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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 mohamed
2
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 Castillo
4
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 Castillo
2
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 Castillo
3
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 Rizaldi
2
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 Elzaurdia
3
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
,
Gmail
6
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-Potion
3
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-Potion
2
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 Balashov
5
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 Balashov
2
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 Muller
10
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