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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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SIP Sorcery (C#) : How the server can reject call & Receiver can detect that server has rejected a call

SIPUserAgent objSIPUserAgent = new SIPUserAgent(); WindowsAudioEndPoint objWindowsAudioEndPoint = new WindowsAudioEndPoint(new AudioEncoder()); VoIPMediaSession objVoIPMediaSession = new…
Smit Rathod
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OpenSips Remove Header Value

I'm new to OpenSips. I have a requirement to remove a single value from header values. Now it has Allow: INVITE,REFER,ACK,NOTIFY We need to remove REFER from Allow header and leave the rest Expected Result Allow: INVITE,ACK,NOTIFY I have written…
Iniamudhan
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SIP server programming

I'm new to SIP/VoIP programming. I want to implement a SIP server mainly for registering and proxing. I know to start from scratch is foolishness, but then what options do I have? I have been doing google-ing for a couple of days, and I got some few…
Vineet Menon
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Sipdroid Authorization Username vs Caller Id

I am very new in VOIP world and working on SIPDroid for one of my client. At the details of SIP Account Preference window of SIPDroid, I can see there are two username field. One is 'Authorization Username' and the other is 'Username or Caller ID'.…
Foyzul Karim
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How to get Status code for Trying and Ringing in JsSip

How to get Status code like 100 and 180 from response when i call with any number from below response in my console.console output for my call response as below so how to get status code for trying and ringing when i call with nay number. [1]:…
kevin ladani
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Process INVITES when recording is enabled on both parties phones in Cisco CUCM

I am using jain-sip to implement a sip server to process call events and then record the calls in Cisco CUCM. It works fine when a call is made from a recording-enabled phone to a recording-disabled phone or vice versa. I receive two INVITES one for…
Sikander
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Freeswitch users directory using a database

I am quite new to freeswitch and want my directory currently hardcoded in XML config files on the server to be in a relational database. I found this tutorial: https://saevolgo.blogspot.com/2012/07/freeswitch-with-sip-users-in-mysql-mod.html But…
Maxi
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Sending Requests to Locally-hosted OpenSips Server using android

I am really new to opensips and lately I was able to install the OpenSip server on my VirtualBox based VM ( Debian 10). Now I want to create an android application which enables SIP calling (Push to talk app) using that locally hosted OpenSip…
Sachith Kasthuriarachchi
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Installing Siremis stuck on Database Configuration

I have install Kamailio 4.3.4 on Ubuntu 16.0.5 LTS. I want to install Siremis 4.3.0 but stuck on Wizard step 2 (Database Configuration) there is message "Something went wrong". I found error 500 action=update on console inspect element. I have try…
wanz
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Free SIP tool support header normalization

I'm looking to a free SIP software supporting Header normalization. My requirement is simple: this tool receives INVITE message from leg1, save Contact header and route headers (maybe multi-route headers here) and then sends INVITE to leg2 using…
Anh Cao
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Appropriate SIP response for status code 477 when SIP proxy isnt able to reach endpoint

In an implementation, I am running into a case when freeswitch is acting as proxy. It generates 477 as the dispatcher server ahead is down , since i wasnt able to find 477 in sip draft anywhere I assumed to keep the reason as "Send Failed"( based on…
Altanai
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Why can't we use "Local tag" & "Remote tag" to identify a dialogue?

A per the "RFC 3261 Session Initiation protocol", I have read that Call-ID, From and To tags are used to identify a "Dialog". Why we need Call-ID here? Why can't we identify a "Dialog" using "Local tag(i.e From tag)" and Remote tag(i.e To tag)?…
Murali Krishna Bellamkonda
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Opensips suddenly crash in two-three days running

I am using opensips, it is working fine but after 2-3 days it suddenly crash. Don't understand following log CRITICAL:core:receive_fd: EOF on 17 INFO:core:handle_sigs: child process 14090 exited by a signal 11 INFO:core:handle_sigs: core was…
Kamal Panhwar
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c# sip server create

Is there a way to create a sip network with multiple telephone numbers and just one public telephone? Do I need to have a sip server for this? Can someone give me some ideas? If I need a sip sever how can I create one? I am new to this stuff.…
elisa
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SIP : fsockopen doesn't reponse correct/worng user

I have tried many times with multi servers with correct login & password, but i dont get any reponse. $ports = array('5060','5061'); foreach($ports as $port){ $socket = fsockopen("71.170.105.118", $port, $errno, $errstr, 30); fputs($socket, "Action:…
Ely
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