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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11 views

Does exists a Sip server with different cost / profile based on number? [on hold]

Does exists a SIP server where it is possible to configure price per minute or second based on destination number ? For instance: 0041 12345677 -> 1 € per minute 0041 22222333 -> 2 € per minute ...
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13 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 ...
1
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1answer
29 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 ...
0
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2answers
30 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 ...
0
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0answers
34 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 ...
0
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1answer
94 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
87 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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1answer
98 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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0answers
24 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 ...
1
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1answer
69 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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30 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
71 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
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0answers
63 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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votes
1answer
70 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
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1answer
284 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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0answers
69 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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0answers
86 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
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2answers
167 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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0answers
146 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
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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 ...
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0answers
66 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
87 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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1answer
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
179 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
102 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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37 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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1answer
40 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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0answers
17 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
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0answers
45 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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2answers
156 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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0answers
173 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
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1answer
63 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
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1answer
140 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
194 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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0answers
90 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 ...
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3answers
308 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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votes
2answers
207 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
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1answer
571 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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2answers
102 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
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 ...
0
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1answer
59 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
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1answer
212 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
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1answer
75 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
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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 ...
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1answer
199 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
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5answers
192 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, ...
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1answer
109 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
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2answers
93 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 ...
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2answers
251 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
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2answers
133 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 ...