Questions tagged [voip]

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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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 ...
iewebguy's user avatar
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2 votes
3 answers
800 views

FAX on VoIP line does not work

My provider switched me to an IP-based line. Now my analog ISDN (G3) Fax does not work anymore. Is there any way I can use the conventional Fax with a SIP connection? My Router supports T.38 Fax ...
NoMad's user avatar
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Getting SIP, RTP and NAT right : who is doing what?

I'm a little confused about who (PBX vs. phones) is doing what in our IP telephone setup. We have our phone system hosted externally. This means that we've got physical IP telephones installed at ...
sbrattla's user avatar
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4 votes
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Redirect Call Back Out To PSTN

I'm wondering if it's possible on a VoIP system (specifically Asterisk) over SIP trunks to redirect a call back out to the PSTN to no longer be routed through the PBX. What I'd like to do is see ...
StrangeWill's user avatar
2 votes
0 answers
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Linphone iOS behind CGNAT sets wrong Contact Header

We are using Linphone client on iOS over Verizon network. Our client sends wrong IP address on Contact header at 200 OK message as a response to INVITE message from Asterisk. SIP trace Our client's ...
Falieris Ilias's user avatar
9 votes
2 answers
9k views

Hacker bypassing iptables

(moved from SO) I have iptables protecting a sip server. It blocks all IPs except ones I specifically opened, and it seems to work for almost everyone. I have tested from lots of ip addresses that ...
David Wylie's user avatar
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Whitelist phones for DISA in asterisk

What would be the best practice to allow only a whitelisted list of numbers to access a DISA system in asterisk? I have a public number that lot of people use but if a certain number calls, it allow ...
user886869's user avatar
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Phone won't register with Asterisk server

I'm trying to register a Grandstream phone on our asterisk server. We've done this tens of times, have documented how to do this, and recently it doesn't work anymore... I've tried with several phones ...
SPRBRN's user avatar
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Asterisk One Way Audio or No Inbound Calls

I have an Elastix server running in Amazon Web services, but I can't seem to resolve an issue that is causing either one way audio or no inbound calls to ring. If I set my SIP extensions to "Nat = ...
June Lewis's user avatar
1 vote
1 answer
400 views

How can I show the DID or Hunt Group on a Mitel 5320e phone?

We have a Mitel 5000 running a building full of 5320e desk phones with PRI from a T1. I recently started setting up custom DIDs for certain projects, but have not yet found a way to let the person ...
Bryan's user avatar
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-2 votes
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using panasonic ip phone kx-nt511abxb over vpn

we are using panasonic pbx system kx-ns500bx. now we started new remote site and planned to use two extensions there through vpn. already we have cisco asa 5505 ipsec vpn for using accounting ...
Mr Y's user avatar
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IP Subnetting vs VLAN for VoIP Traffic

I have a fairly small network with scanners, printers, phones and workstations/servers all sharing the same subnet (10.0.0.0). I am having issues with our VoIP phones and sluggish response on the ...
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How to connect extensions of remote VOIP over remote STUN server?

Image of Architecture I am trying to setup I have configured the Asterisk server in a docker image and it's running, I have also configured the STUN/TURN server and it's also running on the same ...
DVG's user avatar
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VoIP: No audio, local IP in SIP header

I'm trying to get SIP trunking to work on my Bintec Elmeg hybird 130j with sipgate.de as trunking Provider. So far, call signalisation works fine, so I can call outside phones and also get called ...
NoMad's user avatar
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2 votes
2 answers
285 views

VOIP T1 vs Fiber

Currently using a T1 (1.544Mbps) for what I think is pricey sum of 500 a month. Found I could get 30/30 fiber for about the same price. Would there be any downside to switching to fiber, I would think ...
JustAGrump's user avatar
9 votes
1 answer
10k views

High availability Asterisk options?

We currently run FreePBX on a single node. I'd like to have a cluster of asterisk instances for sharing the load, but mainly for failover. I'm curious how other people have solved this problem. ...
Jonathan S. Fisher's user avatar
1 vote
1 answer
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Asterisk with softphone and webphone

I have a problem with Asterisk. I wrote a WebPhone, so I should add to my peers some config details: encryption = yes avpf = yes icesupport = yes dtlsenable =yes dtlsverify = no dtlssetup = actpass ...
Matt's user avatar
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Microsoft UC and Skype for Business

I new to the Hosted PBX and VoIP systems. Currently the company I work for has an old T1 phone system they are looking to move away from. We have Skype for Business and Office 365. Basically my ...
MainStain's user avatar
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2 answers
12k 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. ...
NoMad's user avatar
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1 vote
1 answer
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VoIP Phones not working inside LAN

I have tried to read many documents on running a VoIP system behind a NAT'd router, going out to the Internet, then back through a dedicated connection for the VoIP server. I am not getting anywhere ...
Canadian Luke's user avatar
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1 answer
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Can I set up a VoIP phone to connect to it's gateway over VPN with different subnets?

A little background first: Currently, we have a main office and a remote office. They are connected via site-to-site VPN. Main office uses a Cisco router with DHCP at 10.10.10.0/24 while the remote ...
Jonathan's user avatar
2 votes
2 answers
370 views

Asterisk Transfer call to next extension if previous INUSE

I am trying to transfer call to next extension if previous is using (INUSE) or call is in progress. I tried to use EXTENSION_STATE(extension[@context]) to find the status as follow: [sales] exten =&...
Jibon's user avatar
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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://...
June Lewis's user avatar
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2k 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 ...
mortabis's user avatar
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2 answers
203 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. ...
joerockt's user avatar
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37 views

Need Help Connecting Two VOIP Devices?

I have two Motorola XPR 8400 two way radio repeaters. These repeaters are capable of voice over ip. These two repeaters will be placed at two different sites with different ip addresses. I need to ...
madhatter1919's user avatar
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0 answers
594 views

DNS Server not forwarding requests

I have been working on setting up some VOIP phones for our provider. The phones connect to their provisioning server in order to download the configuration. If I leave the phone as DHCP, it gets our ...
Matt Fogleman's user avatar
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1 answer
276 views

FusionPBX Failed(405) on Cisco 303 IP phone

Since yesterday, my phone started throwing a Failed(405) error when trying to register to a FusionPBX server. I have tried resetting to factory settings and reinstalling the FusionPBX server on ...
ybybeyb's user avatar
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1 answer
109 views

MCU packet capture

I am trying to do a packet capture of a video conference hosted with CISCO [Codian] MCU. The instructions for capturing SIP packets with Wireshark are available. I also configured MCU to accept SIP ...
Moshe Shmukler's user avatar
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1 answer
863 views

Dial 0 for operator not working externally

We have a Deployment with Lync and UM enabled with Exchange, both 2013. When we dial the number of a person in a response group and cannot reach him/her we get the Voicemail auto attendant, also we ...
jmr's user avatar
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-3 votes
1 answer
565 views

Can I use Kamailio Sip Server to handle 500 calling? [closed]

I need a Sip server to build a VIOP server to establish voice call between 500 Android an desktop sip clients. Can Kamailio handle this or I need an Asterisk server too?
user2085899's user avatar
0 votes
1 answer
1k views

Auto DHCP IP Phones to a different subnet

I'm standing up a 3CX PBX system in my office and would like to set my DHCP to assign IP phones a 10.10.8.x address, while giving everything else a 10.10.0.x address (our current net). What would be ...
Joe JMC IT's user avatar
0 votes
1 answer
2k views

How to have dynamic extensions in Asterisk

I have recently swapped VoIP providers. So far so good, but I have ran into one little snag (and I'm honestly not really sure how to ask / Google this). With my old provider I could have inside my ...
Mattisdada's user avatar
1 vote
1 answer
171 views

VoIP calls dropping

Currently I have a proxy server running pfSense connected to a 40mb fibre lease line. This lease line has 60 external IP addresses (assigned to each room by a virtual IP in pfSense). On the LAN side ...
dynamicuser's user avatar
0 votes
1 answer
822 views

How to move company voip phones to a different subnet by modifying DHCP of a Cisco Switch?

Our company has replaced its old voip phones with new ones which are the NEC phones. We are done with the configuration part, and the phones are working fine. But I want the phones to be in a ...
vincent fernandes's user avatar
0 votes
1 answer
4k 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 ...
David's user avatar
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0 votes
1 answer
40 views

voice vs data device authentication

When a device is plugged into a switch (let's say, a Cisco Catalyst switch) how does a switch know whether the device is a VoIP one - and subject to being placed into the voice VLAN - or some other ...
Boris Epstein's user avatar
0 votes
1 answer
620 views

how do I route Asterisk dials from serverA to serverB?

I'm trying to make an Asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but am having a conceptual problem with ...
Thufir's user avatar
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0 votes
2 answers
2k 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 ...
Thufir's user avatar
  • 229
2 votes
1 answer
8k views

Cisco Voice VLAN with 802.1X Authentication

I've got a Cisco Catalyst 2960 that I'm trying to configure for a remote office with 802.1X wired authentication. The setup I'm going for is Switch -> VoIP Phone via Internal Switch -> PC/...
iamacarpet's user avatar
1 vote
1 answer
2k views

why do all softphones fail to authenticate after receiving 401 unauthorized from Asterisk?

When I attempt to connect to the Asterisk server from CSipSimple, over a Cisco router, on the same network, Asterisk logs show the attempted connection: <-------------> [Mar 23 20:01:34] ...
Thufir's user avatar
  • 229
0 votes
1 answer
2k views

FreePBX UK Dial Patterns Route Through Specific Trunk

I've read a ton of links and tutorials and for some reason cannot get my dial patterns to work correctly. I've got Freepbx installed and working properly - no issue there. I want to create a dial ...
Touff's user avatar
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1 vote
4 answers
5k views

SIP Trunk for VOIP behind a SonicWALL

I appreciate this question is quite "out-there" but - has anyone had any success with VOIP over SIP behind a SonicWALL. We are having an issue with one-way audio inbound. Essentially, the person ...
PnP's user avatar
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0 votes
2 answers
4k 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....
Pablo Santa Cruz's user avatar
0 votes
1 answer
142 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. ...
rob7676's user avatar
0 votes
1 answer
2k views

Asterisk Sip Server and "Screen Sharing" function

I'am newbie in asterisk, i followed the tutorials on the asterisk wiki, and installed the latest version (13) of the asterisk server. I was able to setup the voip now i wonder, there is a function im ...
Raziel's user avatar
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0 votes
0 answers
250 views

Is it possible to virtualize a Genband/Broadsoft M6 voip telephony system?

I work with a company that runs a hosted VOIP platform originally known as the Genband M6 acquired by Broadsoft a few years back. This system is going end of life March 2016 and will no longer be ...
user126715's user avatar
0 votes
0 answers
70 views

How to automate updating of a setting on Cisco phones remotely?

Normally, to make a change one would get the IP address of a Cisco VOIP phone, access its settings via a web browser and make any changes to the settings they need to using the web interface. Is it ...
CCNAman's user avatar
4 votes
1 answer
5k views

Asterisk behind NAT sets wrong Contact Header

I'm using SIP with asterisk 13.1.0 behind a statically configured NAT. The servers private_ip differs from the public_ip, where I can reach it. I've already set these options in the sip.conf file. ...
Lukas Bernhard's user avatar
4 votes
1 answer
318 views

Ping time variation correlates with IP phones not working

We have an IP Phone system (most models are snom 320). Today someone's phone has decided to stop working. It has an IP address assigned to it by DHCP, and I can sometimes go to it's IP address to view ...
Nick's user avatar
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