JsSIP

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·~· JsSIP - the JavaScript SIP library ·~·

Welcome to JsSIP mailing list.

Website: jssip.net
Mailing list address: jssip@googlegroups.com

NOTE:

Please avoid questions about Asterisk or FreeSwitch and their WebRTC implementations.

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Pikachu Pikachu

Jan 27

I dont understand how to implement DTMF

Hello! I need a help How add DTMF in this situation? Init: function initUA() { window.ua = new JsSIP.

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I dont understand how to implement DTMF

Hello! I need a help How add DTMF in this situation? Init: function initUA() { window.ua = new JsSIP.

Jan 27

Tobias Wendorff

Jan 27

Chrome/Chromium allows not asking for mic

Hi there, in a project, I just wanted to listen the other side without having to give access or even

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Chrome/Chromium allows not asking for mic

Hi there, in a project, I just wanted to listen the other side without having to give access or even

Jan 27

Karan Mamtora

Jan 18

react native jssip issue with release apk

when we create a release apk it doesnot register

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react native jssip issue with release apk

when we create a release apk it doesnot register

Jan 18

nicolas cardone

Jan 18

transmit audio file to the other end.

hello everyone. I want to send an audio file to the other end of the call this is what i try, const

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transmit audio file to the other end.

hello everyone. I want to send an audio file to the other end of the call this is what i try, const

Jan 18

ahmed mohamed2

Jan 3

jsip client with Asterisk 18 wevrtc one way audio please hepl me

sip debug <— SIP read from UDP:10.11.63.77:5068 —> INVITE sip:20...@10.12.39.223;user=phone SIP/

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jsip client with Asterisk 18 wevrtc one way audio please hepl me

sip debug <— SIP read from UDP:10.11.63.77:5068 —> INVITE sip:20...@10.12.39.223;user=phone SIP/

Jan 3

Dnyaneshwer Pendurkar

Jan 2

Error: node_modules/jssip/lib/RTCSession.d.ts:1:23 - error TS2305: Module '"events"' has no exported member 'Listener'.

I am getting following issue while using jssip in angular. My angular version is 14. Can somebody

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Error: node_modules/jssip/lib/RTCSession.d.ts:1:23 - error TS2305: Module '"events"' has no exported member 'Listener'.

I am getting following issue while using jssip in angular. My angular version is 14. Can somebody

Jan 2

Christian Checcaglini

12/23/22

"This tab is using your camera or microphone"

hello, i created a webphone using this application and works very well, however i got several

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"This tab is using your camera or microphone"

hello, i created a webphone using this application and works very well, however i got several

12/23/22

Maxi

12/22/22

BFCP protocol implementation

Do you plan to implement?

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BFCP protocol implementation

Do you plan to implement?

12/22/22

plus liao

12/13/22

'Socket' is not exported by lib/WebSocketInterface.d.ts, imported by lib/JsSIP.d.ts

When the commit id is 004d160, move the Socket and WeightedSocket of Websocket Interface.d.ts to

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'Socket' is not exported by lib/WebSocketInterface.d.ts, imported by lib/JsSIP.d.ts

When the commit id is 004d160, move the Socket and WeightedSocket of Websocket Interface.d.ts to

12/13/22

Maxi, Iñaki Baz Castillo4

12/13/22

Switching the video via renegotiate don't work

What? El El mar, 13 dic 2022 a las 8:34, Maxi <prolope....@gmail.com> escribió: I am not

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Switching the video via renegotiate don't work

What? El El mar, 13 dic 2022 a las 8:34, Maxi <prolope....@gmail.com> escribió: I am not

12/13/22

Hamad Mohsen

12/12/22

How can I use JsSIP with node

I'm trying to make a desktop application that supports SIP over WebSockets/Webrtc calls. However,

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How can I use JsSIP with node

I'm trying to make a desktop application that supports SIP over WebSockets/Webrtc calls. However,

12/12/22

lang qiu

12/6/22

Can't get debug log any more

I can't figure out since when I can't get debug log any more. No mater I run JsSIP.debug.

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Can't get debug log any more

I can't figure out since when I can't get debug log any more. No mater I run JsSIP.debug.

12/6/22

Peter James, Iñaki Baz Castillo2

11/25/22

Session

Ask in sip.js forum, not here. El El vie, 25 nov 2022 a las 11:37, Peter James <webdialer.

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Session

Ask in sip.js forum, not here. El El vie, 25 nov 2022 a las 11:37, Peter James <webdialer.

11/25/22

Juan Pablo, Iñaki Baz Castillo3

11/16/22

refer event listener does not get executed after calling refer()

Yes, you're right, I'm using Asterisk. Good to know that's the problem, I'll have

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refer event listener does not get executed after calling refer()

Yes, you're right, I'm using Asterisk. Good to know that's the problem, I'll have

11/16/22

Peter James

11/16/22

WebDialer hold function

Hello Guys, I am a new person joined here. I think this would be a great platform where for each

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WebDialer hold function

Hello Guys, I am a new person joined here. I think this would be a great platform where for each

11/16/22

Ravhi Rizaldi

11/8/22

Connect JSSIP to FOP2 Web

Hi, How to listen/whisper using JsSIP to FOP2? Usually I can listen using click2dial (using softphone

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Connect JSSIP to FOP2 Web

Hi, How to listen/whisper using JsSIP to FOP2? Usually I can listen using click2dial (using softphone

11/8/22

Russell Harrower, Ravhi Rizaldi2

11/6/22

Multiple lines

Hi, I think conference call feature is already on asterisk. You need to setup it first, it's not

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Multiple lines

Hi, I think conference call feature is already on asterisk. You need to setup it first, it's not

11/6/22

Mike Remski

10/17/22

Reinvite, merging local and remote SDPs

JsSIP initiating an INVITE, gets a remote ANSWER, then triggering a REINVITE off of the RTCSession.

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Reinvite, merging local and remote SDPs

JsSIP initiating an INVITE, gets a remote ANSWER, then triggering a REINVITE off of the RTCSession.

10/17/22

Juan Carlos Gonzalez Cardona, … Juan Elzaurdia3

10/15/22

Q&A

Asterisk webrtc with JSSIP 3.2.4 dont close connection

Hi Juanca, hope youre doing fine. Regading on this issue you faced, im in the same path, and right

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Q&A

Asterisk webrtc with JSSIP 3.2.4 dont close connection

Hi Juanca, hope youre doing fine. Regading on this issue you faced, im in the same path, and right

10/15/22

ideanet help, Gmail6

10/11/22

screen sharing and recording

Thank you. Let m have a look On Tuesday, October 11, 2022 at 8:46:04 PM UTC+13 mahmood19...@gmail.com

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screen sharing and recording

Thank you. Let m have a look On Tuesday, October 11, 2022 at 8:46:04 PM UTC+13 mahmood19...@gmail.com

10/11/22

WebRTC, Super-Potion3

10/5/22

Modify SDP or change Codecs in SDP

Hello, the only solution is to have a gateway like OverSIP, OpenSIP, FreeSwitch, Asterisk, or some

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Modify SDP or change Codecs in SDP

Hello, the only solution is to have a gateway like OverSIP, OpenSIP, FreeSwitch, Asterisk, or some

10/5/22

ARMAN SINGH BALIYA, Super-Potion2

10/5/22

Incompatible SDP issue with telnyx sip points

I'm facing the same issue, did you ever fix this? On Friday, July 26, 2019 at 8:51:29 AM UTC-4

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Incompatible SDP issue with telnyx sip points

I'm facing the same issue, did you ever fix this? On Friday, July 26, 2019 at 8:51:29 AM UTC-4

10/5/22

Tech ind

9/30/22

Web-Dialer by JsSIP libraries

Hello guys, I am the new member joined in this forum. I am in the process of developing Web-dialer by

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Web-Dialer by JsSIP libraries

Hello guys, I am the new member joined in this forum. I am in the process of developing Web-dialer by

9/30/22

Matias Ellera, … Alex Balashov5

9/24/22

is there a way to respond to a sip 100 trying?

It should be noted that 100 Trying isn't considered a progress indication. It's sent

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is there a way to respond to a sip 100 trying?

It should be noted that 100 Trying isn't considered a progress indication. It's sent

9/24/22

Kamal Radwan, Alex Balashov2

9/22/22

WebRTC SSL Validation Error

Hi, You can go to the https:// URL for your websocket endpoint in the browser, and follow the

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WebRTC SSL Validation Error

Hi, You can go to the https:// URL for your websocket endpoint in the browser, and follow the

9/22/22

Mike Remski

9/21/22

mediaConstraints in ua.call options

One of our use cases we want an SDP with an m video line but no audio line. According to the JsSIP

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mediaConstraints in ua.call options

One of our use cases we want an SDP with an m video line but no audio line. According to the JsSIP

9/21/22

Sebastian Gutierrez, … Corrie Muller10

9/15/22

Q&A

INVALID_STATE_ERROR breaks jssip

Hi I'm experience this issue how did you resolved this issue? On Friday, 12 April 2019 at 09:16:

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Q&A

INVALID_STATE_ERROR breaks jssip

Hi I'm experience this issue how did you resolved this issue? On Friday, 12 April 2019 at 09:16:

9/15/22

Matthias Schicker

9/15/22

Is https://tryit.jssip.net/ down?

Hi everybody, it seems like `https://tryit.jssip.net/` is no longer working, the WebSocket connection

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Is https://tryit.jssip.net/ down?

Hi everybody, it seems like `https://tryit.jssip.net/` is no longer working, the WebSocket connection

9/15/22

David Melo

9/9/22

dash/hyphen being stripped from username in URI

Looking through the grammar in docs and repo, it seems that a hyphen would be valid, however it seems

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dash/hyphen being stripped from username in URI

Looking through the grammar in docs and repo, it seems that a hyphen would be valid, however it seems

9/9/22

Karthik Kumar

9/8/22

SIM to Mobile App

Hi, I am in the process of building a dialpad mobile application. The application allows the user to

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SIM to Mobile App

Hi, I am in the process of building a dialpad mobile application. The application allows the user to

9/8/22