Closed jitin27 closed 5 years ago
error : RTCPeerConnection.getLocalStreams/getRemoteStreams are deprecated. Use RTCPeerConnection.getSenders/getReceivers instead.
Hi, what asterisk version you are using?
asterisk 11.22.0-vici earlier it was working fine after new update on chrome and firefox, it started giving this issue.
On Thu, Nov 1, 2018 at 12:59 PM Taras Chornyi notifications@github.com wrote:
Hi, what asterisk version you are using?
— You are receiving this because you authored the thread. Reply to this email directly, view it on GitHub https://github.com/chornyitaras/PBXWebPhone/issues/19#issuecomment-434956364, or mute the thread https://github.com/notifications/unsubscribe-auth/Aqk6ayb2U5qmYqw05H67y65xqASY8Ylvks5uqqLLgaJpZM4YEMzU .
-- Thanks and Regards,
Jitin Goel (VoIP Engineer) Interbit Solutions Pvt. Ltd.
please pull the latest master and check
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.
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
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
Hi. Root cause for this issue is identified. Will release fix soon
ok....Thank you Please let me know when the issue gets fixed.
Hello,
Even i am facing the same issue, Can we know when will the next release be updated ?
New version will be available till the end of this week
Hi,
any luck on the new version ?
Issue should be fixed in latest master
@amitiyer can you please confirm that issue is fixed for you ?
yes.....issue is fixed for me
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.
@luqman27180 can you please provide an asterisk output?
@luqman27180 can you please provide an asterisk output?
@chornyitaras please check this. please asterisk output.
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
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
we are using vicibox and they are using 11, we are not using standard. asterisk only.
Latest vicibox supports asterisk 13
Latest vicibox supports asterisk 13
hi, @chornyitaras thanks let me installed new one and try , in case of issue will back 👍
Latest vicibox supports asterisk 13
hi @chornyitaras
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.
hi @chornyitaras
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
Firefox 63.0
When we click on "Call Agent Webphone", it comes terminated on webphone and the call is not getting answered. It was working fine in the previous version. Kindly check with the latest firefox and google chrome issue and fix the same.
Thanks in advnace