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

-5
votes
2answers
266 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
852 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
125 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
4k 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
70 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 ...
1
vote
1answer
267 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
79 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
45 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
222 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
214 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
122 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 ...
5
votes
2answers
96 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
309 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
147 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 ...
1
vote
1answer
662 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
452 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
193 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
113 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
314 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
332 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
394 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
541 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
2k 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
713 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. ...
2
votes
0answers
2k views

Setting Shoretel phone time correctly with dhcpd

Our Shoretel 230 phones are not picking up time service correctly and thus, the time on the display is wrong. When I go to the diagnostics of the phone, they are using a wrong time server (old). Here ...
0
votes
1answer
84 views

Which handsets work with IP DECT base?

My workplace uses an Ascom IP DECT base station (IPBS1), and Ascom handsets as well. We're looking to replace some broken handsets; the ones by Ascom, however, are rather expensive, and we're not ...
1
vote
1answer
401 views

Asterisk Connection not working

I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something ...
0
votes
1answer
678 views

asterisk peer with SIP provider through proxy

I'm trying to make a asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but it doesn't seem to cover the case ...
1
vote
1answer
805 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 ...
1
vote
1answer
344 views

How to establish SIP connection, when SIP-proxy is required?

I have Asterisk/1.8.13.1 Asterisk GUI-version : SVN--r Yes, quite old one, but I can not update it since this is installed on my Synology NAS. NAS is connected to internet thru router Asus RT-N16. ...
0
votes
0answers
158 views

Communicate from PC in internet to remote GSM device

I have several remote devices that use GSM modem and gather data. I need to communicate with those devices from a PC station located in a office using the dial up connection. Is there a way to do this ...
0
votes
1answer
334 views

Asterisk dialplan context and label clarifications

I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the ...
1
vote
1answer
443 views

Configuration of a Cisco IP Phone 7916 and 7975g

Hi everyone, I would appreciate some help configuring my Cisco IP Phone 7975. In fact the phone works well with the SIP firmware (version SIP75.8-4-1ES2) and an Asterisk Server. Now I would like to ...
1
vote
1answer
110 views

SIP me@domain.com vs (123) 456-7890

I'm trying to understand SIP, and one thing that keeps confusing me is phone numbers and dialing plans vs SIP URI's like me@domain.com. What is the difference? How do they coexist/work together? For ...
0
votes
1answer
290 views

Cisco CME to CME - phone rings but silence on pickup

I have two Cisco 2901 routers with CME, DSP module and built-in poe switches. They are both linked together via Ethernet. I have 2 phones connected to each router. Making calls between 2 phones on ...
14
votes
2answers
985 views

Can anybody explain why to use Asterisk PBX rather than standard PBX from Matrix or Siemens [closed]

Can anybody explain why open-source Asterisk is favourable in comparison with a traditional PBX system from a vendor such as NEC/Syntel/Matrix or Siemens.
2
votes
1answer
474 views

Elastix and two ISDN lines

Since I am not an expert on communications, I will describe everything from the user perspective. We had a PBX with elastix embedded and two ISDN cards in it (we have two isdn lines). The mobo of the ...
1
vote
0answers
346 views

Cisco VoIP stuck as Unregistered?

Question: Why is one VoIP stuck as Unregistered? Background: We have a Cisco UC540 Small Business switch/router/voip combo. This phone was working until I powered everything down to install a ...
0
votes
1answer
273 views

ShoreTel P2V Audio Issue

I recently P2V'd my ShoreTel server. After doing so everything seems fine. No errors anywhere, and everything looks good in Director except for a licensing issue since the mac address changed. ...
1
vote
1answer
112 views

Turn old rack server into a telephone server

I have an old server lying around and I am thinking of using it as an internal telephone server. Its main use would be to set up a 1 to 1 telephone system that could be used internally to connect to ...
0
votes
0answers
483 views

VPN using pppd on linux keeps diconnecting after a couple of minutes

We have a VoIP box that we purchased from Allworx that I need to connect to using a VPN. I've configured a VPN connection using NetworkManager on my Fedora 14 Laptop and am able to connect, however ...
1
vote
3answers
785 views

Elastix and extensions.conf

My VOIP provider required me to add this into /etc/asterisk/extensions.conf: exten => h,1,Hangup exten => _00X.,1, Answer exten => _00X.,n, Set(CALLERID(num)=96354205144239) ; exten => ...
0
votes
2answers
793 views

Skype Connect and Elastix for incoming and outgoing calls

I ordered Skype Connect, And i want to integrate skype connect with my Elastix server to handle incoming and outgoing calls. I created new SIP Trunk through GUI with the following info : ...
0
votes
1answer
759 views

Elastix call-center with computer telephony crm integration

The title isn't explanatory enough but I can't explain it better. I want to setup a call-center using elastix which would have the following feature: Suppose bob(client) calls the center and ...
0
votes
1answer
130 views

Call history in Cisco Unity

I am looking for call records of a specific user (seeing when they called a specific number and duration) using the Cisco VoIP service. I've tried looking at the CDR Analysis (similar post How to ...
0
votes
2answers
492 views

How do I configure additional phone lines asterisk/trixbox?

I have a 4 port Digium card in there, and have 4 lines running smoothly. Now, we added ANOTHER 4 port card 1AEX410PELF and have 4 more analog lines coming into the Trixbox server. It still runs the 4 ...
2
votes
2answers
91 views

A tricky converged VoIP design

A client (roughly 30 people) wants to replace their existing analog phone system with VoIP. However: they have dual Ethernet drops to some desks, but not all their (recently purchased) router does ...
4
votes
2answers
95 views

Using an extension to block a caller

I have a couple of SIP phones and use callcentric. I get a lot of junk calls. I'd like to implement the following feature and would like some suggestions on how to do this: Once I get a junk call, I ...
0
votes
2answers
120 views

Can a “4 line” Grandstream GXP 2000 handle 8 lines?

We have a whole bunch of GXP 2000 phones here but now we're growing too big for just 4 incoming lines. We use Trixbox (asterisk) for the phone system with 4 analog lines coming into a Dell server. ...
1
vote
0answers
447 views

Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic ...