Comments (24)
error : RTCPeerConnection.getLocalStreams/getRemoteStreams are deprecated. Use RTCPeerConnection.getSenders/getReceivers instead.
from pbxwebphone.
Hi, what asterisk version you are using?
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from pbxwebphone.
please pull the latest master and check
from pbxwebphone.
DOMException: "Local descriptions must have a=mid attributes." sip-0.11.3.min.js:1:9026
e.exports/t.prototype.print
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:9026
e.exports/</t.prototype[r]
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:9216
e.exports/</i.prototype[r]
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:9115
value/<
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:150850
sendRequest
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:63487
setACKTimer/this.timers.ackTimer<
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:72451
Now it rings, and then comes terminated. Call doesn't get answered.
from pbxwebphone.
It is working on chrome but giving error in firefox.
In firefox , it rings and then gets terminated.
error :
DOMException: "Local descriptions must have a=mid attributes." sip-0.11.3.min.js:1:9026
e.exports/t.prototype.print
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:9026
e.exports/</t.prototype[r]
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:9216
e.exports/</i.prototype[r]
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:9115
value/<
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:150850
SEND Request :
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:63487
setACKTimer/this.timers.ackTimer<
https://domain/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:72451
from pbxwebphone.
Hello Sir
Any update on the issue
I am still facing the same issue, it is working on chrome but giving the following error on firefox:
DOMException: "Local descriptions must have a=mid attributes." sip-0.11.3.min.js:1:9026
e.exports/t.prototype.print
https://game888888888.com/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:9026
e.exports/</t.prototype[r]
https://game888888888.com/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:9216
e.exports/</i.prototype[r]
https://game888888888.com/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:9115
value/<
https://game888888888.com/agc/PBXWebPhone/scripts/sip-0.11.3.min.js:1:150850
Please do the needful
from pbxwebphone.
Hi. Root cause for this issue is identified. Will release fix soon
from pbxwebphone.
ok....Thank you
Please let me know when the issue gets fixed.
from pbxwebphone.
Hello,
Even i am facing the same issue, Can we know when will the next release be updated ?
from pbxwebphone.
New version will be available till the end of this week
from pbxwebphone.
Hi,
any luck on the new version ?
from pbxwebphone.
Issue should be fixed in latest master
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@amitiyer can you please confirm that issue is fixed for you ?
from pbxwebphone.
yes.....issue is fixed for me
from pbxwebphone.
hi, i am still get issue on Firefox. there is my sip trace details, please can you help me to fix this issue on chrome its working fine, i am using latest code.
from pbxwebphone.
@luqman27180 can you please provide an asterisk output?
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@luqman27180 can you please provide an asterisk output?
@chornyitaras please check this. please asterisk output.
from pbxwebphone.
ERROR[8201][C-00000038]: res_rtp_asterisk.c:2166 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f01280109a8' due to reason 'no shared cipher', terminating
This is known issue with asterisk 11 and Firefox
https://issues.asterisk.org/jira/plugins/servlet/mobile#issue/ASTERISK-25265
Please use asterisk 13
from pbxwebphone.
ERROR[8201][C-00000038]: res_rtp_asterisk.c:2166 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f01280109a8' due to reason 'no shared cipher', terminating
This is known issue with asterisk 11 and Firefox
https://issues.asterisk.org/jira/plugins/servlet/mobile#issue/ASTERISK-25265Please use asterisk 13
we are using vicibox and they are using 11, we are not using standard. asterisk only.
from pbxwebphone.
Latest vicibox supports asterisk 13
from pbxwebphone.
Latest vicibox supports asterisk 13
hi, @chornyitaras thanks let me installed new one and try , in case of issue will back 👍
from pbxwebphone.
Latest vicibox supports asterisk 13
if this can help or sure we have to use . asterisk 13?
https://stackoverflow.com/questions/48909643/coturn-webrtc-issue-on-azure-vm-ubuntu-server-16-04-lts
please give me your feedbacks
thanks.
from pbxwebphone.
we are using your webtrc webphone with vicidial and having this issue, can you please give me your feedback. what possible reason, ?
[Jan 23 16:31:03] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:07] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:15] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:17] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:21] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:29] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:31] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:35] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:43] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:45] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:49] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:57] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:31:59] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:32:03] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:32:11] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jan 23 16:32:13] ERROR[16280] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
from pbxwebphone.
Related Issues (20)
- No One is your session HOT 1
- Audio issue- Disconnecting call for lack of RTP activity in 61 seconds HOT 1
- Dialpad! HOT 4
- MultiServer HOT 1
- WebPhone hanging up HOT 8
- Websocket connection error HOT 5
- Firefox Test HOT 2
- Obtain LetsEncrypt Certificates
- Hear ringing but not auto answering HOT 8
- WebRTC No Audio on Internet but works on Local Network
- Webphone Issue han_sip.c: 4258 _sip reliable_xmit: Serious Network Trouble; _sip xmit HOT 1
- An Error occurred while connecting to the websocket HOT 2
- webrtc webphone silent suddenly HOT 4
- Unable to reach webphone HOT 4
- Volume control
- SRTCP unprotect failed because of authentication failure HOT 1
- Real Time Monitoring error HOT 3
- An Error occurred while connecting to the websocket. HOT 45
- Webphone works from campaign but not Reports HOT 6
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