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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VOIP in Android..?

I am developing an app where i need to implement VOIP in that. I have gone through some samples in google like SIPDROID, IMSDROID and implemented default android API SIP. They are working good. But here I want to interrupt the message while…
code_finder
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Unable to login and Call SIP/VOIP library doubango with TLS for android?

I am also new to NDK development and I am try to develope application based on SIP/VOIP protocol. Application etc. After searching I found this library which used in application library for app And downloaded and managed code from the given example…
Khan
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SipDemo's onReceive() SipAudioCall NullPointerException give

I am Creating incoming call Receive activity by SipDemo . My activity has two button "accept call" and "reject call". When i got incoming call then my own activity is open such as ----> incoming call class ----> onReceive()----> go to my call pick…
user2160008
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Java and Sip Proxy

I was trying to implement sip redirection. To do this I'm using javax.servlet.sip.Proxy SipFactory sipFactory = (SipFactory) getServletContext().getAttribute(SIP_FACTORY); Proxy proxy =…
Gocha
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kamailio imsdroid one way audio is clear the other way just wind and jitter sound

I have installed Kamailio as sip server/registrar and using default config. I have this situation: mobile cleint using IMSdroid (extension 100) windows client using Linphone (extension 300) Register satus: both clients register successfuly Both…
mitan
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retain original caller id on Call transfer on asterisk

I am running a B2C outbound Campaign on VicidialNow C.E 1.1 as Asterisk Server / SIP Server. The call is made from server to customer and connected to agents waiting for calls. The agents transfers the call to third party (not a blind transfer). The…
user1383088
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asterisk vs freeswitch configuration

Below is an IVR that has been written in lua for freeswitch. I have not copied the whole IVR, just some portion from the start of the file. session:set_tts_parms("flite", "kar"); session:speak("Welcome to the VoIP World!"); …
liv2hak
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how to let the kamailio support Edge Proxy?

As the RFC http://www.rfc-editor.org/rfc/rfc5626.txt describes , how can i extends a Edge Proxy by Kamailio ? should i write a module , or just write the configure file ? have any one already do it ,could give me some advice . thanks .
mike
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how to run my sip servlet in netbeans

I am new in SIP server development. I read a lot about sipservlet and sailfin like SIP servlets, chatserver If I write that code in netbeans I cannot right click on the sip servlet and click run as it told me it doesnot contain main method or it is…
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Want to register my SIP_Server

I want to create sip servlet on netbeans/Tomcat, my purpose is to register with Cisco Call Manager Server with ip [192.168.2.2], I tried multiple servlets but actually i donot know how to run my servlet in netbeans,
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RTP unreachable SIP Phone

I try to create sip phone application on android and I have some problem which is the RTP packet is unreachable to android emulator. I'm not sure about this problem because of IP emulator. I can do SIP Register and can call to other soft phone but…
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FreeSwitch iPhone app communication issue

I have an app based on pjsip for doing some Sip calls. The app works fine with other SIP Switches/servers but with freeSwitch the app is not receiving calls in the background mode. With freeSwitch everything is fine when the app is active but when…
alinoz
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make sample sip server in local host for js sip and web interface

How can i create sip server on local computer (windows) for use js sip javascript. jssip has a demo at "https://tryit.jssip.net/" but it doesn't work.
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Genesys BYOC Purecloud Outgoing Call issue

We recently implemented Genesys BYOC PureCloud in our organization, and we have encountered an issue with outgoing calls not working. I would like to provide some background information on our setup and the problem we are facing. We have a Yeastar…
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Couldn't detect that call is hung up/Cancelled by the user in SIPSorcery C#

I am developing VOIP application in MAUI using SIPSorcery v6.0.12. I am having below status in mentioned cases. The user1 calls user2, The user2 doesn't accepts call and declines it : Success The user1 calls user2, The user2 doesn't accepts call…
Smit Rathod
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