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Questions tagged [sip]

SIP (Session Initiated Protocol) is a protocol used to govern voice and video sessions over IP.

37 questions
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Secure FOSS alternative to Skype on Linux & OpenBSD?

Criteria: Makes audio/video calls Encrypts the whole traffic (using good encryption) Is cross-platform (including Windows 7, etc.) Runs on modern Linux distributions (Fedora, Ubuntu, etc.) Runs on OpenBSD Does anybody know a good Free and…
LanceBaynes
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Easy voice talk between 2 computers in 2 countries?

My plan is to enable 2 sets of unsophisticated users (children and their grandparents) in 2 different countries to talk easily. In practice I see it as: A quiet Linux computer is on all the time in both places. The user at one computer should be…
r0berts
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How to turn off RTP buffering for SIP calls in FreeSWITCH pbx software?

I want to turn off buffering of SIP calls in freeswitch pbx software. Freeswitch holds RTP data from clients in buffer and sends it every 20ms. I want to freeswitch pass throught packets without holding. How to configure it? ----- EDIT (additional…
sibislaw
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Send a fax via SIP

How can I send a fax from my Linux machine using a SIP connection? I don't have a modem. Ideally, I'd like to provide a PDF or TIFF file, and send it via my SIP provider. Is this possible?
Bob
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Fedora16, ekiga just works in superuser mode

I've got a fedora16 running and I'd like to use ekiga as a softphone. Now I'm running into some problems. Starting ekiga as a normal user gives me could not register (globally not acceptable) So I was pretty stuck. Now I tried running ekiga with…
daniel
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how do I make a SIP phone ring with sipsak?

How can I use sipsak to simulate a call so that there's an actual ring, or possibly even a sound is played? thufir@mordor:~$ thufir@mordor:~$ sudo sipsak -v --usrloc-mode --from sip:demo_alice@localhost --password 123 --sip-uri…
Thufir
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VoIP dialer for SIP

I have some Java, so am biased in that direction, but not overly so. None of the Java…
Thufir
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listen to a conversation using port forwarding

I have been able to make a phone calls using the SIP client Pjsua from one Linux computer to another one. In case you have not heard of this user agent, explaining it's functionality is quite easy. It uses IP and port number of each linux to create…
fer y
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Configure network interfaces for sip trunk

I have a Debian 9.5 server that I'm trying to use as a PBX server with a sip trunk, this machine has two network interfaces, one pointing to LAN another one pointing to my sip provider. This is the configuration: iface LAN inet static address…
Juan Pablo Gomez
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nf_conntrack_sip does not work SOMETIMES, restarting iptables USUALLY fixes it

I'm trying to use nf_conntrack_sip on box that is running Asterisk, that is, not routing traffic for another VoIP box. Setup works until I reboot. After reboot nf_conntrack_sip ALMOST always stops working and media traffic is dropped. conntrack…
AnyDev
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Unable to register SIP via WiFi

We run a FreePBX server on our LAN and softphones can register using the local SIP server IP. I need these softphones to be able to register over the internet too so we have configured the firewall and created a dns entry for sip.ourdomain.com.…
Dercni
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how to do a sip call from sipp

I am newest user using SIPp , I am trying to made a sip call from SIPp program on Linux but i fail on it . i want a command to call phone number (14042835351) from sip domain (80.81.82.83) with same username and password (2004) . this is a Tutorials…
ukhlof
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Chat application for monitoring with scripts

I am looking for a chat application, which would have the ability to interact with a shell script. I need this for a home project. An example of usage: A client sends a chat message to a server. It analyzes the incoming message and takes an action…
ndandanov
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IPTables to limit high "Call-Per-Second" and redirect to another program (same machine)

I'm looking for a way to "control" high volume of SIP VoIP INVITEs (UDP) per second (Call Per Seconds) at iptables level in my VoIP server reaching port 5060. What i need to do is limit amount of INVITE's per second to a certain rate (for example…
Ricardo
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FreePBX No connection to Asterisk

I am new to asterisk and freePBX. I set up a server years ago but it was very basic and used for less than a month. I’m now launching a new company and found myself in need of a VOIP system so I’m back to asterisk. I installed asterisk on my debian…
dnld
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