Session Initiation Protocol is one of the protocols used for controlling sessions (signaling) in internet telephony.

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SIP address for groups in Lync

For opening the chat window with Microsoft Lync, we can call either Lync SIP:user@company.com or SIP:user@company.com However I am looking for single command for open a group chat. It ...
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15 views

no sip account work with any application but linphone account work with all, so, what to do?

I want to use sip to call my friends and family so first I installed jitsi on windows7 64-bit laptop then I registered with linphone which work perfectly, but, I want another separate account from ...
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1answer
38 views

Asterisk SIP calls ending after 2 minutes using Twilio's trunking service

I have an Asterisk server hosted on DigitalOcean that is having calls drop after exactly 120 seconds using Twilio's trunking service. It is notable that directly connected softphones do not drop their ...
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1answer
47 views

How to call a PSTN number from a soft SIP Phone through Asterisk

I am getting started with Asterisk. All I want to do is to call a PSTN number from a SIP softphone. I have gone through the latest edition of the book: Asterisk, the Definitive Guide and done the ...
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2answers
60 views

Jitsi and Telekom Germany - outgoing calls “Call failed - Not Acceptable”

I am trying to configure Jitsi to operate as a softphone for my Telekom Germany VoIP-connection. So, I went to the website of Telekom Germany, which provides the following details for connecting: ...
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1answer
28 views

Make one way SIP video call

Using Asterisk, CSipSimple and MicroSIP I was able to make a video call, but to make it work both parties should click on "video" and video will be transferred in both directions. I do not like this. ...
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14 views

Asterisk server asks peer on other server to Authenticate

I am trying to have an SIP peer on one asterisk server call an SIP peer on another asterisk server. I have SRTP enabled between the two servers not the SIP clients. When I initiate the call, i.e SIP1 ...
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1answer
38 views

Do I need STUN?

My router has a "real" static outer IP address. Behind router's NAT there is an Asterisk server. Router is configured to forward 5060 UDP port to Asterisk machine. Do I need STUN server assuming that ...
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65 views

SIP: The other party can't hear me but I can hear them when using Asterisk

Please read carefully all the question and do not rush to a conclusion. === Test environment for point-to-point calls, without Asterisk === City1: A router with NAT and WiFi and 5060 UDP port ...
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1answer
87 views

Failed to authenticate SIP peer

I am trying to connect an SIP peer using Zoiper to my asterisk server. The peer is a soft-phone on my server. I have the following config for the peer: [201] disallow=all allow=alaw ...
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1answer
34 views

What's Lax and Lax2 protocols?

Are LAX and LAX2 communication protocols like SIP? What are these? Some telephone systems support them beside SIP but I cannot find information on google
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23 views

Why do Ekiga touchtones only work with the RFC2833 setting?

I have Ekiga acting as a softphone for my VOIP account. I made a call to an automated switchboard which required me to "press 2 to continue". I attempted to press "2" on the keypad pane but the DTMF ...
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2answers
69 views

How to get and install g723 codecs for sip applications?

I install an ubuntu desktop in my company network (windows network) that use SIP on software phone and hard SIP phones. I use Jitsi that is a good client for SIP and works well with others software ...
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0answers
27 views

Interactively play sound clips in VoIP call

I'm looking for a solution to digitally mix sound clips in a SIP or IAX VoIP conversation. Ideally I want a solution on Linux platform. My end game is to have a sound board to play funny clips while ...
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0answers
63 views

No audio after SIP reinvite

I am trying to get a Gigaset C470 IP to work on my Internet connection. I have figured out the correct configuration for the device and my router (a pfSense sitting between the DSL modem and the LAN) ...
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0answers
14 views

VoipMonitor - how to limit number of MOS calculations

VoipMonitor begins to calculate a given call's MOS score right after the call ends. If there are many calls that drop at the same time, VoipMonitor will spin up many threads to calculate the MOS ...
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29 views

IPTables and SIP Proxy-Authorization Header

Is it possible for iptables to check SIP packets for Proxy-Authorization headers, and if there's more than one header in a SIP packet, modify the packet so only the latest Proxy-Authorization header ...
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46 views

set Cisco SPI firewall to high, but allow SIP traffic?

How can I set the SPI firewall settings to "high" for the router, and yet allow SIP traffic? If I set the firewall to medium, the call will connect but I don't hear anything.
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41 views

Ekiga softphone shows as “registered” — but is it actually?

Is Ekiga actually registered? How would I know? I think Ekiga connects, or at least it says "registered": But, when I exit Ekiga, and use SFLphone, it fails to register: While sipsak shows a ...
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134 views

ekiga — could not register (Forbidden, please check that the username and password are correct)

What are the connection parameters for ekiga? I went to ekiga.net and reset my password, still no luck. I've tried more than a few variants: Add a SIP account To add a SIP account, simply click ...
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1answer
67 views

topology for an unmanaged switch (firewall)

I'm considering an unmanaged switch. My understanding is that many of these devices explicitly lack a firewall. In that case, what do you use as a firewall? I suppose one option would be to use a ...
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1answer
79 views

Not registering on Asterisk over VPN

I want to work in local area from home, I configured OpenVPN on my Asterisk server (it is only one Linux server in LAN), it works fine. I enabled forwarding on Asterisk server, and client now can ...
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1answer
152 views

Getting Asterisk PBX to do SIP URI dialling properly?

I've installed AsteriskNOW in a VM and I'm having a hard time getting calls from the PBX to head outbound to another SIP address on the SIP2SIP network I use for occasional testing purposes. However, ...
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1answer
207 views

Plivo SIP problems with NAT traversal

Ok, I've looked and looked, and can't seem to find anyone that's written about this problem. I have a simple office setup where there are two SIP phone extensions (Polycom 331) behind a Sonicwall ...
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1answer
132 views

Can Asterisk accept and send SMS?

Is it possible to send and receive SMS over Asterisk? I am using Asterisk for my communications and it works pretty good. I just wish that I could send SMS to my callers. thanks
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209 views

Installing SIP-4.16.4 and Py3.4-Qt5.3.1-x64 on windows 8.1

I have unpacked sip-4.16.4.zip from riverbankcomputing and ran configure.py from within the sip-4.16.4 directory. In Python34 I now have the folders sipgen and siplib along with sipconfig.py and a ...
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144 views

How to debug “SIP registration failed”?

My SIP phone, which I use from Turkey with a virtual UK number, has recently stopped working with "SIP registration failed" and the supplier of the UK number has told me the times of the last ...
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1answer
839 views

Android won't register SIP on FritzBox router

In the past I've used VoIP/SIP apps to call from my Android phone. Recently I got the upgrade to Android KitKat (4.4.x) and I've noticed that it includes SIP accounts natively (accessible through ...
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187 views

FreeSWITCH simple configuration for audio call between two computers

I have two computers and I want to make audio calls between them. Each one of them is behind a different NAT. I am an absolute beginner and I have no idea what I need to configure. After reading and ...
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130 views

how to check if a line is pri or sip trunk

I know this is kind of stupid but i have to clear my doubts. I have 2 port digium card and lines for each port with 30 channels each .One line is pri and other for sip .Both lines channels are ...
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17 views

Make a SIP address call

I have a Polycom Deskphone (model 321) that uses SIP/VOIP to make calls etc via my provider. Calling normal numbers works fine as they are then translated by my provider & routed accordingly to ...
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183 views

Sort out - kamailio rtpproxy-ng and mediaproxy-ng configurations

I have been trying to setup Kamailio for calling between webrtc and classic sip phones. I have been able to setup websockets module and webrtc to webrtc calls are working fine (though chrome to firfox ...
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42 views

tcpdump prints nothing while trying to listen communication with sip server

The following command sudo tcpdump -nqt -s 0 -A -i eth0 port 5060 works and prints incoming SIP messages. Now I want to see entire communication and entering the IP address of the SIP server: ...
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1answer
3k views

Failed to make SIP calls (Asterisk + PhonerLite + Zoiper + Blink)

I've installed Asterisk PBX on a VirtualBox virtual machine, then created 3 phone extensions: 101 - PhonerLite, 102 - Zoiper, 103 - Blink all of them are configured as "friends". All the clients ...
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1answer
125 views

Why isn't my router allowing my PAP2T to connect to my VoIP service?

My tech knowledge is middle-low: as you can see I'm not shy about getting into the interfaces for my router, but have no idea what the more complex settings mean. I got a new wireless router (D-Link ...
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2answers
128 views

What is SIP, in plain English? [closed]

I have tried reading the Wikipedia page for SIP but I can't follow at all (incidentally, it is currently marked as too technical). What exactly is SIP? I understand how it is a protocol for ...
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1answer
86 views

How to create a proxy to access sip which is IP locked

Here is what i want to do , I have a sip phone configured on my home system which only works on my static IP , Using zoiper soft phone. I want to be able to connect to my system remotely and be able ...
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2answers
412 views

What happens when two SIP clients log in at the same time

If two SIP clients are logged in to a service provider at the same, what happens when an inbound call comes and what happens if you try to make an outbound call? (This question is regarding SIP ...
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3answers
2k views

Why can't SSH tunneling be used for SIP VOIP encryption?

On http://www.voip-info.org/wiki/index.php?page_id=2716 it says "Note that SSH tunneling is not a viable method for VoIP media path encryption." Why can't SSH tunneling (as opposed to a VPN) be used ...
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2answers
119 views

Asterisk Configuration on wireless network

I have installed Asterisk in my laptop on Ubuntu. Afterwards I created wireless network on the same machine (on which I have installed Asterisk). Next I installed 3CX softphone client on two laptops ...
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1answer
536 views

Traffic Route mismatch with two NIC Windows Server 2008

I have small issue on my network I have two NIC, NIC 1 Connected to the internet and local network, NIC 2 Connected to Sip Trunk Circuit (No Internet) Each one of them has different Gateway. My ...
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1answer
267 views

Windows based Virtual Modem software

I am trying to find a virtual software that will allow multiple users to "dial in" to a machine at any given time. However my efforts have been stopped at every step of the way. The phone number we ...
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0answers
950 views

Android SIP / VoIP with doorbell: View video before answering

I am currently using the 2N Helios IP Vario doorbell device. It can be used via SIP protocol, but can also communicate with the 2N Android app directly, without an additional server, which I would ...
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2answers
770 views

How do I make a SIP call from behind an HTTP Proxy?

We browse websites using an HTTP proxy; there's no direct internet access. The only VoIP program that works out-of-the-box in our network is Skype and it's using that proxy too. For example, I want to ...
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0answers
231 views

pfSense allowing inbound sip phones, but not sip trunk

I am currently trying to get an inbound trunk working on a remote Asterisk machine. The ip phones are remote and connecting just fine through port 5060 (I can make internal calls to voicemail, other ...
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2answers
10k 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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2answers
252 views

How to Implement a VoIP Auto (speech) Dialler

Can anyone suggest/recommend a way to implement a speech dialler using a SIP VoIP service? The software/VoIP equivalent of this I want to dial out and play a pre-recorded message as part of an alarm ...
0
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1answer
280 views

Cisco SPA112 ATA Only Rings Half the Time

I have a Cisco SPA112 ATA Unit which I have set up to connect to an online SIP account (no LAN Asterisk servers in use) and I have this issue where sometimes the unit will not ring when someone calls ...
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0answers
614 views

SIP plugin for Trillian

I like using Trillian for chats (Skype, Google Talk, ...) but I cannot add a SIP Account to make calls with Trillian directly. Google always tells me to download the SIP Plugin for Trillian, but this ...
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1answer
326 views

Firefox can't find the file at sip: xxxx

Hi have a voip service that works great and use Bria Counterpath softphone - also good. I have installed a click to call add on for Firefox and keep getting this error: Firefox can't find the file ...