Questions tagged [sip]

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

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Openwrt and sipproxy why I am unable to perform a call via a softphone?

I have setup in my router openwrt and I try to perform a call using my provider's VOIP settings that I retrieved for its router that provided. My network settings are the following: config interface '...
Dimitrios Desyllas's user avatar
1 vote
1 answer
505 views

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 ...
Ricardo's user avatar
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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 ...
dnld's user avatar
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NodeJS child_process.spawn() behaving different when run as systemd service on Debian 10

I am working on a NodeJS application, that runs expressJS and uses twinkle to dial a telephonenumber. Given the following function: export const call = (telNr: string, res: Response|undefined = ...
Marcel Kohlmeyer's user avatar
1 vote
2 answers
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No module named 'PyQt4.sip'

So, I have a .py file: from PyQt4.QtGui import * from PyQt4.QtCore import * When I'm executing it - I have this error message: Traceback (most recent call last): File "kek.py", line 1, in <...
Curlindus's user avatar
2 votes
1 answer
1k views

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 ...
Juan Pablo Gomez's user avatar
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Dual Network Gateway on CentOS 6.7

I have dual NIC machine running CentOS 6.7 and asterisk. First NIC is for LAN & Internet connectivity and second is for trunk provider's connectivity. Both of these have gateways configured. I don'...
user1263746's user avatar
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Logging of failed SIP calls (sipcmd) on a Linux box (Debian)

I have set up a little Raspberry Pi (with Debian 8) behind a router (Fritz!Box), which does check/analyse the connectivity or rather quality of service of a certain phone line per SIP calls. My phone ...
vega's user avatar
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2 votes
1 answer
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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 ...
AnyDev's user avatar
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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 ...
Dercni's user avatar
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Iptables (port forwarding from vps openvpn server to vpn client)

I install openvpn server in Centos VPS. I can connect from my pfsense router. I forwrad rdp a port to my local pc, but can not forward rtp port. iptables -t nat -A PREROUTING -p tcp -m tcp --dport ...
Khandaker Shahriar Amin's user avatar
1 vote
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204 views

Debian configure OpenSIPS for Websocket and UDP

I'm trying to configure OpenSIPS with OverSIP so that I could make a SIP call between a webrowser and a sip client app like Yate or Linphone. The call between two client apps (Yate/Linphone) works ...
Laci K's user avatar
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2 answers
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Installing pyqt5 on linux failed due to SIP dependency

I am trying to install pyqt5 on linux; $ cat /proc/version Linux version 4.11.4-1.el7.elrepo.x86_64 (mockbuild@Build64R7) (gcc version 4.8.5 20150623 (Red Hat 4.8.5-11) (GCC) ) #1 SMP Wed Jun 7 12:...
so.very.tired's user avatar
4 votes
2 answers
1k views

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 ...
sibislaw's user avatar
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Asterisk: SIP and IAX registration failed on remote connection [duplicate]

I set up two asterisk servers (on Fedora) in different networks. My goal is to make a call from softphone (on windows lite with ip: 192.168.20.3) to the asterisk server 2 which is in the other network ...
Y. Dabbous's user avatar
1 vote
1 answer
2k views

NAT configuration for EIP (Elastic IP) in Asterisk

When running Asterisk on Amazon EC2 with an EIP, what are the NAT configurations for Asterisk? [general] nat=yes externip=xxx.yyy.zzz.vvv localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses ...
Thufir's user avatar
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1 vote
2 answers
997 views

Voice-over-IP with a VPN

I use an IPFire (linux) router as a VPN client. My voice-over-IP is working properly without the VPN active. However, when the VPN is up, my SIP client (Twinkle) can't register with my VoIP provider. ...
MountainX's user avatar
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3 votes
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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@...
Thufir's user avatar
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How can sipsak simulate a call with usrloc?

Building on a stackoverflow question, how can I actually get the device to ring? thufir@mordor:~$ sudo sipsak -vv -s sip:6003@localhost No SRV record: _sip._tcp.localhost No SRV record: _sip._udp....
Thufir's user avatar
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How to send a ghost email message?

When I was running tcpdump and watching output on my new created VPS (OS:CentOS 7.1) via SSH, a message appears suddenly: OPTIONS sip:100@My_VPS_IP SIP/2.0 Via: SIP/2.0/UDP 69.64.57.15:5596;branch=...
TJM's user avatar
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How to use a tty as a sound device for pjsua?

I have a bi-directional PCM device that appears as /dev/ttyUSBxx. It provides and consumes 16-bit linear PCM at 8KHz (though that can be changed to u-LAW or a-LAW). I would like to use the device in ...
Clay's user avatar
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1 vote
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623 views

Capture screen and output to video device

I'm using linphone, I I'd like to share my screen instead of my webcam. Currently, it looks like linphone only supports reading video input from /dev/video using V4L, though I'll be happy to be ...
Nathaniel Bubis's user avatar
0 votes
1 answer
3k views

Asterisk v13 on Kali Linux: No RTP engine was found. Do you have one loaded?

I installed Asterisk and tried to make a call with zoiper but I get an error [call failure 401 forbidden] and Asterisk return this message : Connected to Asterisk UNKNOWN__and_probably_unsupported ...
Y. Dabbous's user avatar
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1 answer
192 views

How to effectively capture traffic on port 5060 between specific time internals?

users. I want to capture traffic on port 5060 for one week in a file, than wants wireshark to rotate file and start capturing for next week with a different file name. I know little about tshark, ...
user3310052's user avatar
1 vote
1 answer
1k views

SIPp "hello world" message to Asterisk

How do I run diagnostics against Asterisk? Asterisk is running on tleilax; and doge is on the same network (My network topology isn't optimal). Specifically, I want to do something like: sipp 345@...
Thufir's user avatar
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how to connect a softphone to asterisk

To give context, here's my network: On the tleilax server, there's clearly a result from sipsak on doge: tleilax*CLI> tleilax*CLI> core show version Asterisk 1.8.29.0-vici built by abuild @ ...
Thufir's user avatar
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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 ...
ukhlof's user avatar
  • 29
5 votes
1 answer
613 views

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 ...
r0berts's user avatar
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2 votes
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461 views

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 ...
ndandanov's user avatar
3 votes
0 answers
320 views

VoIP dialer for SIP

I have some Java, so am biased in that direction, but not overly so. None of the Java suggestions: https://stackoverflow.com/questions/16471795/sip-client-for-java https://stackoverflow.com/...
Thufir's user avatar
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3 votes
1 answer
1k views

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 ...
fer y's user avatar
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0 votes
1 answer
772 views

Bad SIP URI in To header field

We are creating Asterisk<->Lync trunk. The Asterisk is under providers control. I am in charge of Lync. I'm slightly familiar with Asterisk. We created a trunk. Everything was fine until provider ...
Anton Moiseev's user avatar
1 vote
1 answer
2k views

How do I get my phone to ring using Linphone and MagicJack?

I have a VOIP number setup with Linphone and IPKall. I am using my old magicJack to connect my handset. The handset works for voice input and output, but when I receive a call, it only rings in the ...
TheCodeGeek's user avatar
3 votes
1 answer
321 views

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) ...
daniel's user avatar
  • 31
4 votes
1 answer
2k views

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's user avatar
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8 votes
7 answers
4k views

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 ...
LanceBaynes's user avatar
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