Questions tagged [jssip]

Use this tag for questions related to the JavaScript SIP Library.

With JsSIP you can build a complete SIP user agent in your Web page:

  • Send and/or receive multimedia calls.
  • Send and/or receive text messages.

JsSIP is a SIP WebSocket client. It needs a SIP WebSocket capable server to which connect and exchange SIP messages.

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Error: ast_sockaddr_resolve: getaddrinfo("a783543c-1911-44c4-9ba1-52114bbdccb4.local", "(null)", ...): Name or service not known

We connect JsSip to Astersik and long time all worked perfect. After than unexpectedly voice dissapear without any reason. We see in astersik log next message ast_sockaddr_resolve: getaddrinfo("a783543c-1911-44c4-9ba1-52114bbdccb4.local",…
Vladimir
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How to set STUN servers in JsSIP 3.3.0

I'm trying to set up a webapp using JsSIP 3.3.0 connection to a Asterisk server. I can find some documentation regarding TURN servers in an old version (0.3.0), but apparently this feature was removed in version 0.6.0. I also found this:…
majbom
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JSSIP and React audio issue

So I am using jssip 3.2.10 to make calls on a React project. The server is setup on Asterisk and CentOS. I can make calls where the call receiver hears me well, but I can't hear their audio, nor the waiting (traditional) beep noises it should make…
galgo
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Sender's packet count is very big when we made calls using JsSIP

We are making calls using JsSIP. We are seeing "Sender's packet count" is very high in Wireshark. Can some one explain what is this Sender's count and why this count is so much big?
Cherry
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can't connect to my Asterisk server with jssip library, Wrong Password error

i can connect to my freePbx server with jssip. but registeration got faild with wrong password in Asterisk Logs. i can connect and register with none WebRtc and WebSocket clients with same password for my PjSip Extension. it's work in .net library…
sadegh
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Using jsSIP in A Project

We're using jsSIP in our project and I'm trying to get phone number of the caller when receiving an incoming call. I could't find the answer in the jsSIP documentation. In the above image, i want to take "1004" telephone number. How can i do that?
Jsawyer
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Why do I get 'WebSocket opening handshake was canceled' trying to connect with JSSip?

I am trying to establish connection with JSSip with an already running SIP server. I have followed the documentation and came up with the following code. Here is my code.
MiniGunnR
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Migration sipjs to jssip

I changed lib sipjs to jssip. I have problem on session transfer. in sipjs this look like this session_from.refer(session_to); How i can do this on jssip?
UJin
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Browserify - JsSip

I have a new project where I'm using browserify to convert node modules into an sdk that can run inside the browser. I'm requiring a number of other npm packages like: var log4js = require('log4js'); That run fine and give me no problems in the…
sauce
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JSSIP WebRTC phone auto disconnect after 30 second

I have embed JSSIP http://tryit.jssip.net/ phone to our application and it use Freeswitch for calling, everything words but call getting disconnect after 30 second or so and in Browser JS console logs we are seeing following, In Freeswitch side I am…
Satish
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WebRTC internet explorer support via JsSIP

We are working on call to SIP via JsSIP library, with modern browsers like Chrome Firefox we don't have any poblems, but IE support is hell. Does anyone has usecases with JsSIP in IE?
Egor
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RequireJS + NodeJS returning XMLHttpRequest cannot load 'http' No 'Access-Control-Allow-Origin' header

Im trying to use JsSIP, but when I use the RequireJS to enable require in the client side CHROME returns: XMLHttpRequest cannot load http://localhost/videochat/videonodejs/scripts/jssip. No 'Access-Control-Allow-Origin' header is present on the…
Moisés
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JSSIP/SIP-JS calls dropping out

I am getting a dropout while making a call using jssip/sipjs library. There is no audio too. Following is shown in javascript console. ==== Fri Apr 04 2014 10:14:30 GMT+0530 (IST) | sip.sanitycheck | Via sent-by in the response does not match UA…
Kasinath Kottukkal
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JSSIP application is not hearable in some nets?

Why this can be so, that JSSIP application (connected to Freeswitch) is absolutely silent on some networks and works normal on other? As I could found, namely network matters. Server is always the same, computers are different, browsers are…
Dims
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Opus codec deploy in asterik made unable to call establish

I was running a asterisk 11.5 with no error. After installed Opus codec with patch in my asterisk btw i am using browser based sip softphone jssip Now whenever I am making audio call between 2 peer, no call is getting establish and throwing below…
new developer
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