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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Shoretel Reporting Question

This might be a bit off topic, but I'll ask anyways. With Shoretel reports, the report that generates how long someone is on the phone(actively talking on the phone), if they are listed as "off hook",...
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41 views

add presence for a remote user to a legacy telephone system?

we have a small call center that uses an old nortel phone system with analog lines. one of our sales people works from home so her calls do not go through the phone system. this creates a problem at ...
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110 views

SIP server (“gateway”) for joining accounts

I have a phone supporting 2 accounts, but I need 4 accounts. Is it possible to install some sort of SIP server/gateway/proxy (on a linux server), that would register those 4 accounts and I would be ...
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892 views

Calls cut off in Asterisk 1.6.2

We are offering hosted VoIP. The Asterisk servers are running in our data center and customers connect their Aastra phones remotely to it. However, I have one customers connected to his own PBX and he ...
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1answer
127 views

Looking for a DID (or VoIP-in) number in the Beijing area

I am looking for an option to have a Direct-Inwards-Dialing option as provided by Skype or Gizmo5, but in the Beijing area (China). Both these providers do not have such an option. I do not search for ...
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2answers
282 views

Validity of my basic Performance Testing Results on a LAN

Networking is not my main job and i am hoping you guys can comment on my test procedure so far to identify PCs best to host a few voip softphones. Our 100Mb LAN comprises of 40 Win XP/2000 PCs and a ...
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4answers
690 views

Simple bandwidth monitor

I have a CentOS machine that is being used as a PBX. One interface is going out to the local network and the other interface is currently (pending equipment arrival) plugged directly into a Metro ...
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2answers
157 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. ...
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6answers
422 views

Does anyone know of a inexpensive NAT router that has the ability to limit access to the Internet to a specific MAC address? [closed]

Does anyone know of a inexpensive NAT router that has the ability to limit access to the Internet to a specific MAC address? I know the Linksys routers have a MAC filtering feature, but it is the ...
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2answers
692 views

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
483 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 192.168.0.2....
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32 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: https://...
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2answers
201 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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2answers
3k 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
285 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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2answers
9k 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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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
26 views

ISDN BRI vs SIP Trunks over EDIA

I am installing Asterisk in a small office with just 4 lines and 8 Extensions. I have two choices from my local telco (Fairpoint): 1) Old School ISDN BRI lines which I would connect to Asterisk ...
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2answers
571 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
504 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 http:///cgi-bin/ConfigManApp....
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1answer
62 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
275 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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453 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
568 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
90 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
1k 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
508 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
407 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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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
175 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
155 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
972 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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2answers
51 views

Recommendation needed: IVRS + VoIP solution in Linux [closed]

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
626 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
836 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
14 views

Logs concerning failed IAX2 registration attempt missing on Asterisk server

I am trying to register a VoIP softphone (Zoiper app for iOS) with an Asterisk installation via IAX2. At present, this results in an error message "Registration Refused (29)" on the side of the ...
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1answer
35 views

freeswitch received audio delayed for 20-30 seconds

I manage to make a very simple configuration for basic phone to phone calls using FreeSwitch, When i Calls A -> B (and B answered), A can hear B instantly, but B cannot hear A, after waiting for 20-30 ...
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2answers
772 views

SIP: No matching Codecs / 480 Temporarily not available

I'm installing a SIP Phone in a VoIP environment. There are 2 System Phones which work flawlessly (same manufacturer as PBX), and the third phone can be called, but it can't call the other 2 phones. ...
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1answer
223 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
2k 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 username=595XXYYZZZZZZ@prepago....
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1answer
51 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
251 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
1k 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
398 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 10.3.0....
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2answers
996 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 ...