Questions tagged [sip]
SIP (Session Initiated Protocol) is a protocol used to govern voice and video sessions over IP.
36 questions
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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 '...
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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 ...
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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 ...
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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 = ...
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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 <...
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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 ...
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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'...
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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 ...
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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 ...
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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 ...
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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 ...
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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 ...
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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:...
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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 ...
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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 ...
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