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 do you set up speed dial on Polycom IP6000 phones?

Basically I would like speed dial on this so I can press 1 and it will dial 101 AND I can press 2 and it will dial 555-555-5555. Is there a way to do this?
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3answers
1k views

VOIP over Comcast causing problems

We're running a VOIP system over Comcast. The problems are: Some calls don't complete (both incoming and outgoing) Some phone calls get terminated in the middle of the call. I want to run some ...
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4answers
475 views

VoIP on Netgear DGN 2000 and Lynksys PAP2

I have a VoIP service running perfectly for outgoing calls but I cannot receive incoming calls. I have a Netgear DGN 2000 modem router on 192.168.0.1 and a Lynksys Phone Adapter PAP2 on ...
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2answers
31 views

EC2 Server Seems to have port issues

I have installed an Elastix server on a m4.large instance,but I cannot get sound on any of my calls. The server is configured as follows: CentOS 5.9 Elastix 2.4 Using the following guide: ...
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2answers
93 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 ...
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3answers
2k 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 ...
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1answer
3k 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. ...
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3answers
280 views

Are Cisco Small Business 300 series switches compatible with pre-standard PoE detection of old Cisco IP-phones?

I have a humble Cisco SF302-08MP managed PoE switch that I'm trying to use for powering an old Cisco 7902G IP phone. This phone uses the proprietary Cisco pre-standard (i.e., non-802.3af) PoE ...
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3answers
1k views

Mitel 3300 - replacing a phone (one died, swapping for another)

I have to replace someones phone on a Mitel 3300 and we just took over this system so a bit new to us. do I just change the mac address in the users account info or is it more complicated then that? ...
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2answers
365 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
305 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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1answer
60 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 ...
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2answers
243 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 ...
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1answer
417 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. ...
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1answer
533 views

How to configure Transparent IP Address Sharing (TAS) on a Mediatrix 4102 with DGW 2.0 firmware?

I am making the switch to VoIP. I chose voip.ms as my service provider and Mediatrix 4102 as my ATA. One reason why I chose the Mediatrix over other popular consumer ATAs is that it's supposed to be ...
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1answer
89 views

Voice over ip setup for small business [closed]

I have no experience with Voip but if I wanted to use VIOP in a business with around 50 viop phones spread over 3 offices would I need to use some sort of voip Server or do most routers these days ...
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1answer
962 views

How can I call my voicemail in Bluebox Freeswitch?

I'm using Bluebox to manage Freeswitch. How can I call my voicemail? If I dial my extension it takes my to the part where I need to record a voicemail. If I dial 4000, *89 or 879 its just doesn't ...
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1answer
481 views

Click to call with a soft phone

I'm trying to set up click to call with a soft phone. I'm using QuteCom for the soft phone, vTiger for the CRM and Asterisk, through Elastix for the PBX. I was wondering if someone could point me in ...
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2answers
2k views

Cisco Phone (7961) won't find voice vlan

We have a cisco phone system setup successfully. I am setting up a bunch of phones, of which, one keeps getting an data vlan ip address from the DHCP server. Any suggestions? I have factory reset ...
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2answers
397 views

Asterisk Sip trunk though a PIX 501 firewall

I'm using an asterisk VOIP server, and I'm able to make called though with my SIP client. The problem I get is one-way voice. I can hear the incoming voice but the caller can not hear me. I have no ...
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2answers
1k views

Setup Cisco RVS4000 QOS for VOIP

Just switched from dedicated T1's with analog phone lines to cable modem with 10/2 uplink. We're having some VOIP call quality issues on the outgoing side when bandwidth is stressed and I need to ...
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1answer
172 views

VoIP PBX dilemma

Our phones are currently provided through a local hosted VoIP company and it isn't working all that great. As a result, we have decided to move to a local PBX server and bring a few PSTN lines to our ...
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2answers
8k views

How to remove static IP from Mitel 5312 and enable DHCP

I'm not sure this is the right forum for this question -- although I'm confident I'll be told if not! -- but I've read the fine manual (at least, such a manual as I have), I've googled and I cannot ...
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2answers
148 views

Verifying hosted VOIP PBX uptime

For the second time, I'm outsourcing my VOIP hosting to a provider, whom I have to trust their sales pitch of 45 minutes down per 365 days. Is there an automated way I can test their server(s) ...
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2answers
928 views

Embedded VoIP (Asterisk) install in home router?

Is it possible to flash an average home wireless router with a new OS that includes an embedded PBX server to route VoIP calls (such as Asterisk)? Does the average $50 router in the living room have ...
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1answer
44 views

Recommendation needed: IVRS + VoIP solution in Linux

I know there are some solutions on IVRS and VoIP in some Linux distributions, but I am not sure which one is reliable and stable. Or should I custom build a Linux dedicated for these services? Please ...
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2answers
606 views

Setup wiped Polycom phone without SIP server

I'm troubleshooting a Polycom SoundPoint IP 550. I have wiped the hard disk of the phone (via a menu option) and now it's stuck in a reboot cycle. Apparently the only way to setup the firmware of the ...
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2answers
3k views

VoIP phones getting back ip address: problem with dhcp zone or vlan

I have some Mitel IP phones that are giving me trouble on a network I recently inherited. I literally came in on for my first day supporting the system on Monday 12/28 to find the fiber connection to ...
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2answers
827 views

Asterisk & IVR & provider

I'm trying to setup an IVR or to be more specific an automated attendant using Asterisk. I don't want anything fancy besides the simple automated menu system, and instead of dialing an extension (for ...
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1answer
100 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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1answer
1k 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
41 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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2answers
190 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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1answer
872 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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1answer
326 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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2answers
843 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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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 ...
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1answer
10k 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 ...
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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 ...
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1answer
287 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 ...
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1answer
218 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. ...
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1answer
114 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 ...
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1answer
825 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 ...
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1answer
2k 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 ...
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2answers
148 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. ...
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1answer
369 views

Get Asterisk/Freepbx to ignore digits that were unnecessarily dialed (area code for local calls)

This is the second part of this question, worded better. Remove local area code on outbound calls, and remove "dial 1" for long distance in Trixbox/Asterisk I have removed the NEED to ...
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1answer
557 views

Voip, connect to asterisk servers over WAN with A580 base station

I have LAN infrastructure with asterisk and SIP phones(Gigaset A580) in it, I have done port forwarding on my public static ip address to my asterisk server in my LAN network, I also have trunk which ...
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1answer
1k views

How to pass/process extension numbers while calling Freeswitch from outside SIP gateway?

Suppose I have Freeswitch, which has SIP gateway configured. So, the local users of this Freeswitch can call outside via <extension name="russia"> <condition field="destination_number" ...
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1answer
2k views

How to associate a softphone with a phone url

I'm trying to get click to call to work using Qutecom for a softphone and a Greesemonkey script to turn phone numbers into clickable urls. Whenever I click on the phone number link I get an error ...
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1answer
264 views

Make VOIP-phone Cisco SPA504G to receive only 1 call

I don't want the phone to receive more that 1 call. So if an operator have one ongoing call, a new incoming call will received busy signal. As far as I understand, there is no option to set this ...