An Error occurred while connecting to the websocket. #7

symbiose1 · 2016-11-15T22:22:05Z

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

chornyitaras · 2016-11-16T19:01:56Z

hi @symbiose1
Are you using valid ssl sertificate?
You can also try to check your system configuration using https://tryit.jssip.net/

symbiose1 · 2016-11-16T19:02:58Z

Hi, no, i use a self-sign.

Is it the problem ?

De : Taras Chornyi [mailto:notifications@github.com]
Envoyé : 16 novembre 2016 14:02
À : chornyitaras/PBXWebPhone PBXWebPhone@noreply.github.com
Cc : symbiose1 c.crevier@symbioseccc.com; Mention mention@noreply.github.com
Objet : Re: [chornyitaras/PBXWebPhone] An Error occurred while connecting to the websocket. (#7)

hi @symbiose1 https://github.com/symbiose1
Are you using valid ssl sertificate?
You can also try to check your system configuration using https://tryit.jssip.net/

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chornyitaras · 2016-11-16T19:24:05Z

@symbiose1
It might be.
Try to open https://:8089/ws in your browser and confirm securitu exeptions.
PS
why not to get free SSL cert form letsencrypt ?

symbiose1 · 2016-11-16T19:47:24Z

When i try to access at https://:8089/ws https://:8089/ws its doesnt work.

I will try with letsencrypt.

De : Taras Chornyi [mailto:notifications@github.com]
Envoyé : 16 novembre 2016 14:24
À : chornyitaras/PBXWebPhone PBXWebPhone@noreply.github.com
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Objet : Re: [chornyitaras/PBXWebPhone] An Error occurred while connecting to the websocket. (#7)

@symbiose1 https://github.com/symbiose1

It might be.
Try to open https://:8089/ws https://:8089/ws in your browser and confirm securitu exeptions.
PS
why not to get free SSL cert form letsencrypt ?

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chornyitaras · 2016-11-16T20:24:53Z

is port 8089 open? when accessing https://dial1.sym-it.ca:8089/ws using browser you should get something like this:

can you execute following command on your server?

netstat -tulpan | grep asterisk

this will show what ports asterisk is currently listening

symbiose1 · 2016-11-18T15:58:58Z

Free community-based ViciDial Support is available
at http://www.vicidial.org/VICIDIALforum

ViciBox v.7.0.3-160505
accesdistant:~ # netstat -tulpan | grep asterisk
tcp 0 0 0.0.0.0:5038 0.0.0.0:* LISTEN 1642/asterisk
tcp 0 0 127.0.0.1:5038 127.0.0.1:9186 ESTABLISHED 1642/asterisk
tcp 0 0 127.0.0.1:5038 127.0.0.1:9185 ESTABLISHED 1642/asterisk
udp 0 0 0.0.0.0:5060 0.0.0.0:* 1642/asterisk
udp 0 0 0.0.0.0:4520 0.0.0.0:* 1642/asterisk
udp 0 0 0.0.0.0:4569 0.0.0.0:* 1642/asterisk

i dont see the port 8089

chornyitaras · 2016-11-18T18:52:47Z

Please check your asterisk configuration
https://github.com/chornyitaras/PBXWebPhone/wiki/Asterisk-configuration

symbiose1 · 2016-11-22T21:29:43Z

hi,
i have just install letsencrypt,
now i see the port 8089 but when i tru to access to https://:8089/ws isnt working.
The port is properly open in the firewall, i have test with the local ip also and i have nothing ...

chornyitaras · 2016-11-22T21:38:58Z

What do you mean by not working? Can you please share a screenshot.

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On Tue, Nov 22, 2016 at 11:29 PM +0200, "symbiose1" <notifications@github.com mailto:notifications@github.com> wrote:

hi,
i have just install letsencrypt,
now i see the port 8089 but when i tru to access to https://dial1.sym-it.ca:8089/ws isnt working.
The port is properly open in the firewall, i have test with the local ip also and i have nothing ...
[image]https://cloud.githubusercontent.com/assets/23487620/20542722/daac82be-b0d0-11e6-961f-104217a7d357.png

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symbiose1 · 2016-11-22T21:42:07Z

Hi Taras,
when i try to connect onto the campaign,
in the web phone, i have a message error occurred web socket.

Le 22 nov. 2016 à 16:39, Taras Chornyi notifications@github.com a écrit :

What do you mean by not working? Can you please share a screenshot.

Завантажити Outlook для Androidhttps://aka.ms/ghei36

On Tue, Nov 22, 2016 at 11:29 PM +0200, "symbiose1" <notifications@github.com mailto:notifications@github.com> wrote:

hi,
i have just install letsencrypt,
now i see the port 8089 but when i tru to access to https://dial1.sym-it.ca:8089/ws isnt working.
The port is properly open in the firewall, i have test with the local ip also and i have nothing ...
[image]https://cloud.githubusercontent.com/assets/23487620/20542722/daac82be-b0d0-11e6-961f-104217a7d357.png

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symbiose1 · 2016-11-22T21:45:00Z

chornyitaras · 2016-11-22T21:47:22Z

What do you see when accessing https://server_ip:8089/ws ?

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On Tue, Nov 22, 2016 at 11:45 PM +0200, "symbiose1" <notifications@github.com mailto:notifications@github.com> wrote:

[image]https://cloud.githubusercontent.com/assets/23487620/20543187/018c420a-b0d3-11e6-8ff0-aa1234fd2bae.png

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symbiose1 · 2016-11-22T21:49:13Z

This site isnt accessible, Error Connection time out

symbiose1 · 2016-11-22T21:53:36Z

Oh, i have find. now
i have enter the port on iptables but in opensuse is a diferent firewall..
i just stop rcSuSEfirewall2.
now i see the page

chornyitaras · 2016-11-22T21:55:25Z

Use command " yast firewall " to configure suse firewall

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???: symbiose1
?????????: ????????, 22 ????????? 23:53
????: Re: [chornyitaras/PBXWebPhone] An Error occurred while connecting to the websocket. (#7)
????: chornyitaras/PBXWebPhone
?????: Taras Chornyi, Comment

Oh, i have find. now
i have enter the port on iptables but in opensuse is a diferent firewall..
i just stop rcSuSEfirewall2.
now i see the page

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chornyitaras · 2016-11-22T21:56:06Z

Now webphone should work

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On Tue, Nov 22, 2016 at 11:53 PM +0200, "symbiose1" <notifications@github.com mailto:notifications@github.com> wrote:

Oh, i have find. now
i have enter the port on iptables but in opensuse is a diferent firewall..
i just stop rcSuSEfirewall2.
now i see the page
[image]https://cloud.githubusercontent.com/assets/23487620/20543534/34c1fd76-b0d4-11e6-8727-7c6088f47ec9.png

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symbiose1 · 2016-11-22T22:03:53Z

Now, the phone is ready but i dont receive the connect call, "you are now the only one at the conference"
and after 10 second, my session is automaticly disconnected.

i have try to make a manual dial and i have no sound...

chornyitaras · 2016-11-22T22:07:25Z

So you don't receive incoming call after pressing "Call Agent Webphone"?

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On Wed, Nov 23, 2016 at 12:03 AM +0200, "symbiose1" <notifications@github.com mailto:notifications@github.com> wrote:

Now, the phone is ready but i dont receive the connect call, "you are now the only one at the conference"
and after 10 second, my session is automaticly disconnected.

i have try to make a manual dial and i have no sound...

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symbiose1 · 2016-11-22T22:17:38Z

no ...

chornyitaras · 2016-11-22T22:20:12Z

Please check the sip peer status in asterisk.

asterisk -r
sip show peers

On Wed, Nov 23, 2016 at 12:18 AM +0200, "symbiose1" <notifications@github.com mailto:notifications@github.com> wrote:

no ...

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symbiose1 · 2016-11-22T22:22:14Z

the webphone is gs102

symbiose1 · 2016-11-22T22:23:41Z

chornyitaras · 2016-11-22T22:34:10Z

Is your server behind NAT?

On Wed, Nov 23, 2016 at 12:23 AM +0200, "symbiose1" <notifications@github.com mailto:notifications@github.com> wrote:

[image]https://cloud.githubusercontent.com/assets/23487620/20544496/6919a6b0-b0d8-11e6-9201-26cc38b7249f.png

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chornyitaras · 2016-11-22T22:48:16Z

Don't forget to start firewall. :)

symbiose1 · 2016-11-22T23:19:52Z

Yes, it is behind nat.

Le 22 nov. 2016 à 17:48, Taras Chornyi notifications@github.com a écrit :

Don't forget to start firewall. :)

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symbiose1 · 2016-11-23T14:36:48Z

Hi, i have re-check my settings into the firewall and i think all is good,,,,
i see the phone connected but i dont receive the incomming call.
the webphone is gs102

chornyitaras · 2016-11-23T20:17:17Z

plase enable sip debug
asterisk -r
sip set debug peer gs102

Also I advise you to check your Vicidial installation using softphone(Linphon, Xlite ets).
login as agent but don't press "Call Agent Webphone"
Then crete one more phone using Vicidial Admin, register it in softphone and call gs102 fron softphone

symbiose1 · 2016-11-24T14:59:11Z

Hi Taras,
i have make the sip debud and i see this

chornyitaras · 2016-11-24T15:11:25Z

looks like something wrong with asterisk ssl configuration.
please add me @ Skype tarasucho. So we can have live debug session

symbiose1 · 2016-11-24T15:25:53Z

You are now added, thank you very much ! De : Taras Chornyi [mailto:notifications@github.com] Envoyé : 24 novembre 2016 10:11 À : chornyitaras/PBXWebPhone <PBXWebPhone@noreply.github.com> Cc : symbiose1 <c.crevier@symbioseccc.com>; Mention <mention@noreply.github.com> Objet : Re: [chornyitaras/PBXWebPhone] An Error occurred while connecting to the websocket. (#7) looks like something wrong with asterisk ssl configuration. please add me @ Skype tarasucho. So we can have live debug session — You are receiving this because you were mentioned. Reply to this email directly, view it on GitHub <#7 (comment)> , or mute the thread <https://github.com/notifications/unsubscribe-auth/AWZkhN_tIugzjxqj5j4zYdVH2FUNCDKdks5rBakdgaJpZM4KzGMN> . <https://github.com/notifications/beacon/AWZkhAsyEupk2IGovflgFbQoNrsV195gks5rBakdgaJpZM4KzGMN.gif>

chornyitaras · 2016-11-24T15:29:09Z

sorry wrong skype name(( correct one: tarasukcho

zues75 · 2017-02-25T05:56:30Z

I see that this issue has been resolved and i am having the same issues. How was the asterisk ssl issue resolved?

chornyitaras · 2017-02-25T13:10:08Z

port 8089 was closed, and incorrect certificate configuration

zues75 · 2017-02-25T16:10:54Z

Thanks, I think my ports are all ok. Firewall is on but disabled. I am behind a NAT and that has all the ports configured. I did make a letsencrypt cert and used that for both the apache2 and http.conf settings, but if i am seeing this right the http.conf may need to use asterisk ssl certificate is that correct?

chornyitaras · 2017-02-25T21:28:50Z

You need to use same certificate for Apache and asterisk. Make sure that asterisk is listening 8090 port.

netstat -tulpan | grep 8089

Please execute this command on your server.

Then please open in browser https://asterisk_server_domain_name:8089

zues75 · 2017-02-25T22:35:47Z

Thanks for getting back to me i tried what you said and the results did come up with port 8089 but it didn't look right. I also went to the website and it didn't come up with a verified cert. I have this on a test vm that i am using to verify the installation and troubleshooting of this before i bring it out to a production machine. At this time i reverted the settings back to before I put anything into the machine and will begin again. Shouldn't take me to long to recreate the dialer with the setting back in it. and i will test them out without any addition settings and see what i get at that time.

zues75 · 2017-02-25T23:22:36Z

i executed the netstat -tulpan | grep 8089 and came back with this.

tcp 0 0 192.168.5.94:8089 0.0.0.0:* LISTEN 3410/a

OK i am now at this when going to https://asterisk_server_domain_name:8089 , and yes i did input my url name in place of asterisk_server_domain_name :)

Not Found

The requested URL was not found on this server.
Asterisk Server

chornyitaras · 2017-02-26T09:37:36Z

That's actually good results. This mean that you can connect to asterisk HTTP server.
When opening https://asterisk_domai_name:8089/ws can you see "Upgrade required'?.
If yes please check your Vicidial configuration. Make sure that "Web Socket URL" is set to https://asterisk_domai_name:8089/ws

zues75 · 2017-02-26T17:08:14Z

Great I'm on the right track. I read through your earlier post and have test out that https://asterisk_domai_name:8089/ws and i can see. sorry if image didn't come up first time on github chat.

As for the Web Socket URL : is that suppose to be wss://asterisk_domain_name:8089/ws or https://asterisk_domain_name:8089/ws ?

zues75 · 2017-02-26T17:11:32Z

At this time with wss://asterisk_domain_name:8089/ws i am running into two issues.

1st on is :
There is no " You are to only one in the conference" message that plays when the phone is first picked up.

2nd issue is

Disconnecting call for lack of RTP activity in 61 seconds

chornyitaras · 2017-02-26T17:53:00Z

looks like NAT related issue.
please add stun severer config(file /etc/asterisk/rtp.conf)
icesupport=yes
stunaddr=stun.l.google.com:19302
under [general] section

zues75 · 2017-02-26T18:18:48Z

That worked!!!! Thank you. It looks like i will need to look further into the UDP port. hmm strange.

I am getting a strange error that i haven't seen before after i put the stunaddr in there. This action happens when i make a call. Do you have any ideas what this could be from ?

res_srtp.c:415 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10

zues75 · 2017-02-26T18:23:34Z

And i am still receiving this error also.

tcptls.c:397 tcptls_stream_close: SSL_shutdown() failed: 5

chornyitaras · 2017-02-26T18:37:48Z

Those errors are expected

zues75 · 2017-02-26T18:54:54Z

Again Thank you for all your help.

chornyitaras added the question label Nov 16, 2016

chornyitaras closed this as completed Nov 24, 2016

chornyitaras mentioned this issue Dec 5, 2017

Websocket connection error #16

Closed

An Error occurred while connecting to the websocket. #7

An Error occurred while connecting to the websocket. #7

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