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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14 views

Zoiper softphone cannot register to Asterisk with IAX2: cause connection refused, cause code 29

I am trying to configure a Zoiper softphone (iOS app version) so that it can connect to my Asterisk PBX. The Zoiper configuration is as follows: Account name: Account1 Server: <ip-address-of-my-...
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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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0answers
21 views

Attaching to Kubernetes container (for Asterisk) in GKE leads to endless cycle of attach/detach

I have managed to deploy Asterisk to Google Container Engine (GKE) currently to the extent that a VoIP softphone can register. I'd now like to attach to Asterisk's container. kubectl attach -it <...
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8answers
15k views

How can I simulate a modem over a VOIP connection?

I have a hardware device I need a server to dial into at regular intervals. The problem is I no longer have a POTS line or a modem in any of development computers, and all my production servers are ...
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1answer
1k views

Call duration limit & least cost routing

We have Elastix version 2.4 setup and we have some little questions which if you could please answer to solve via web based panel of Elastix, as agents are making calls via Elastix there are little ...
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1answer
1k views

Asterisk / Elastix Address Book and Call Recording

I am trying to build a small asterisk based PBX using Elastix. It'll be having 4 FXO (2 nos.no ISDN, normal analog POTS, 2 nos. GSM connections using a GSM Terminal) and 4 FXS (2 IP Phones and 2 ...
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3answers
39 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 ...
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1answer
1k views

Is there a traceroute command for VXWorks? Specifically a Shoretel Phone?

I can't seem to find if there is a traceroute command for VXworks online or specifically with a Shoretel phone. Once I telnet into the phone, I would like the ability to see whether packets from my ...
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1answer
32 views

VOIP phone working intermittently

I'm having problems with my Polycom VVX400 phones. It's the first time i've configured them to work with Aircall.io, our SIP provider. I have them registered just fine and I can make calls, but when ...
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1answer
24 views

Call from '201' to extension '202' rejected because extension not found in context 'test'

I have this code on my extensions.conf [test] exten => 20,1,Answer() exten => 20,n,Playback(hello-world) exten => 20,n,Hangup() and this is my sip.conf [201] user=201 secret=123456 type=...
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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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0answers
22 views

Connecting to the PSTN from Voip [closed]

I have 60+ computers running Skype and to reduce costs I have been looking into alternative VoIP technologies. I understand that I can get free software to put on the computers to receive VoIP calls ...
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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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0answers
28 views

VoIP NAT to connect to companies of the same ip address range together

I am currently using packet tracer to configure the network for Company A. I have configured DHCP and telephony service successfully but the problem I'm facing is when company A tries to contact ...
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1answer
33 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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1answer
32 views

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 ...
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0answers
68 views

Cisco ASA PBR and multiple ISPS for multiple SIP Providers

Have a question about policy based routing on my Cisco ASA 5515 running 9.5 and think despite it being available in this software version that I might be screwed. I have an asterisk PBX behind the ...
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2answers
2k views

How can I disable the audible MWI on Polycom phones?

By default, when a Polycom VoIP phone has a voicemail message waiting, it both lights the MWI light, and plays a sound every several minutes. I've never met a more self-important feature in a phone, ...
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0answers
66 views

SIP Extension Registration Port 5060 and Port 1024 1026 1028 etc

First time here. I have a problem registering remote extension on a Multi tenant FusionPBX/FreeSWITCH. I have signed up with a FusionPBX/FreeSWITCH EC2 in AWS for training purposes. I have setup ...
4
votes
1answer
22 views

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 ...
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0answers
145 views

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 ...
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1answer
62 views

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 ...
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0answers
151 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 ...
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2answers
759 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
36 views

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 ...
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0answers
111 views

It is possible to use not NAT able VoIP Softswitch behind pfsense/siproxd

We are using diffrent VoIP Softswitches (asterisk/squire) behind a pfsense Firewall and connect to diffrent SIP Endpoints. The local softswitches only have a local IP and the Endpoints are public IPs. ...
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5answers
3k views

Running a Vonage router behind a Linksys Firewall

I have a Vonage router used for outgoing VOIP calls. It gets a LAN ip by connecting to a Linksys WRT54g via DHCP. From there, it cannot actually get a WAN ip or connect out. What do I need to port ...
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6answers
723 views

Video call software [closed]

In our company we have about 40 offices distributed all over the country. Every time the Human resources department want to do a training session the people have to go to the central offices for the ...
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2answers
44 views

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 = ...
1
vote
1answer
32 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 ...
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votes
4answers
7k views

Prioritise VOIP traffic over a VPN link

We have recently opened a branch office for developers to work out of. The guys there all have Aastra VoIP 'hard' phones on their desks. The main office and branch offices are linked together via a ...
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2answers
2k views

Accept calls on external cell phones - FreePBX Distro

We installed FreePBX distro last week and configured it so that when someone calls in, it goes to all the phones so someone here can answer the call. After trying all the internal extensions, it ...
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0answers
19 views

Screen Share using Opensips and msrp relay

I am using opensips and msrp relay to implement screen sharing between blink clients but getting following errors at msrp relay: debug: session ...
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0answers
51 views

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 ...
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0answers
26 views

How to provide QOS for VOIP in Juniper?

I have 10 computers using browser based VOIP solution and they have problem with the internet connection. they are connected (wire) to Juniper Switches then to Fortigate. How can improve the quality ...
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0answers
98 views

DTMF Tone issue with Elastix

I am very new to Elastix and have just gone through the setup of a new system on Elastix 2.5.0-8, I have Elastix in the middle of 2 sip trunks, one is the service provider and the other is a 3rd Party ...
3
votes
1answer
4k views

QoS settings for Wi-Fi calling on pfSense firewall/gateway?

The new version of iOS 8 as well as Android supports Wi-Fi Calling with multiple cellular providers including T-Mobile in the US. I am currently running pfSense as the primary Internet gateway for ...
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0answers
39 views

Intercom for all phones on Cisco Unified Call Manager

In a Cisco based telephony (VoIP) system (CUCM 9.1.2.11900), I have successfully configured and used the intercom feature. It allows a user (typically a secretary) to push a button and automatically ...
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1answer
90 views

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 ...
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votes
3answers
1k 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
47 views

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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0answers
46 views

Kamailio 4.3 replecing opensips 1.11

We have a customized sip routing script in opensip 1.11 now for good performance and stability we are moving to kamailio 4.3. So here am facing some issues related to kamailio module and funciotns ,...
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0answers
59 views

Route VoIP traffic over VPN between Routerboards?

Context (the Why): We have a couple of remote offices in shared/coworking spaces across the country. We've had significant issues getting our VoIP phones to work with those internet connections (...
0
votes
1answer
71 views

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 ...
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2answers
70 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 ...
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1answer
144 views

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 ...
4
votes
1answer
832 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. ...
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0answers
31 views

What do I need to setup a small scale videoconf system on linux?

I have the task of deploying a small scale (for small groups, up to 15 persons) videoconf system. In principle it shouldn't be that hard, I thought, but after a day of reading I think I miss the main ...
0
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1answer
26 views

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 ...
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0answers
41 views

VoIP phones not registering on second zone - SonicWALL TZ210

Environment: Hosted VoIP SonicWALL TZ210 running SonicOS Enhanced 5.9.1.1-39o (latest) ADSL WAN connection The SonicWALL has two zones (subnets), X0 and X2, LAN and Voice respectively. Both are ...