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discuss-doubango
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Craig Marston
5
Apr 4
WebRTC Calls from Safari 15
Hi Craig, Thanks a lot! On Tuesday, April 4, 2023 at 10:57:07 AM UTC+5 Craig Marston wrote: We used
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WebRTC Calls from Safari 15
Hi Craig, Thanks a lot! On Tuesday, April 4, 2023 at 10:57:07 AM UTC+5 Craig Marston wrote: We used
Apr 4
Virendra Bhati
2
Apr 3
SIPML5 Demo calls URL always faulty
Any update team.. On Wednesday, March 29, 2023 at 4:58:19 PM UTC+5:30 Virendra Bhati wrote: Dear Team
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SIPML5 Demo calls URL always faulty
Any update team.. On Wednesday, March 29, 2023 at 4:58:19 PM UTC+5:30 Virendra Bhati wrote: Dear Team
Apr 3
Craig Marston
Mar 26
Re: [Doubango] Digest for doubango@googlegroups.com - 2 updates in 1 topic
Yes I will ask my dev chaps tomorrow for you. On Sun, 26 Mar 2023, 15:26 , <doubango@googlegroups.
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Re: [Doubango] Digest for doubango@googlegroups.com - 2 updates in 1 topic
Yes I will ask my dev chaps tomorrow for you. On Sun, 26 Mar 2023, 15:26 , <doubango@googlegroups.
Mar 26
Craig Marston
2
Feb 18
getPlugin().createPeerConnection is not a function
We had to abandon SIPML5, but we have it all working well now as a result. On Friday, 2 September
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getPlugin().createPeerConnection is not a function
We had to abandon SIPML5, but we have it all working well now as a result. On Friday, 2 September
Feb 18
Ramin Zamanighiri
,
Virendra Bhati
2
Feb 17
webRTC and sipML5
403 means user not found, as asterisk is not able to handle this. so check password else SIP
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webRTC and sipML5
403 means user not found, as asterisk is not able to handle this. so check password else SIP
Feb 17
Pooja Yadav
12/23/22
boghe IMS with linux OS?
Hi, Can anyone confirm that can we boghe IMS client with Linux Operating System. Thanks, Pooja Yadav
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boghe IMS with linux OS?
Hi, Can anyone confirm that can we boghe IMS client with Linux Operating System. Thanks, Pooja Yadav
12/23/22
Rebecca Almeida
10/27/22
How to save the audio of calls
There is a method to save the links I would like to make a log for eventual queries if necessary.
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How to save the audio of calls
There is a method to save the links I would like to make a log for eventual queries if necessary.
10/27/22
Tom @ Orto
10/3/22
ANPR mobile, per application?
Hi, Does anyone know if there are any plans for a Per Application license for the ANPR mobile SDK?
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ANPR mobile, per application?
Hi, Does anyone know if there are any plans for a Per Application license for the ANPR mobile SDK?
10/3/22
MagnusT
8/15/22
Build failure
Hi, I fail at build... Guess something obvious/basic. Any ideas? Build command failed. Error while
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Build failure
Hi, I fail at build... Guess something obvious/basic. Any ideas? Build command failed. Error while
8/15/22
Abdul Rauf
,
Komil Gulboev
5
7/27/22
Building a softphone using SIPML5 with Grandstream Asterisk PBX
Sure, will do that. Unable to add this number of Whatsapp, we can meeting on Monday, just email in
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Building a softphone using SIPML5 with Grandstream Asterisk PBX
Sure, will do that. Unable to add this number of Whatsapp, we can meeting on Monday, just email in
7/27/22
Javier Delgado
7/12/22
WebRTC SIP Call: init audio call and receive both audio and video
I'm working on a web app which uses SIP video calls. To support these calls over WebRTC I'm
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WebRTC SIP Call: init audio call and receive both audio and video
I'm working on a web app which uses SIP video calls. To support these calls over WebRTC I'm
7/12/22
Diane ntchadji
2/24/22
Long latency time between SIPML5 client and Asterisk server
Hello, I am working on a communication system between a client (SIPML5) and an Asterisk server. I
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Long latency time between SIPML5 client and Asterisk server
Hello, I am working on a communication system between a client (SIPML5) and an Asterisk server. I
2/24/22
Deepak Kumar
12/16/21
SDP answer is not sent from Chrome Beta 97
Hi All, I am facing an issue on Chrome 97 Beta Version(97.0.4692.56) where my call is not being
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SDP answer is not sent from Chrome Beta 97
Hi All, I am facing an issue on Chrome 97 Beta Version(97.0.4692.56) where my call is not being
12/16/21
Jerry Geis
11/8/21
Failed to Authenticate
I am getting this from my asterisk side: [Nov 8 16:33:23] NOTICE[18067][C-00000001] chan_sip.c:
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Failed to Authenticate
I am getting this from my asterisk side: [Nov 8 16:33:23] NOTICE[18067][C-00000001] chan_sip.c:
11/8/21
Jayesh Purohit
10/25/21
Sipml5 Disable Camera Permission
Hello good morning everyone, I want to disable camera permission how to disable can any one help? I
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Sipml5 Disable Camera Permission
Hello good morning everyone, I want to disable camera permission how to disable can any one help? I
10/25/21
AD Rehman
9/1/21
sipMI on Mac safari
Hello friends. all working good on chrome exactly as expected, whereas on safari browser sipML is
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sipMI on Mac safari
Hello friends. all working good on chrome exactly as expected, whereas on safari browser sipML is
9/1/21
AD Rehman
8/21/21
sipML ws Connection timeout
Hi all, help is required to define WS connection timeout. actually, sometimes it is taking too long
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sipML ws Connection timeout
Hi all, help is required to define WS connection timeout. actually, sometimes it is taking too long
8/21/21
kenn
8/18/21
res_pjsip_session.c:937 handle_incoming_sdp: 888: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
Am currently using a digital ocean marketplace instance of FreePBX +asteris, after setting up
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res_pjsip_session.c:937 handle_incoming_sdp: 888: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
Am currently using a digital ocean marketplace instance of FreePBX +asteris, after setting up
8/18/21
Virendra Bhati
2
7/30/21
Webrtc Socket disconnect automatically after call end in Asterisk Queue
After a lot of code review issue found at my custom JS of Call handling. Issue fixed.. It's was
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Webrtc Socket disconnect automatically after call end in Asterisk Queue
After a lot of code review issue found at my custom JS of Call handling. Issue fixed.. It's was
7/30/21
G.Lake
, …
Nandrianina RAFANOMEZATSOA
5
7/28/21
sipml5: Issue with SIP registration
Hi I am facing the same issue as well. Could someone assist me please Thank you in advance. Kind
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sipml5: Issue with SIP registration
Hi I am facing the same issue as well. Could someone assist me please Thank you in advance. Kind
7/28/21
Saravanamurthy K
,
Mamadou DIOP
3
7/22/21
RTSP Stream Delay
Hi, Can help me how to implement rtsp stream in c++ Thanks K.Saravanamurthy. 9894802877 On Thu, Jul
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RTSP Stream Delay
Hi, Can help me how to implement rtsp stream in c++ Thanks K.Saravanamurthy. 9894802877 On Thu, Jul
7/22/21
kumar gajj
7/22/21
Sending audio stream like pre-recorded greeting message over sip channel
Hi, I want to send an audio stream (read from a local wav file) from one sip client to other peer
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Sending audio stream like pre-recorded greeting message over sip channel
Hi, I want to send an audio stream (read from a local wav file) from one sip client to other peer
7/22/21
Nandrianina RAFANOMEZATSOA
,
AD Rehman
2
7/9/21
Test SIPML5 on local
in case of secure you need to write wss://ScoketServerIP:Port/ws with asterisk server usually port is
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Test SIPML5 on local
in case of secure you need to write wss://ScoketServerIP:Port/ws with asterisk server usually port is
7/9/21
Marcel Ackermann
6/21/21
Configuring sipML5
Hi there, First of all, thank you for developing this! I love to see that you managed to build a
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Configuring sipML5
Hi there, First of all, thank you for developing this! I love to see that you managed to build a
6/21/21
Doe Mortu
6/17/21
getPlugin(...).createPeerConnection is not a function
Hi all, why do I get the below error when I host the application? SIPml5.js:1281 TypeError: getPlugin
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getPlugin(...).createPeerConnection is not a function
Hi all, why do I get the below error when I host the application? SIPml5.js:1281 TypeError: getPlugin
6/17/21
Spencer Scott
6/4/21
[Help] sipml5 demo wont register
Hello Friends, I am trying to build a webRTC softphone in our chrome extension but I can not seem to
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[Help] sipml5 demo wont register
Hello Friends, I am trying to build a webRTC softphone in our chrome extension but I can not seem to
6/4/21
Dudi Daabul
5/26/21
Hi
I'm trying to connect to a given SIP server, by using a given SIP account, by using your client,
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Hi
I'm trying to connect to a given SIP server, by using a given SIP account, by using your client,
5/26/21
silencioseu
,
shrey shah
2
4/15/21
sipML.js SIPml.Session.Call.call() returns -3.
Having same issue in safari browser, gettiing -3 when calling, have you found any solution for this?
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sipML.js SIPml.Session.Call.call() returns -3.
Having same issue in safari browser, gettiing -3 when calling, have you found any solution for this?
4/15/21
shrey shah
4/15/21
Working with Angular 10
Hi, I want to implement SIPml5 in my angular 10 project. Are there any tutorials or sample projects
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Working with Angular 10
Hi, I want to implement SIPml5 in my angular 10 project. Are there any tutorials or sample projects
4/15/21
Ad Rehman
4/4/21
tsk_utils.js?svn=252:128 Failed to parse message:
Hello all, right after i process login, it seems like i am connected to my asterisk... but in
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tsk_utils.js?svn=252:128 Failed to parse message:
Hello all, right after i process login, it seems like i am connected to my asterisk... but in
4/4/21