An Error occurred while connecting to the websocket.

[chornyitaras]

An Error occurred while connecting to the websocket.

[chornyitaras]

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

[symbiose1]

Hi, no, i use a self-sign.

Is it the problem ?

De : Taras Chornyi [mailto:[email protected]]
Envoyé : 16 novembre 2016 14:02
À : chornyitaras/PBXWebPhone [email protected]
Cc : symbiose1 [email protected]; Mention [email protected]
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]

@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]

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

I will try with letsencrypt.

De : Taras Chornyi [mailto:[email protected]]
Envoyé : 16 novembre 2016 14:24
À : chornyitaras/PBXWebPhone [email protected]
Cc : symbiose1 [email protected]; Mention [email protected]
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]

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]

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]

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

[symbiose1]

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]

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" <[email protected]mailto:[email protected]> 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]

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 [email protected] a écrit :

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" <[email protected]mailto:[email protected]> 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]

[chornyitaras]

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" <[email protected]mailto:[email protected]> wrote:

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

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[symbiose1]

This site isnt accessible, Error Connection time out

[symbiose1]

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]

Use command " yast firewall " to configure suse firewall

??????????? Outlook https://aka.ms/ghei36 ???https://aka.ms/ghei36 Androidhttps://aka.ms/ghei36

???: 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]

Now webphone should work

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On Tue, Nov 22, 2016 at 11:53 PM +0200, "symbiose1" <[email protected]mailto:[email protected]> 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]

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]

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

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

On Wed, Nov 23, 2016 at 12:03 AM +0200, "symbiose1" <[email protected]mailto:[email protected]> 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]

no ...

[chornyitaras]

Please check the sip peer status in asterisk.

asterisk -r
sip show peers

On Wed, Nov 23, 2016 at 12:18 AM +0200, "symbiose1" <[email protected]mailto:[email protected]> wrote:

no ...

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[symbiose1]

the webphone is gs102

[symbiose1]

[chornyitaras]

Is your server behind NAT?

On Wed, Nov 23, 2016 at 12:23 AM +0200, "symbiose1" <[email protected]mailto:[email protected]> wrote:

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

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[chornyitaras]

Don't forget to start firewall. :)

[symbiose1]

Yes, it is behind nat.

Le 22 nov. 2016 à 17:48, Taras Chornyi [email protected] a écrit :

Don't forget to start firewall. :)

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[symbiose1]

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]

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]

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

[chornyitaras]

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

[symbiose1]

You are now added, thank you very much ! De : Taras Chornyi [mailto:[email protected]] Envoyé : 24 novembre 2016 10:11 À : chornyitaras/PBXWebPhone <[email protected]> Cc : symbiose1 <[email protected]>; Mention <[email protected]> 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]

sorry wrong skype name(( correct one: tarasukcho

[zues75]

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

[chornyitaras]

port 8089 was closed, and incorrect certificate configuration

[zues75]

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]

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]

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]

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]

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]

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]

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]

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]

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]

And i am still receiving this error also.

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

[chornyitaras]

Those errors are expected

[zues75]

Again Thank you for all your help.