chornyitaras/pbxwebphone

WebRTC based webphone for Vicidial

pbxwebphone's Issues

Make autoanswer optional

Add "Answer" and "Hangup" buttons when auto answer option is disabled in Vicidial ADMIN

Webphone works from campaign but not Reports

I have the Webphone working great from within campaigns. When I access the Realtime Report (/vicidial/realtime_report.php#) there's a link to Webphone on the top right which opens an iframe with an error:
Got wrong web socker url. Please check your settings

Any chance it can work from the reports screen to realtime monitor.

Thanks.

An Error occurred while connecting to the websocket.

Hi, i have follow your guide and i have an issue..
The phone open but i have this error: An Error occurred while connecting to the websocket.
I have properly make the config in asterisk. The port is open..
Do you have an idea ? can you help me with that ?
thanks

Audio issue- Disconnecting call for lack of RTP activity in 61 seconds

I have followed your steps and configured this module and it seems to be working fine except audio issue.

My system information is as below:

Admin Vicidial Information
VERSION: 2.14-611a
BUILD: 170425-1353
© 2017 ViciDial Group

Agent Vicidial Information
VERSION: 2.14-520c
BUILD: 170416-1640

Asterisk Version
Asterisk 11.25.1-vici

Log for rtp is as below:

[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033387, ts 480640, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033388, ts 480800, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033389, ts 480960, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033390, ts 481120, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033391, ts 481280, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033392, ts 481440, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033393, ts 481600, len 000160)

When agent is logging in, and press call Agent webphone:

[Apr 28 06:57:14] == Using SIP RTP CoS mark 5
[Apr 28 06:57:15] > Channel SIP/5555-00000002 was answered
[Apr 28 06:57:15] -- Executing [8600052@default:1] MeetMe("SIP/5555-00000002", "8600052,F") in new stack
[Apr 28 06:57:15] == Parsing '/etc/asterisk/meetme.conf': Found
[Apr 28 06:57:15] == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[Apr 28 06:57:15] -- Created MeetMe conference 1023 for conference '8600052'
[Apr 28 06:57:15] -- <SIP/5555-00000002> Playing 'conf-onlyperson.gsm' (language 'en')

[Apr 28 06:57:46] NOTICE[7292]: chan_sip.c:29370 check_rtp_timeout: Disconnecting call 'SIP/5555-00000002' for lack of RTP activity in 61 seconds
[Apr 28 06:57:46] -- Hungup 'DAHDI/pseudo-1305219282'

[Apr 28 06:57:46] == Spawn extension (default, 8600052, 1) exited non-zero on 'SIP/5555-00000002'
[Apr 28 06:57:46] -- Executing [h@default:1] AGI("SIP/5555-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----44---------------") in new stack

It looks like issue with RTP, we have already opened rtp port range from firewall.

Can you please suggest me how to make it working?
Is there anything wrong in configuration?

Please let me know.

Obtain LetsEncrypt Certificates

WebPhone hanging up

I was testing this webphone for vicidial and I encountered this

[Jun 17 14:53:38] ERROR[25141][C-00000001]: res_rtp_asterisk.c:2131 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f3508010da8' due to reason 'missing tmp ecdh key', terminating
[Jun 17 14:53:38] WARNING[25141][C-00000001]: res_rtp_asterisk.c:4234 ast_rtp_read: RTP Read error: Unspecified. Hanging up.

any clue on how to resolve this hurdle?

Hear ringing but not auto answering

I'm using latest version of vicidial.

I get this on asterisk CLI:

[Nov 1 16:11:11] NOTICE[11250]: chan_sip.c:24639 handle_response_peerpoke: Peer '2727' is now Reachable. (85ms / 2000ms)
[Nov 1 16:11:15] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 16:11:15] == DTLS ECDH initialized (automatic), faster PFS enabled
[Nov 1 16:11:15] == Using SIP RTP CoS mark 5
[Nov 1 16:11:15] -- Called 2727
[Nov 1 16:11:15] -- SIP/2727-00000008 is ringing

It looks like asterisk is not detecting the answer.

webrtc webphone silent suddenly

Hi @chornyitaras,
webrtc webphone silent suddenly no voice. peerconnection.close into browser consloe log, we restart asterisk, restart vicidial service nothing work, no error into asterisk CLI call connected but no voice, its only work when server restart. any idea its really top urgent,

thanks.

After new update in chrome and firefox, webphone stopped working

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

MultiServer

Want to say that this is a great addition and saves time in setting up new agents and phones. My question is, in a multi-server setup we create the phones but are currently creating a template for each asterisk server. After phone creation we need to assign template to each phone, if we create 3 phones to load balance, we have to change template in each phone. Is there an easier way to accomplish this?

Unable to reach webphone

Hello,

I am facing the below issue which logging in the campaign, Asterisk shows this. cannot request the channel.

channel.c:5690 __ast_request_and_dial: Unable to request channel SIP/gs102

WebRTC No Audio on Internet but works on Local Network

Hi,

This is more like a help request as I am not sure if it is a bug.

I have configured WebRTC to work successfully on the local network. It works fine without any issue. But if I login from internet, the WebRTC audio is not coming through. I can place calls and receive calls but there is no audio.

I am attaching two logs, internal and external.

Internal.txt is the browser webrtc log from internal network where audio is coming through.

External.txt is the browser webrtc log from external network where audio is not working.

I have attached the sip/http/phone configuration files as well as the asterisk sip debug logs.

internal.txt
external.txt
sip-vicidial.conf.txt
sip.conf.txt
http.conf.txt
asterisk_sip_external.txt
asterisk_sip_internal.txt

I am unable to find out where in the configuration have I done it wrong. I have been tracing the net for resolution.

What bothers me is the following log in asterisk_sip_external.txt file:

[Dec  5 12:05:39] Really destroying SIP dialog 'mcq3e4qigdn0ec0l3m46lh' Method: REGISTER
[Dec  5 12:05:39] ERROR[2227]: tcptls.c:447 tcptls_stream_close: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[Dec  5 12:05:39]   == WebSocket connection from '122.170.155.41:59019' forcefully closed due to fatal write error

If you check the config.png, you will see the layout of what I have. We can get on a quick call as well to see the scenario live.

Mute option

Add an option for mute when the call is setup

Add support for Vicidial options

vicidial posts several options to webphone frame.
options:
INITIAL_LOAD--DIALPAD_OFF_TOGGLE--AUTOANSWER_Y--DIALBOX_Y--MUTE_Y--VOLUME_Y

Firefox Test

Hi, I have been testing with firefox (different versions and different OS), it rings the first time and show it as answered, but there is no sound, a few seconds after it returns a message "No one is in your session" but webphone is still in answer state. This doesn't happen with chrome, in chrome everything works fine. Have you tested in a specific firefox version?

An Error occurred while connecting to the websocket

Hi Taras, As mentioned previoulsy all is running fine when suddenly issue starts coming and below message starts appearing
"An Error occurred while connecting to the websocket."
Then we have to restart asterisk and it gets fixed...

Any help would be highly appreciated. Thanks.

Add installation instructions.

Provide install steps

SRTCP unprotect failed because of authentication failure

hi @chornyitaras
i am getting this error when webphone connected, webphone working fine but getting this message on asterisk.
SRTCP unprotect failed because of authentication failure

Volume control

Is there any way to adjust the volume? It doesn't need to be on the UI, if a file can be modified to rise the volume it would be great. Thanks in advance.

No One is your session

Hi,

As you included image that it should show Ready. Its showing ready but when agent to login it should ring and stay connected but its saying no one is your session.

Is working for anyone in vicidial?

Dialpad!

Hello,
After long research I landed on PBXWebPhone, it worked without any tweaking except for dialpad.
Am I missing something here? I could not seem to find a way to call out.
Also the auto-answer ! why should it be always enabled ?
This project is great and very simple, it lacks the following

1. Dialpad unless it is there and I cannot find
2. Auto answer not always on
3. Ring tones.

Hope to see some collaboration on this
Thank you

Websocket connection error

I have followed your guide to install webphone on the server. But getting the below error in Chrome dev console.

WebSocket connection to 'wss://domain.com:8089/ws' failed: Error in connection establishment: net::ERR_CONNECTION_REFUSED

All the certificates are properly installed.

Also added firewall exceptions.

Webphone Issue han_sip.c: 4258 _sip reliable_xmit: Serious Network Trouble; _sip xmit

hi @chornyitaras

[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

i am getting this error any idea?

asterisk 11.25

Real Time Monitoring error

At first was error not finding index.php, so copied PBXWebPhone folder to vicidial folder and now Real Time screen finds the index.php. However, have websocket connection error....yet agent phones work on this same server.