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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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build SIP server similar iptel.org by kamailio

I want build SIP server like iptel.org. I use this tutorial (http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour) to install Kamailio SIP Server. But I have some problems. Server does not work with UDP.(while I configured…
MJH
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How to get call info from Kamailio

I have setup a Kamailio server and am able to establish calls. I need a way to get call related information like from, to, duration,etc. I have enabled the dialog module in the config but no avail. I am not well versed with config files and I am not…
suchitra nair
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OpenSIP not sending cancel to UAS in case it received 200 from UAC. Verified in 1.7.2 and 1.8

SIP Call Graph Diagram when Bug comes: A = UAC B = OpenSIPS C = UAS A ---------- INVITE ---------> B A <-------- STATUS 100 TRYING ------- B B ---------- INVITE ---------> C B <-------- STATUS 100 TRYING --------- C B <-------- STATUS 200 OK…
Mani
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Run Jitsi SIP communicator in eclipse

I am using Jitsi in my application for chat.I downloaded Jitsi source code through SVN into my eclipse.I have to build this project with Ant and start working.I executed the build.xml using Ant and the build was successful.But when i run the project…
vijay
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call from mobile web browser without registration

Is there any possibility to place a SIP/PSTN call with out having registered from a mobile/desktop browser ,just click to call something like that - No registration if yes , how ? and if yes, is there any possibility to track it and getting user…
mahi
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Implement SIP instant messaging framework

I want to implement an instant messaging framework working on SIP. I have SIP clients and an element that will get all the messages from the clients, and will handle them and forward them to other sip clients. The clients and the managing element…
Elad Aharon
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SIP over TCP versus SIP over UDP [ SIP: Session Initiation Protocol ]

What is the difference between SIP over UDP and SIP over TCP? What does "SIP over TCP" really means? Does it mean: both SIP and RTP use TCP or SIP use TCP and RTP use UDP
Hippias Minor
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problems in outbound calls in asterisk

i am having problem in making calls i have vicidial CE 1.1 i am having following errors -- Remote UNIX connection -- Executing AGI("SIP/sou101-0996c9e0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log…
user1383088
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Outgoing calls via SIP trunk

I have one simple question which stuck me. I have FreePBX server with SIP trunk to ISP. ISP Dedicated one PSTN number. So all internal SIP Extension go out with the same number. Does it mean that at any time just one Extension can call…
Atlas
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SIP getting 407 Response for BYE Request

In my VOIP Application, i am sending Invite request to ( INVITEE_NUMBER ), and getting 200 OK Response when invitee picks up a call, But when i send Bye request to terminate the call, i am getting 407 Response code from the server, Should i treat…
Amitg2k12
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converting freeswitch to asterisk

I have the following code in freeswitch. We have decided to use asterisk instead. I've setup so that if you dial 8XXXX you will dial the other server. sip1:/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml
liv2hak
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Sip INVITE error and send BYE message

Hi i have two questions, 1-) When i send sip INVITE message and get errors bigger than 2xx then if i send BYE message is it reasonable to do this ? Server will response BYE message ? 2-) ACK will be sent for all 2xx status code and dont be sent for…
nihasmata
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asterisk dialplan context clarifications

I have been learning asterisk dialplan and I have created a very simple IVR menu.I would like someone to verify that what I have created is correct.I want to create a very simple IVR with 2 menu levels and an exit option. [incoming] exten =>…
liv2hak
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asterisk dialplan clarifications

Below are some asterisk dial plan examples that I have copied from somewhere. exten => s,1,Answer() exten => s,n,Playback(hello-world) exten => s,n,Hangup() The first line indicates when a new call comes into the channel it goes to extension s (top…
liv2hak
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Asterisk dial plan priority

I have an example asterisk dial plan below.It just the main (no extension or start) and it has 3 priorities.In the first priority of our extension, we’ll answer the call. In the second, we’ll play a sound file named hello-world.gsm, and in the third…
liv2hak
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