requests for VoIP calls I have custom built DNS server in Java that handles ENUM requests for my company's VoIP services. ENUM is used to ...
0
votes
1answer
89 views

Variable to store Asterisk extension dialed by user

I am using Asterisk for voice calling. I am using below context If user call back- [from-pstn] exten => _X.,1,Playback(demo-thanks) suppose my number is 74900 on which user can call back. ...
1
vote
0answers
238 views

UDP Range Nat Translation port forwarding RPT Traffic to inside host PBX Cisco ASA 5505 8.2a

I have an Asterisk based PBX that I'm providing VoIP services with, My gateway is a Cisco ASA 5505 running 8.2 and I only have one public IP Available along with other services I provide with that IP. ...
16
votes
4answers
325 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 ...
5
votes
2answers
346 views

Is it possible for Wireshark to drop packets purposely?

I would like to test something like VoIP. I would like to test with some "artificial packet loss". Is Wireshark able to do this? Or is there any good solutions?
1
vote
1answer
324 views

VOIP phones stop working when spanning-tree (STP) enabled on switch

I have a few different types of managed switches, including mostly 3com Super Stacks, Baselines (for example, one specifically is a 3com baseline pwr-plus 2928). Most of them are configured with two ...
0
votes
1answer
1k views

How to route/forward/connect incoming SIP call through other SIP provider to a phone number?

I have 2 VOIP SIP accounts. One is for a landline (incoming). The other should be purely outgoing and allows for calls to my mobile phone. Basically I want to connect the landline to my mobile phone. ...
4
votes
2answers
457 views

What would cause a PBX or SIP phone to not register on the network when using Netgear Switches?

We have 3 different types of switches in our network. All of our IP Phones are running on HP 2520-24 PoE switches. Our primary backbone is "supposed to be" running on Netgear GS748T switches. ...