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Questions tagged [sip-server]

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.

The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.

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kamailio sip server with tls

i am trying to configure kamailio on my locak systen . i have installed kamailio on my linux i getting call properly. but when i try to connect by tls getting following errors in log file. ct 21 12:12:48 localhost /usr/sbin/kamailio[3612]: INFO:…
user2916639
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How can I use Twilio as a SIP trunk for my Asterisk to send and receive SMS for my Bria/Xlite?

I recently manage to do inbound/outbound calling on twilio using my Asterisk server, thanks to this topic on stackoverflow. How can I use Twilio as a SIP trunk for my Asterisk to make and receive calls? Now the only thing left is SMS, how can i…
Page Liker
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RTCP communication (sip client-IMS server)

I am testing a setup with one sip client (tested with sipp and a softphone) and an IMS server that auto answers (so it is a two way RTP-communication between those 2 network elements). In RTCP, must both the two entities communicating in such a…
John
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How to register to a SIP server by domain instead of IP address and create a SIP Account from client side?

I've successfully compiled Liblinphone library for android and use it to register to Brekeke SIP server and make calls between Android clients and PC client(x-Lite,linphone). but I'm facing two problems: Client register to the server with their IP…
Wael
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How to figure out when SIP call is started

I'm writing simple SIP-proxy application which stands between Astreisk and SIP client (any softphone). The purpose of the application is to calculate the duration of the call. Below is example of regular flow: Client sends INVITE to SIP-Proxy,…
user12384512
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Call failed using Asterisk

I am trying on 2 PCs, First acts as sip server & client (has asterisk & twinkle installed) and the other as a client only(has twinkle only installed) . I try to make a call between them using Ethernet cable -No internet- so I established the wired…
Sarsoura
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Android Telephony System using Wireless Network

Error here is unreachable statement, but I have imported Sipmanager. Still I'm getting error. public class MainActivity extends ActionBarActivity { public SipManager mSipManager = null; @Override protected void onCreate(Bundle…
Amzad Hossain
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Writing java code for retrieving sip users list

I want to retrieve the SIP users list from Asterisk server onto the SIP client. Asterisk-java API can be used for this purpose but I am not clear on how should I do that? I need to send Action "SIPpeers" to Asterisk AMI which in turn would return…
Pankaj
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Does a SIPS server need a RADIUS server?

This may be a naive question, but I'll ask it anyway. According to http://www.kamailio.org/docs/openser-radius-1.0.x.html, a SIP server should be paired with a RADIUS server for authentication. How about SIPS? Does a SIPS server have a built-in…
Yevgeniy
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Opensips+Nat+RTPProxy in bridged mode

I want to use 2 interfaces one eth0 one tun0(ovpn). client <-LAN-> NAT(router) <-Internet-> opensips <-OVPN-> asterisk <-LAN-> client <---eth0-----> <-----tun0-----> The opensips is running with mhomed=1…
CR7
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Asterisk Realtime and External SIP table

I'm having spleepless night thanks to Asterisk Realtime. I have some trouble understanding the documentation ( like http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip ). Too many tables, many parameters, fragmented informations, no exhaustive…
M4rk
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Kamailio needs to block 200 OK from CANCELed branches, how?

I have a Kamailio 4.0.4 proxy (K) running registrar and tm. There are multiple clients for some AORs and they all auto-accept certain INVITEs which is causing a race condition and having 200 OKs from multiple branches being sent to the…
vaizki
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SIP client register periodically with server

I have written a SIP client with a SIP SDK that you can get online. Classic SIP stuff: you register with SIP server, make calls, get called ... This works all fine, but suppose that the SIP server restarts or for some other reason loses the…
user968698
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SIP multiple 2** responses

i am receiving multiple 2** responses for a call. The problem is that once the call is connected and the server receives the "ACK" packet and the call starts successsfully, server again sends the "OK" response packet back to the callee and recieves…
Zain Ahmed
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Connection failed in QuteCom SIP client

I have chosen QuteCom SIP client for windows to chat.I have installed and configured the account with my public server. My SIP server is kamailio.The connection to the server is not established. The application is connecting to the server for a long…
vijay
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