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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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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18 views

How can I add user to Gnu Gatekeeper?

I've installed Gnu Gatekeeper from Debian wheezy repository. And I can't figure out how I can add user account to Gnu Gatekeeper server to use it in Ekiga or NetMeeting? What I should write in ...
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11 views

Report on how many received calls on Cisco unified CM

I've been struggling with this a couple of days now and I feel like I've seen every webpage about Cisco Unity CM but just can't figure it out... The request I got sounded fairly simple: We want to ...
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3 views

Firewall Ports for Toshiba CIX server

Other then SIP ports: 5060 and 5061 what other ports need to be opened to have a phone set up in a satellite office with via external IP address. Phone Server CIX100 Firewall ASA 5510
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1answer
19 views

How configure 2 Lan cards in Windows 7/8 pc one to connect to Internet and other to Local Network [on hold]

    I am about to install a dedicated VOIP server in our office. It is a 3CX pbx system on Windows 7/8 machine. The environment currently is a Windows SBS 2011 with 8 client machines. I want ...
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18 views

Routing all network traffic through a VPN for VOIP purposes

We have recently moved offices and as we are in a rural area there is no broadband, but there is a strong 4G signal. Problem I have is that our VOIP phones are not allowed to pass through the 4G ...
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28 views

RTP analysis - Discerning ptime (packetization time) for a given VoIP packet capture

I would like some help on the subject of an automated way of discerning the average packetization time (ptime) of a VoIP call's packet capture. The reason I am not depending on the value in the SDP ...
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15 views

Combining Elastix And Zultys PABX

have anyone integrated Zultys MX250 with Elastix? I got some problem with it. Before, my PABX is Elastix, now i want to upgrade using MX250 but keep Elastix as gateway for MX250. But i can't call out ...
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1answer
57 views

route incoming asterisk sip calls - fake auth rejected

we are trying to handle incoming sip_calls on our asterisk server, but somehow we always end up getting either 403 or 603 (which should be the same?) what should happen if it works: user calls ...
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26 views

can't connect E1 digium te420 to freeswitch on CentOS

i'm experiencing an unknown problem for me, while i'm trying to connect digium te420 to freeswitch running on CentOS, digium card gives an alarm light while i'm connecting it. any ideas, can IRQ ...
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1answer
47 views

qos with tc - high latency voip

I'm having following setup: local pbx doing calls for remote vpn clients. And have trouble with high latency a and quality of voip calls in remote office. Remote office is 10/10mbps link while local ...
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2answers
48 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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49 views

Switch from E1 to Analog PSTN line on IP-PBX failure

I have two Asterisk servers, E1(PRI) Gateway Asterisk 1.4 with E1(PRI) card IP-PBX Asterisk 1.8 on VM Server When my e1 gateway is failed (ex. shutdown, no power...), PSTN automatically switch to ...
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1answer
119 views

SIP INVITE packet has WAN address rather than call manager LAN IP

I am using SIP between two subnets (192.168.3.0/24 and 192.168.30.0/24) each connected via VPN. I have a call server on 192.168.3.100, and two phones 192.168.30.118 (Ext. 3128) and 192.168.30.119 ...
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1answer
137 views

Freeswitch : Audio handshake failure 1 error while making call using Sipml5

I just installed FreeSwitch and successfully connected to server with user 1001. Details -> OS - Ubuntu 12.04 LTS 64 bits FS - 1.5.13b+git~20140614T114905Z~fc7a74905b~64bit OpenSSL - 1.0.1 chrome - ...
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179 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) ...
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35 views

Cisco VoIP telephony - Can't save Night Service Bell Configuration

When trying to save "Night Service Bell Configuration" settings under Cisco Unified Communications Manager Express the following error message pops up. Configuration change failed. Please report the ...
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1answer
86 views

QoS Degrading VOIP

I have the privilege of trying to optimize our network at work to help improve the quality of the voip system we have. The general setup is: Bandwidth is 18 Mbps down, 3 Mbps up. ~10 people in ...
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36 views

How do I make Cisco phones use <user>@<domain> instead of <user>@<ip> in SIP messages

I am using Cisco 8000 and 9000 series IP phones and registering them to Opensips. The problem is that I want to communicate with the proxy using @ instead of @ so I can have multiple domains on one ...
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1answer
75 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 ...
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83 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 ...
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1answer
71 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 ...
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1answer
395 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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80 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 ...
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93 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 ...
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239 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 ...
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178 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 - ...
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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 ...
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73 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 ...
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1answer
107 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 ...
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2k 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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3answers
250 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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3answers
107 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 ...
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39 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 ...
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43 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 ...
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19 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.
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48 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?
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178 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 ...
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188 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 ...
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1answer
64 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 ...
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1answer
148 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 ...
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2answers
240 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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3answers
330 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 ...
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2answers
226 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 ...
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1answer
673 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 ...
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113 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 ...
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2answers
3k 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 ...
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1answer
63 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 ...
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1answer
235 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 ...
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1answer
78 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 ...