vicidial.org

I have Viciphone 2.0 working in Chrome and Firefox (at least on the 3 PCs I have tested so far). Would love some community feedback. If you are interested in testing it please do the following:

Change your WebRTC Template in Vicidial to this:

type=friend
host=dynamic
encryption=yes
icesupport=yes
directmedia=no
transport=wss
dtlsenable=yes
dtlsverify=no
dtlscertfile=/PATH/TO/YOUR/SSL/CERT
dtlsprivatekey=/PATH/TO/YOU/SSL/KEY
dtlssetup=actpass
rtcp_mux=yes

Then in System Setting change the Webphone URL to:

https://dev.viciphone.com/v2/viciphone.php

Everything all other configurations should be the same as version one. As I have said I have only tested this on 3 PCs so I recommend that you try this on a test dialer and not on a production system.

What are the changes/improvements compared to v1?

It uses the new SIP.js library which is very different. It also has the potential to support MS Edge.

Michael,

Could you please publish this new code under a new branch in Github? I'll love to check out the new code and see whats new.

ccabrera wrote:Michael,

Could you please publish this new code under a new branch in Github? I'll love to check out the new code and see whats new.

He'll be doing that AFTER he's finished with this round of testing. That's precisely the point of this post. He doesn't want to deal with 200 guys posting the same flaws to the code he's already working on. It's annoying to deal with that sort of thing, and he wants to see the activity in HIS server first so he can see what needs to be fixed before release.

Respect the coders methods, his work is excellent.

William, I believe that is the reason for which GitHub was created: Share code so that others may review, suggest, even improve.

He doesn't need to work on all the flaws by himself. We could help too, but for that we need to be able to join while the product is being built, not after.

As for the flaws he's already working on, that's completely understandable. But then again: GitHub has an issue tracker which would allow anyone to have a closer and more detailed follow up of whatever he's coding.

So what I'm saying is: there's some of us who want to contribute in developing, not just testing, and for that, we need to see code while it's being written.

Let us help.

Yes, but not during initial build-out. During initial buildout, proof of concept and prototype stages, it's common to keep it in-house and not something you argue about with a technician who will be publishing the final prototype when it's ready.

He likes to get to "I think it works" and request bug checking after it's fully operational. I'll never argue with that approach on a free project (or bug the guy who has the code prematurely: If he wanted to publish his GIT already he would have).

lol: The rough equivalent of hearing "are we there yet?" from the back seat. IMHO

I tested in chrome and it worked fine. Also tested in Edge but didn't get any audio and after 1 min the message there was no connection with my phone. Edge did ask me if my microphone could be used. Anything specific you wanted me to test?

Also, can you add different ringtones for the webphone?

I will be releasing the code on github once I am positive everything is working correctly. With regards to Edge, I am not bothering with finalizing that till they get their core switched over to Chromium. At which point I am 99% sure it will just work. As for the ringtones, I am happy to switch them, but we need an audio file that is is released under a compatible license which was hard to find in the first place.

mcargile wrote:I will be releasing the code on github once I am positive everything is working correctly. With regards to Edge, I am not bothering with finalizing that till they get their core switched over to Chromium. At which point I am 99% sure it will just work. As for the ringtones, I am happy to switch them, but we need an audio file that is is released under a compatible license which was hard to find in the first place.

just a place for the ringtone would be fine, even if it's always got the same filename in it. the ability to set the ringtone to a file in the audio store would be perfect.

Honestly do not see this being added any time soon unless someone paid for it. This is mainly because it would require a ton of work in the agent interface of Vicidial to pass the option to Viciphone and Matt is swamped with paid development.

In other news the Canary build of Edge based off Chromium was just released and it works flawlessly with Viciphone.

mcargile wrote:Honestly do not see this being added any time soon unless someone paid for it. This is mainly because it would require a ton of work in the agent interface of Vicidial to pass the option to Viciphone and Matt is swamped with paid development.

In other news the Canary build of Edge based off Chromium was just released and it works flawlessly with Viciphone.

ah, well. putting a note of where that file could be modified somewhere in the code for the lay-people wouldn't be bad either, if it's not too troublesome. Or here. Here is good, too.

williamconley wrote:

mcargile wrote:Honestly do not see this being added any time soon unless someone paid for it. This is mainly because it would require a ton of work in the agent interface of Vicidial to pass the option to Viciphone and Matt is swamped with paid development.

In other news the Canary build of Edge based off Chromium was just released and it works flawlessly with Viciphone.

ah, well. putting a note of where that file could be modified somewhere in the code for the lay-people wouldn't be bad either, if it's not too troublesome. Or here. Here is good, too.

Good idea.

The files is called ringing.mp3 and is in the sounds directory.

This is the line in vici_phone.js where that is set:

Code: Select all

// setup the ringing audio file
ringAudio = new Audio('sounds/ringing.mp3');


I am open to suggestions on how to make this more "lay-people" friendly than it already is.

You're right, that's already very simple.

perfect. if someone wants to pull that from a settings container or something it'll be easy enough to code for their call center. in the meantime, just editing the js file will suffice for those who care (and they'll just have to remember to update it whenever they upgrade).

Michael, can you please take a look at this topic: viewtopic.php?f=4&t=39066&p=139116

I am toying around with it on our test dialer here and it looks a lot more complete. I am seeing that ICE and STUN are making connections with less complaints on the code level in the console.

When will viciphone 2.0 be released in Github so it can be used in production?
I can still only download 1.0.0 from 2 years ago.

Guys,

Any way we could help development?
Any way we could access "alpha" source code?

We know it may need work, but currently there's no way to download the source, and we can only run it from https://dev.viciphone.com, but what if we want to host it in our own servers for offline installations?

Please let us know what we can do.

Regards,

you can run it from dev.viciphone.com? I would be interested in testing it?

Hello again,

Guys, after more than a year without hearing development news in this thread nor the Github website, I decided to create an updated version of the Viciphone, which was forked from Michael's original code.

You can browse the code at https://github.com/ccabrerar/ViciPhone or you can download the v2.0-beta1 release directly. Here's a minor list of changes:

- It's updated to use SIPJS 0.15.10, which is the most current up to date version
- SIPJS by default loads from jsdelivr CDN, but you can easily switch to offline by uncommenting one line of code
- Includes support for a translations file, in case agents don't speak english
- Spanish support added
- Changed the content of vici_phone.js to reuse code by implementing several functions which do repetitive work
- Can prefill the dial number box, and can also auto dial out once loaded

I tried it under Windows in Chrome, Firefox, Opera, Brave and Chromium Edge, but I'd love to get community feedback.

If you have any code suggestions, please open a pull request in Github.

Regards,

Thanks! Great work, i will test this and give you feedback.

ccabrera wrote:Hello again,

Guys, after more than a year without hearing development news in this thread nor the Github website, I decided to create an updated version of the Viciphone, which was forked from Michael's original code.

You can browse the code at https://github.com/ccabrerar/ViciPhone or you can download the v2.0-beta1 release directly. Here's a minor list of changes:

- It's updated to use SIPJS 0.15.10, which is the most current up to date version
- SIPJS by default loads from jsdelivr CDN, but you can easily switch to offline by uncommenting one line of code
- Includes support for a translations file, in case agents don't speak english
- Spanish support added
- Changed the content of vici_phone.js to reuse code by implementing several functions which do repetitive work
- Can prefill the dial number box, and can also auto dial out once loaded

I tried it under Windows in Chrome, Firefox, Opera, Brave and Chromium Edge, but I'd love to get community feedback.

If you have any code suggestions, please open a pull request in Github.

Regards,

Hi i want to test your viciphone 2.0 and i have used same template of viciphone.com with no luck, if i set webphone link to https://phone.viciphone.com/viciphone.php it works good and without problem but if i change it to my local directory ../agc/viciphone/viciphone.php where i have downloaded your project it's not working

this is my template :


type=friend
host=dynamic
encryption=yes
icesupport=yes
directmedia=no
transport=wss
dtlsenable=yes
dtlsverify=no
dtlscertfile=/PATH/TO/YOUR/SSL/CERT changed with my folder
dtlsprivatekey=/PATH/TO/YOU/SSL/KEY changed with my folder
dtlssetup=actpass
rtcp_mux=yes


here screenshot of errors in asterisk consolle






the phone is registered but nothing is working (no call at login and no sound) and system says me none in this session...can you help me?

Resolved overwriting sip.js and vici_phone.js with original one from viciphone.com and now all it's working.

If you overwrite those files, you are pretty much going back to Viciphone 1.0.

Your screenshots reveal that the call is being established, otherwise Viciphone would read "Registered" instead of displaying the CallerID info. My best guess is there is a JS problem, which I cannot debug unless you share your browsers JS console. Perhaps the directory structure where you locally downloaded it is somehow unaccesible?

If you want to further test, you may try configuring Vicidial > Admin > System Settings and point to the following webphone URL: https://enlaza.mx/tools/viciphonev2/viciphone.php . This URL is for *testing* only, and thus I cannot guarantee it will be available 100% of the time, but is a quick way to try Viciphone 2.0 without having the need to download/install the code from Github.

Give it a spin, and let me know your results.

this is error in consolle, can you help me?

You loaded it as an image, so I can't copy/paste the relevent URLs. But the question is whether those URLs return 404 as suggested in the output. If they do, that's your problem. Do you have a valid cert installed at that location and is that file accessible?

@Smartercom

The only relevant line of code is the one which reads about the 'isSupported' variable. However, this line isn't from my code, because I don't use that variable at all.

Seems to me that you've mixed up files from v1.0, v2.0 and the development v2.0 from Michael.

I suggest the following:
1. Remove all files you have installed locally
2. Replace all of them with my code from Github
3. Clean your browser's cache

If that doesn't help, try the live version I set up at https://enlaza.mx/tools/viciphonev2/viciphone.php . As I just tried, calling that version from a live Vicidial works just fine right now, so I'm almost sure there is something getting mixed inside your installation.

forget to clear cache..sorry

SESSION_DESCRIPTION_HANDLER_ERROR: unable to acquire streams

Seems like the connection to Vicidial isn't being established by a proper secure stream. Perhaps your certificates aren't fully trusted by the browser or you haven't given the PC permission to use your headset?

With same browser and PC all work fine If i modify SIP.js with original viciphone.

This is what I'm getting using the github code ccaberra:

Code: Select all

[Jun  9 19:46:22]     -- Called 55558600053@default
[Jun  9 19:46:22]     -- Executing [55558600053@default:1] MeetMeAdmin("Local/55558600053@default-0000001d;2", "8600053,K") in new stack
[Jun  9 19:46:22] WARNING[30910][C-000008b1]: app_meetme.c:5261 admin_exec: Conference number '8600053' not found!
[Jun  9 19:46:22]     -- Executing [55558600053@default:2] Hangup("Local/55558600053@default-0000001d;2", "") in new stack
[Jun  9 19:46:22]   == Spawn extension (default, 55558600053, 2) exited non-zero on 'Local/55558600053@default-0000001d;2'
[Jun  9 19:46:22] WARNING[30910][C-000008b1]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Jun  9 19:46:22]     -- Executing [h@default:1] AGI("Local/55558600053@default-0000001d;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Jun  9 19:46:22]     -- <Local/55558600053@default-0000001d;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0

[Jun  9 19:46:25]   == WebSocket connection from '12.1xx.24.xx0:50844' for protocol 'sip' accepted using version '13'
[Jun  9 19:46:25]     -- Registered SIP '2222' at 12.1xx.24.xx0:50844

[Jun  9 19:46:56] ERROR[16243]: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor
[Jun  9 19:46:56] ERROR[16243]: tcptls.c:488 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor
[Jun  9 19:46:56]   == WebSocket connection from '12.1xx.24.xxx:50729' forcefully closed due to fatal write error


It looks like its not gaining access to the conferences or its like dahdi isnt running but it is because the zoipers work

meetme admin wants to "manage" a conference. But the conference doesn't exist. So there's a question as to why it's managing the conference instead of invoking the conference (or perhaps "before" when it should be "after"?).

Bill,

Any ideas of this part:

ERROR[16243]: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor
[Jun 9 19:46:56] ERROR[16243]: tcptls.c:488 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor
[Jun 9 19:46:56] == WebSocket connection from '12.1xx.24.xxx:50729' forcefully closed due to fatal write error

williamconley wrote:meetme admin wants to "manage" a conference. But the conference doesn't exist. So there's a question as to why it's managing the conference instead of invoking the conference (or perhaps "before" when it should be "after"?).

Code: Select all

MeetMeAdmin("Local/55558600051@default-00000011;2", "8600051,K")

https://www.voip-info.org/asterisk-cmd-meetmeadmin/

MeetMeAdmin(confno,command[,user])

‘K’ — Kick all users out of conference\n”

This is a "terminate all channels in this conference" attempt. IE: This is meant to shut it down, and Vicidial does not bother to check if it's there before issuing this command. So the problem occurs before this. There is a failure that causes termination, this is merely the "termination" happening.

carpenox wrote:Bill,

Any ideas of this part:

ERROR[16243]: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor
[Jun 9 19:46:56] ERROR[16243]: tcptls.c:488 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor
[Jun 9 19:46:56] == WebSocket connection from '12.1xx.24.xxx:50729' forcefully closed due to fatal write error

Seems likely that this is another "shutting down" problem due to an earlier error. Was the channel/meetme room ever actually "Up"? That's the area to attack.

no it wasnt however it does initiate it if the agents use zoiper, just not viciphone 2.0

Here is the CLI during the webphone user connecting to the interface:

sorry for its length but this is all just during the connect part not anything extra, so something going on thats hinky

Code: Select all

[Jun 12 09:28:24] Reliably Transmitting (NAT) to xx1.171.xx.x:60747:
[Jun 12 09:28:24] CANCEL sip:8q8hlhsg@192.0.2.53;transport=wss SIP/2.0
[Jun 12 09:28:24] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK19440885;rport
[Jun 12 09:28:24] Max-Forwards: 70
[Jun 12 09:28:24] From: "ACagcW15919684842222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as6cbe7484
[Jun 12 09:28:24] To: <sip:8q8hlhsg@192.0.2.53;transport=wss>
[Jun 12 09:28:24] Call-ID: 1ffa144c100cff2617ee78a271da90e5@xx1.171.xx.x:5060
[Jun 12 09:28:24] CSeq: 102 CANCEL
[Jun 12 09:28:24] User-Agent: Asterisk PBX 13.32.0-vici
[Jun 12 09:28:24] Content-Length: 0
[Jun 12 09:28:24]
[Jun 12 09:28:24]
[Jun 12 09:28:24] ---
[Jun 12 09:28:24] Scheduling destruction of SIP dialog '1ffa144c100cff2617ee78a271da90e5@xx1.171.xx.x:5060' in 6400 ms (Method: INVITE)
[Jun 12 09:28:24]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 12 09:28:24]     -- Called 55558600051@default
[Jun 12 09:28:24]     -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000001;2", "8600051,K") in new stack
[Jun 12 09:28:24] WARNING[8664][C-00000001]: app_meetme.c:5261 admin_exec: Conference number '8600051' not found!
[Jun 12 09:28:24]     -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000001;2", "") in new stack
[Jun 12 09:28:24]   == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000001;2'
[Jun 12 09:28:24] WARNING[8664][C-00000001]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Jun 12 09:28:24]     -- Executing [h@default:1] AGI("Local/55558600051@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Jun 12 09:28:24]     -- <Local/55558600051@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Jun 12 09:28:24]
[Jun 12 09:28:24] <--- SIP read from WS:xx1.171.xx.x:60747 --->
[Jun 12 09:28:24] SIP/2.0 200 OK
[Jun 12 09:28:24] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK19440885;rport
[Jun 12 09:28:24] From: "ACagcW15919684842222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as6cbe7484
[Jun 12 09:28:24] To: <sip:8q8hlhsg@192.0.2.53;transport=wss>;tag=lbgp25b290
[Jun 12 09:28:24] CSeq: 102 CANCEL
[Jun 12 09:28:24] Call-ID: 1ffa144c100cff2617ee78a271da90e5@xx1.171.xx.x:5060
[Jun 12 09:28:24] Supported: outbound
[Jun 12 09:28:24] User-Agent: VICIphone 2.0
[Jun 12 09:28:24] Content-Length: 0
[Jun 12 09:28:24]
[Jun 12 09:28:24] <------------->
[Jun 12 09:28:24] --- (9 headers 0 lines) ---
[Jun 12 09:28:24]
[Jun 12 09:28:24] <--- SIP read from WS:xx1.171.xx.x:60747 --->
[Jun 12 09:28:24] SIP/2.0 487 Request Terminated
[Jun 12 09:28:24] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK19440885;rport
[Jun 12 09:28:24] From: "ACagcW15919684842222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as6cbe7484
[Jun 12 09:28:24] To: <sip:8q8hlhsg@192.0.2.53;transport=wss>;tag=p9e784nmbn
[Jun 12 09:28:24] CSeq: 102 INVITE
[Jun 12 09:28:24] Call-ID: 1ffa144c100cff2617ee78a271da90e5@xx1.171.xx.x:5060
[Jun 12 09:28:24] Supported: outbound
[Jun 12 09:28:24] User-Agent: VICIphone 2.0
[Jun 12 09:28:24] Content-Length: 0
[Jun 12 09:28:24]
[Jun 12 09:28:24] <------------->
[Jun 12 09:28:24] --- (9 headers 0 lines) ---
[Jun 12 09:28:24] Transmitting (NAT) to xx1.171.xx.x:60747:
[Jun 12 09:28:24] ACK sip:8q8hlhsg@192.0.2.53;transport=wss SIP/2.0
[Jun 12 09:28:24] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK19440885;rport
[Jun 12 09:28:24] Max-Forwards: 70
[Jun 12 09:28:24] From: "ACagcW15919684842222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as6cbe7484
[Jun 12 09:28:24] To: <sip:8q8hlhsg@192.0.2.53;transport=wss>;tag=p9e784nmbn
[Jun 12 09:28:24] Contact: <sip:202xxx8707@xx1.171.xx.x:5060;transport=ws>
[Jun 12 09:28:24] Call-ID: 1ffa144c100cff2617ee78a271da90e5@xx1.171.xx.x:5060
[Jun 12 09:28:24] CSeq: 102 ACK
[Jun 12 09:28:24] User-Agent: Asterisk PBX 13.32.0-vici
[Jun 12 09:28:24] Content-Length: 0
[Jun 12 09:28:24]
[Jun 12 09:28:24]
[Jun 12 09:28:24] ---
[Jun 12 09:28:24] Scheduling destruction of SIP dialog '1ffa144c100cff2617ee78a271da90e5@xx1.171.xx.x:5060' in 6400 ms (Method: INVITE)
[Jun 12 09:28:25]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 12 09:28:25]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 12 09:28:25]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 12 09:28:26]
[Jun 12 09:28:26] <--- SIP read from WS:xx1.171.xx.x:60747 --->
[Jun 12 09:28:26] REGISTER sip:xx1.171.xx.x SIP/2.0
[Jun 12 09:28:26] Via: SIP/2.0/TCP 192.0.2.53;branch=z9hG4bK6085544
[Jun 12 09:28:26] To: "2222" <sip:2222@xx1.171.xx.x>
[Jun 12 09:28:26] From: "2222" <sip:2222@xx1.171.xx.x>;tag=aghq6bk6dn
[Jun 12 09:28:26] CSeq: 7461 REGISTER
[Jun 12 09:28:26] Call-ID: umejua6b0p2bs8li5plnol
[Jun 12 09:28:26] Max-Forwards: 70
[Jun 12 09:28:26] Authorization: Digest algorithm=MD5, username="2222", realm="cyburity.tk", nonce="2ee20980", uri="sip:xx1.171.xx.x", response="6e7dc3cac5c9ad6ec4092922f93f56a4"
[Jun 12 09:28:26] Contact: <sip:8q8hlhsg@192.0.2.53;transport=wss>;expires=0
[Jun 12 09:28:26] Supported: outbound, path, gruu
[Jun 12 09:28:26] User-Agent: VICIphone 2.0
[Jun 12 09:28:26] Content-Length: 0
[Jun 12 09:28:26]
[Jun 12 09:28:26] <------------->
[Jun 12 09:28:26] --- (12 headers 0 lines) ---
[Jun 12 09:28:26] Sending to xx1.171.xx.x:60747 (NAT)
[Jun 12 09:28:26] Sending to xx1.171.xx.x:60747 (NAT)
[Jun 12 09:28:26]
[Jun 12 09:28:26] <--- Transmitting (NAT) to xx1.171.xx.x:60747 --->
[Jun 12 09:28:26] SIP/2.0 401 Unauthorized
[Jun 12 09:28:26] Via: SIP/2.0/TCP 192.0.2.53;branch=z9hG4bK6085544;received=xx1.171.xx.x;rport=60747
[Jun 12 09:28:26] From: "2222" <sip:2222@xx1.171.xx.x>;tag=aghq6bk6dn
[Jun 12 09:28:26] To: "2222" <sip:2222@xx1.171.xx.x>;tag=as0b89a4bd
[Jun 12 09:28:26] Call-ID: umejua6b0p2bs8li5plnol
[Jun 12 09:28:26] CSeq: 7461 REGISTER
[Jun 12 09:28:26] Server: Asterisk PBX 13.32.0-vici
[Jun 12 09:28:26] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jun 12 09:28:26] Supported: replaces, timer
[Jun 12 09:28:26] WWW-Authenticate: Digest algorithm=MD5, realm="cyburity.tk", nonce="7ecfc1ff"
[Jun 12 09:28:26] Content-Length: 0
[Jun 12 09:28:26]
[Jun 12 09:28:26]
[Jun 12 09:28:26] <------------>
[Jun 12 09:28:26] Scheduling destruction of SIP dialog 'umejua6b0p2bs8li5plnol' in 32000 ms (Method: REGISTER)
[Jun 12 09:28:28]
[Jun 12 09:28:28] <--- SIP read from UDP:76.110.127.205:35978 --->
[Jun 12 09:28:28]
[Jun 12 09:28:28]
[Jun 12 09:28:28] <------------->
[Jun 12 09:28:29]   == WebSocket connection from 'xx1.171.xx.x:60751' for protocol 'sip' accepted using version '13'
[Jun 12 09:28:30]
[Jun 12 09:28:30] <--- SIP read from WS:xx1.171.xx.x:60751 --->
[Jun 12 09:28:30] REGISTER sip:xx1.171.xx.x SIP/2.0
[Jun 12 09:28:30] Via: SIP/2.0/TCP 192.0.2.218;branch=z9hG4bK2555983
[Jun 12 09:28:30] To: "2222" <sip:2222@xx1.171.xx.x>
[Jun 12 09:28:30] From: "2222" <sip:2222@xx1.171.xx.x>;tag=2caclbpj8d
[Jun 12 09:28:30] CSeq: 5594 REGISTER
[Jun 12 09:28:30] Call-ID: ndgfl0oaeb75rbvca0bsud
[Jun 12 09:28:30] Max-Forwards: 70
[Jun 12 09:28:30] Contact: <sip:18s70scg@192.0.2.218;transport=wss>;expires=600
[Jun 12 09:28:30] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
[Jun 12 09:28:30] Supported: outbound, path, gruu
[Jun 12 09:28:30] User-Agent: VICIphone 2.0
[Jun 12 09:28:30] Content-Length: 0
[Jun 12 09:28:30]
[Jun 12 09:28:30] <------------->
[Jun 12 09:28:30] --- (12 headers 0 lines) ---
[Jun 12 09:28:30] Sending to xx1.171.xx.x:60751 (NAT)
[Jun 12 09:28:30] Sending to xx1.171.xx.x:60751 (NAT)
[Jun 12 09:28:30]
[Jun 12 09:28:30] <--- Transmitting (NAT) to xx1.171.xx.x:60751 --->
[Jun 12 09:28:30] SIP/2.0 401 Unauthorized
[Jun 12 09:28:30] Via: SIP/2.0/TCP 192.0.2.218;branch=z9hG4bK2555983;received=xx1.171.xx.x;rport=60751
[Jun 12 09:28:30] From: "2222" <sip:2222@xx1.171.xx.x>;tag=2caclbpj8d
[Jun 12 09:28:30] To: "2222" <sip:2222@xx1.171.xx.x>;tag=as0c812ff7
[Jun 12 09:28:30] Call-ID: ndgfl0oaeb75rbvca0bsud
[Jun 12 09:28:30] CSeq: 5594 REGISTER
[Jun 12 09:28:30] Server: Asterisk PBX 13.32.0-vici
[Jun 12 09:28:30] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jun 12 09:28:30] Supported: replaces, timer
[Jun 12 09:28:30] WWW-Authenticate: Digest algorithm=MD5, realm="cyburity.tk", nonce="5a59971d"
[Jun 12 09:28:30] Content-Length: 0
[Jun 12 09:28:30]
[Jun 12 09:28:30]
[Jun 12 09:28:30] <------------>
[Jun 12 09:28:30] Scheduling destruction of SIP dialog 'ndgfl0oaeb75rbvca0bsud' in 32000 ms (Method: REGISTER)
[Jun 12 09:28:30]
[Jun 12 09:28:30] <--- SIP read from WS:xx1.171.xx.x:60751 --->
[Jun 12 09:28:30] REGISTER sip:xx1.171.xx.x SIP/2.0
[Jun 12 09:28:30] Via: SIP/2.0/TCP 192.0.2.218;branch=z9hG4bK6408062
[Jun 12 09:28:30] To: "2222" <sip:2222@xx1.171.xx.x>
[Jun 12 09:28:30] From: "2222" <sip:2222@xx1.171.xx.x>;tag=2caclbpj8d
[Jun 12 09:28:30] CSeq: 5595 REGISTER
[Jun 12 09:28:30] Call-ID: ndgfl0oaeb75rbvca0bsud
[Jun 12 09:28:30] Max-Forwards: 70
[Jun 12 09:28:30] Authorization: Digest algorithm=MD5, username="2222", realm="cyburity.tk", nonce="5a59971d", uri="sip:xx1.171.xx.x", response="997e9db6adacbc38e474e7277f7c9972"
[Jun 12 09:28:30] Contact: <sip:18s70scg@192.0.2.218;transport=wss>;expires=600
[Jun 12 09:28:30] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
[Jun 12 09:28:30] Supported: outbound, path, gruu
[Jun 12 09:28:30] User-Agent: VICIphone 2.0
[Jun 12 09:28:30] Content-Length: 0
[Jun 12 09:28:30]
[Jun 12 09:28:30] <------------->
[Jun 12 09:28:30] --- (13 headers 0 lines) ---
[Jun 12 09:28:30] Sending to xx1.171.xx.x:60751 (NAT)
[Jun 12 09:28:30]     -- Registered SIP '2222' at xx1.171.xx.x:60751
[Jun 12 09:28:30] Reliably Transmitting (NAT) to xx1.171.xx.x:60751:
[Jun 12 09:28:30] OPTIONS sip:18s70scg@192.0.2.218;transport=wss SIP/2.0
[Jun 12 09:28:30] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK09093d89;rport
[Jun 12 09:28:30] Max-Forwards: 70
[Jun 12 09:28:30] From: "asterisk" <sip:asterisk@xx1.171.xx.x>;tag=as42294779
[Jun 12 09:28:30] To: <sip:18s70scg@192.0.2.218;transport=wss>
[Jun 12 09:28:30] Contact: <sip:asterisk@xx1.171.xx.x:5060;transport=ws>
[Jun 12 09:28:30] Call-ID: 423fc67b6896944204cad2b5461d98e1@xx1.171.xx.x:5060
[Jun 12 09:28:30] CSeq: 102 OPTIONS
[Jun 12 09:28:30] User-Agent: Asterisk PBX 13.32.0-vici
[Jun 12 09:28:30] Date: Fri, 12 Jun 2020 13:28:30 GMT
[Jun 12 09:28:30] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jun 12 09:28:30] Supported: replaces, timer
[Jun 12 09:28:30] Content-Length: 0
[Jun 12 09:28:30]
[Jun 12 09:28:30]
[Jun 12 09:28:30] ---
[Jun 12 09:28:30]
[Jun 12 09:28:30] <--- Transmitting (NAT) to xx1.171.xx.x:60751 --->
[Jun 12 09:28:30] SIP/2.0 200 OK
[Jun 12 09:28:30] Via: SIP/2.0/TCP 192.0.2.218;branch=z9hG4bK6408062;received=xx1.171.xx.x;rport=60751
[Jun 12 09:28:30] From: "2222" <sip:2222@xx1.171.xx.x>;tag=2caclbpj8d
[Jun 12 09:28:30] To: "2222" <sip:2222@xx1.171.xx.x>;tag=as0c812ff7
[Jun 12 09:28:30] Call-ID: ndgfl0oaeb75rbvca0bsud
[Jun 12 09:28:30] CSeq: 5595 REGISTER
[Jun 12 09:28:30] Server: Asterisk PBX 13.32.0-vici
[Jun 12 09:28:30] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jun 12 09:28:30] Supported: replaces, timer
[Jun 12 09:28:30] Expires: 600
[Jun 12 09:28:30] Contact: <sip:18s70scg@192.0.2.218;transport=wss>;expires=600
[Jun 12 09:28:30] Date: Fri, 12 Jun 2020 13:28:30 GMT
[Jun 12 09:28:30] Content-Length: 0
[Jun 12 09:28:30]
[Jun 12 09:28:30]
[Jun 12 09:28:30] <------------>
[Jun 12 09:28:30] Scheduling destruction of SIP dialog '52389f23727c297b3e045a1e77f9e0e0@xx1.171.xx.x:5060' in 6400 ms (Method: NOTIFY)
[Jun 12 09:28:30] Reliably Transmitting (NAT) to xx1.171.xx.x:60751:
[Jun 12 09:28:30] NOTIFY sip:18s70scg@192.0.2.218;transport=wss SIP/2.0
[Jun 12 09:28:30] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK295a54ea;rport
[Jun 12 09:28:30] Max-Forwards: 70
[Jun 12 09:28:30] From: "asterisk" <sip:asterisk@xx1.171.xx.x>;tag=as28ac362b
[Jun 12 09:28:30] To: <sip:18s70scg@192.0.2.218;transport=wss>
[Jun 12 09:28:30] Contact: <sip:asterisk@xx1.171.xx.x:5060;transport=ws>
[Jun 12 09:28:30] Call-ID: 52389f23727c297b3e045a1e77f9e0e0@xx1.171.xx.x:5060
[Jun 12 09:28:30] CSeq: 102 NOTIFY
[Jun 12 09:28:30] User-Agent: Asterisk PBX 13.32.0-vici
[Jun 12 09:28:30] Event: message-summary
[Jun 12 09:28:30] Content-Type: application/simple-message-summary
[Jun 12 09:28:30] Content-Length: 107
[Jun 12 09:28:30]
[Jun 12 09:28:30] Messages-Waiting: no
[Jun 12 09:28:30] Message-Account: sip:asterisk@xx1.171.xx.x;transport=WS
[Jun 12 09:28:30] Voice-Message: 0/0 (0/0)
[Jun 12 09:28:30]
[Jun 12 09:28:30] ---
[Jun 12 09:28:30] Scheduling destruction of SIP dialog 'ndgfl0oaeb75rbvca0bsud' in 32000 ms (Method: REGISTER)
[Jun 12 09:28:30]
[Jun 12 09:28:30] <--- SIP read from WS:xx1.171.xx.x:60751 --->
[Jun 12 09:28:30] SIP/2.0 200 OK
[Jun 12 09:28:30] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK09093d89;rport
[Jun 12 09:28:30] From: "asterisk" <sip:asterisk@xx1.171.xx.x>;tag=as42294779
[Jun 12 09:28:30] To: <sip:18s70scg@192.0.2.218;transport=wss>;tag=8nmb4j28cr
[Jun 12 09:28:30] CSeq: 102 OPTIONS
[Jun 12 09:28:30] Call-ID: 423fc67b6896944204cad2b5461d98e1@xx1.171.xx.x:5060
[Jun 12 09:28:30] Supported: outbound
[Jun 12 09:28:30] User-Agent: VICIphone 2.0
[Jun 12 09:28:30] Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE
[Jun 12 09:28:30] Accept: application/sdp,application/dtmf-relay
[Jun 12 09:28:30] Content-Length: 0
[Jun 12 09:28:30]
[Jun 12 09:28:30] <------------->
[Jun 12 09:28:30] --- (11 headers 0 lines) ---
[Jun 12 09:28:30]
[Jun 12 09:28:30] <--- SIP read from WS:xx1.171.xx.x:60751 --->
[Jun 12 09:28:30] SIP/2.0 481 Call/Transaction Does Not Exist
[Jun 12 09:28:30] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK295a54ea;rport
[Jun 12 09:28:30] From: "asterisk" <sip:asterisk@xx1.171.xx.x>;tag=as28ac362b
[Jun 12 09:28:30] To: <sip:18s70scg@192.0.2.218;transport=wss>;tag=2vkm928jpc
[Jun 12 09:28:30] CSeq: 102 NOTIFY
[Jun 12 09:28:30] Call-ID: 52389f23727c297b3e045a1e77f9e0e0@xx1.171.xx.x:5060
[Jun 12 09:28:30] Supported: outbound
[Jun 12 09:28:30] User-Agent: VICIphone 2.0
[Jun 12 09:28:30] Content-Length: 0
[Jun 12 09:28:30]
[Jun 12 09:28:30] <------------->
[Jun 12 09:28:30] --- (9 headers 0 lines) ---
[Jun 12 09:28:31] Really destroying SIP dialog '423fc67b6896944204cad2b5461d98e1@xx1.171.xx.x:5060' Method: OPTIONS
[Jun 12 09:28:31] Really destroying SIP dialog '52389f23727c297b3e045a1e77f9e0e0@xx1.171.xx.x:5060' Method: NOTIFY
[Jun 12 09:28:31] Really destroying SIP dialog '1ffa144c100cff2617ee78a271da90e5@xx1.171.xx.x:5060' Method: INVITE
[Jun 12 09:28:33]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 12 09:28:33]   == Using SIP RTP CoS mark 5
[Jun 12 09:28:33] Audio is at 10228
[Jun 12 09:28:33] Adding codec ulaw to SDP
[Jun 12 09:28:33] Adding codec alaw to SDP
[Jun 12 09:28:33] Adding codec gsm to SDP
[Jun 12 09:28:33] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 12 09:28:33] Reliably Transmitting (NAT) to xx1.171.xx.x:60751:
[Jun 12 09:28:33] INVITE sip:18s70scg@192.0.2.218;transport=wss SIP/2.0
[Jun 12 09:28:33] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK005fc234;rport
[Jun 12 09:28:33] Max-Forwards: 70
[Jun 12 09:28:33] From: "ACagcW15919685122222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as10894185
[Jun 12 09:28:33] To: <sip:18s70scg@192.0.2.218;transport=wss>
[Jun 12 09:28:33] Contact: <sip:202xxx8707@xx1.171.xx.x:5060;transport=ws>
[Jun 12 09:28:33] Call-ID: 27285e18662b1446606fc5bf7e8f6432@xx1.171.xx.x:5060
[Jun 12 09:28:33] CSeq: 102 INVITE
[Jun 12 09:28:33] User-Agent: Asterisk PBX 13.32.0-vici
[Jun 12 09:28:33] Date: Fri, 12 Jun 2020 13:28:33 GMT
[Jun 12 09:28:33] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jun 12 09:28:33] Supported: replaces, timer
[Jun 12 09:28:33] Remote-Party-ID: "ACagcW15919685122222222222222222" <sip:202xxx8707@xx1.171.xx.x>;party=calling;privacy=off;screen=no
[Jun 12 09:28:33] Content-Type: application/sdp
[Jun 12 09:28:33] Content-Length: 694
[Jun 12 09:28:33]
[Jun 12 09:28:33] v=0
[Jun 12 09:28:33] o=root 1265590450 1265590450 IN IP4 xx1.171.xx.x
[Jun 12 09:28:33] s=Asterisk PBX 13.32.0-vici
[Jun 12 09:28:33] c=IN IP4 xx1.171.xx.x
[Jun 12 09:28:33] t=0 0
[Jun 12 09:28:33] m=audio 10228 RTP/SAVPF 0 8 3 101
[Jun 12 09:28:33] a=rtpmap:0 PCMU/8000
[Jun 12 09:28:33] a=rtpmap:8 PCMA/8000
[Jun 12 09:28:33] a=rtpmap:3 GSM/8000
[Jun 12 09:28:33] a=rtpmap:101 telephone-event/8000
[Jun 12 09:28:33] a=fmtp:101 0-16
[Jun 12 09:28:33] a=ptime:20
[Jun 12 09:28:33] a=maxptime:150
[Jun 12 09:28:33] a=ice-ufrag:478bbaab38a2a10451b23ee53b17003c
[Jun 12 09:28:33] a=ice-pwd:7b28ee400f6613f3065a691104a8595e
[Jun 12 09:28:33] a=candidate:H1fab84d5 1 UDP 2130706431 xx1.171.xx.x 10228 typ host
[Jun 12 09:28:33] a=candidate:H1fab84d5 2 UDP 2130706430 xx1.171.xx.x 10229 typ host
[Jun 12 09:28:33] a=connection:new
[Jun 12 09:28:33] a=setup:actpass
[Jun 12 09:28:33] a=fingerprint:SHA-256 A5:F5:5B:36:D1:74:DD:85:B3:D8:44:0D:ED:D3:E9:A0:4B:34:C7:E7:8F:B0:CA:AF:34:DB:44:BA:BB:83:04:05
[Jun 12 09:28:33] a=sendrecv
[Jun 12 09:28:33]
[Jun 12 09:28:33] ---
[Jun 12 09:28:33]     -- Called 2222
[Jun 12 09:28:33]
[Jun 12 09:28:33] <--- SIP read from WS:xx1.171.xx.x:60751 --->
[Jun 12 09:28:33] SIP/2.0 100 Trying
[Jun 12 09:28:33] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK005fc234;rport
[Jun 12 09:28:33] From: "ACagcW15919685122222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as10894185
[Jun 12 09:28:33] To: <sip:18s70scg@192.0.2.218;transport=wss>
[Jun 12 09:28:33] CSeq: 102 INVITE
[Jun 12 09:28:33] Call-ID: 27285e18662b1446606fc5bf7e8f6432@xx1.171.xx.x:5060
[Jun 12 09:28:33] Supported: outbound
[Jun 12 09:28:33] User-Agent: VICIphone 2.0
[Jun 12 09:28:33] Content-Length: 0
[Jun 12 09:28:33]
[Jun 12 09:28:33] <------------->
[Jun 12 09:28:33] --- (9 headers 0 lines) ---
[Jun 12 09:28:33]
[Jun 12 09:28:33] <--- SIP read from WS:xx1.171.xx.x:60751 --->
[Jun 12 09:28:33] SIP/2.0 180 Ringing
[Jun 12 09:28:33] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK005fc234;rport
[Jun 12 09:28:33] From: "ACagcW15919685122222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as10894185
[Jun 12 09:28:33] To: <sip:18s70scg@192.0.2.218;transport=wss>;tag=8g8bg611he
[Jun 12 09:28:33] CSeq: 102 INVITE
[Jun 12 09:28:33] Call-ID: 27285e18662b1446606fc5bf7e8f6432@xx1.171.xx.x:5060
[Jun 12 09:28:33] Supported: outbound
[Jun 12 09:28:33] User-Agent: VICIphone 2.0
[Jun 12 09:28:33] Contact: <sip:18s70scg@192.0.2.218;transport=wss>
[Jun 12 09:28:33] Content-Length: 0
[Jun 12 09:28:33]
[Jun 12 09:28:33] <------------->
[Jun 12 09:28:33] --- (10 headers 0 lines) ---
[Jun 12 09:28:33] sip_route_dump: route/path hop: <sip:18s70scg@192.0.2.218;transport=wss>
[Jun 12 09:28:33]     -- SIP/2222-00000005 is ringing
[Jun 12 09:28:33]
[Jun 12 09:28:33] <--- SIP read from WS:xx1.171.xx.x:60751 --->
[Jun 12 09:28:33] SIP/2.0 480 Temporarily Unavailable
[Jun 12 09:28:33] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK005fc234;rport
[Jun 12 09:28:33] From: "ACagcW15919685122222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as10894185
[Jun 12 09:28:33] To: <sip:18s70scg@192.0.2.218;transport=wss>;tag=8g8bg611he
[Jun 12 09:28:33] CSeq: 102 INVITE
[Jun 12 09:28:33] Call-ID: 27285e18662b1446606fc5bf7e8f6432@xx1.171.xx.x:5060
[Jun 12 09:28:33] Supported: outbound
[Jun 12 09:28:33] User-Agent: VICIphone 2.0
[Jun 12 09:28:33] Content-Length: 0
[Jun 12 09:28:33]
[Jun 12 09:28:33] <------------->
[Jun 12 09:28:33] --- (9 headers 0 lines) ---
[Jun 12 09:28:33] Transmitting (NAT) to xx1.171.xx.x:60751:
[Jun 12 09:28:33] ACK sip:18s70scg@192.0.2.218;transport=wss SIP/2.0
[Jun 12 09:28:33] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK005fc234;rport
[Jun 12 09:28:33] Max-Forwards: 70
[Jun 12 09:28:33] From: "ACagcW15919685122222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as10894185
[Jun 12 09:28:33] To: <sip:18s70scg@192.0.2.218;transport=wss>;tag=8g8bg611he
[Jun 12 09:28:33] Contact: <sip:202xxx8707@xx1.171.xx.x:5060;transport=ws>
[Jun 12 09:28:33] Call-ID: 27285e18662b1446606fc5bf7e8f6432@xx1.171.xx.x:5060
[Jun 12 09:28:33] CSeq: 102 ACK
[Jun 12 09:28:33] User-Agent: Asterisk PBX 13.32.0-vici
[Jun 12 09:28:33] Content-Length: 0
[Jun 12 09:28:33]
[Jun 12 09:28:33]
[Jun 12 09:28:33] ---
[Jun 12 09:28:33]     -- SIP/2222-00000005 is busy
[Jun 12 09:28:33] Scheduling destruction of SIP dialog '27285e18662b1446606fc5bf7e8f6432@xx1.171.xx.x:5060' in 6400 ms (Method: INVITE)
[Jun 12 09:28:34]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 12 09:28:34]
[Jun 12 09:28:34] <--- SIP read from UDP:xx1.171.xx.x:5060 --->
[Jun 12 09:28:34]
[Jun 12 09:28:34]
[Jun 12 09:28:34] <------------->
[Jun 12 09:28:40] Really destroying SIP dialog '27285e18662b1446606fc5bf7e8f6432@xx1.171.xx.x:5060' Method: INVITE
[Jun 12 09:28:47] Really destroying SIP dialog 'cuZoK8c5DmKnJnJF4LupYw..' Method: REGISTER
[Jun 12 09:28:51] Reliably Transmitting (no NAT) to 1X2.212.218.xx:5060:
[Jun 12 09:28:51] OPTIONS sip:1X2.212.218.xx SIP/2.0
[Jun 12 09:28:51] Via: SIP/2.0/UDP xx1.171.xx.x:5060;branch=z9hG4bK742729f2
[Jun 12 09:28:51] Max-Forwards: 70
[Jun 12 09:28:51] From: "asterisk" <sip:asterisk@xx1.171.xx.x>;tag=as7a291737
[Jun 12 09:28:51] To: <sip:1X2.212.218.xx>
[Jun 12 09:28:51] Contact: <sip:asterisk@xx1.171.xx.x:5060>
[Jun 12 09:28:51] Call-ID: 23e32801001b5241764a9c3266203e5f@xx1.171.xx.x:5060
[Jun 12 09:28:51] CSeq: 102 OPTIONS
[Jun 12 09:28:51] User-Agent: Asterisk PBX 13.32.0-vici
[Jun 12 09:28:51] Date: Fri, 12 Jun 2020 13:28:51 GMT
[Jun 12 09:28:51] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jun 12 09:28:51] Supported: replaces, timer
[Jun 12 09:28:51] Content-Length: 0
[Jun 12 09:28:51]
[Jun 12 09:28:51]
[Jun 12 09:28:51] ---
[Jun 12 09:28:51]
[Jun 12 09:28:51] <--- SIP read from UDP:1X2.212.218.xx:5060 --->
[Jun 12 09:28:51] SIP/2.0 200 ok
[Jun 12 09:28:51] Via: SIP/2.0/UDP xx1.171.xx.x:5060;branch=z9hG4bK742729f2
[Jun 12 09:28:51] From: "asterisk" <sip:asterisk@xx1.171.xx.x>;tag=as7a291737
[Jun 12 09:28:51] To: <sip:1X2.212.218.xx>;tag=3348068d66121f4810c19dd2a2f673ed.07fd
[Jun 12 09:28:51] Call-ID: 23e32801001b5241764a9c3266203e5f@xx1.171.xx.x:5060
[Jun 12 09:28:51] CSeq: 102 OPTIONS
[Jun 12 09:28:51] Server: AlcazarProxy 1.30
[Jun 12 09:28:51] Content-Length: 0
[Jun 12 09:28:51]
[Jun 12 09:28:51] <------------->
[Jun 12 09:28:51] --- (8 headers 0 lines) ---
[Jun 12 09:28:51] Really destroying SIP dialog '23e32801001b5241764a9c3266203e5f@xx1.171.xx.x:5060' Method: OPTIONS
[Jun 12 09:28:56]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 12 09:28:56]   == Using SIP RTP CoS mark 5
[Jun 12 09:28:56] Audio is at 18528
[Jun 12 09:28:56] Adding codec ulaw to SDP
[Jun 12 09:28:56] Adding codec alaw to SDP
[Jun 12 09:28:56] Adding codec gsm to SDP
[Jun 12 09:28:56] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 12 09:28:56] Reliably Transmitting (NAT) to xx1.171.xx.x:60751:
[Jun 12 09:28:56] INVITE sip:18s70scg@192.0.2.218;transport=wss SIP/2.0
[Jun 12 09:28:56] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK3ce6f173;rport
[Jun 12 09:28:56] Max-Forwards: 70
[Jun 12 09:28:56] From: "ACagcW15919685352222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as2e59afb1
[Jun 12 09:28:56] To: <sip:18s70scg@192.0.2.218;transport=wss>
[Jun 12 09:28:56] Contact: <sip:202xxx8707@xx1.171.xx.x:5060;transport=ws>
[Jun 12 09:28:56] Call-ID: 5a1745f518bc2fca26ac8cf21c854eda@xx1.171.xx.x:5060
[Jun 12 09:28:56] CSeq: 102 INVITE
[Jun 12 09:28:56] User-Agent: Asterisk PBX 13.32.0-vici
[Jun 12 09:28:56] Date: Fri, 12 Jun 2020 13:28:56 GMT
[Jun 12 09:28:56] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jun 12 09:28:56] Supported: replaces, timer
[Jun 12 09:28:56] Remote-Party-ID: "ACagcW15919685352222222222222222" <sip:202xxx8707@xx1.171.xx.x>;party=calling;privacy=off;screen=no
[Jun 12 09:28:56] Content-Type: application/sdp
[Jun 12 09:28:56] Content-Length: 694
[Jun 12 09:28:56]
[Jun 12 09:28:56] v=0
[Jun 12 09:28:56] o=root 1926833548 1926833548 IN IP4 xx1.171.xx.x
[Jun 12 09:28:56] s=Asterisk PBX 13.32.0-vici
[Jun 12 09:28:56] c=IN IP4 xx1.171.xx.x
[Jun 12 09:28:56] t=0 0
[Jun 12 09:28:56] m=audio 18528 RTP/SAVPF 0 8 3 101
[Jun 12 09:28:56] a=rtpmap:0 PCMU/8000
[Jun 12 09:28:56] a=rtpmap:8 PCMA/8000
[Jun 12 09:28:56] a=rtpmap:3 GSM/8000
[Jun 12 09:28:56] a=rtpmap:101 telephone-event/8000
[Jun 12 09:28:56] a=fmtp:101 0-16
[Jun 12 09:28:56] a=ptime:20
[Jun 12 09:28:56] a=maxptime:150
[Jun 12 09:28:56] a=ice-ufrag:60eb87497a76de4e7ffa7bb63618c61d
[Jun 12 09:28:56] a=ice-pwd:5b45bb9c1b06b8151b6685ff17af4cda
[Jun 12 09:28:56] a=candidate:H1fab84d5 1 UDP 2130706431 xx1.171.xx.x 18528 typ host
[Jun 12 09:28:56] a=candidate:H1fab84d5 2 UDP 2130706430 xx1.171.xx.x 18529 typ host
[Jun 12 09:28:56] a=connection:new
[Jun 12 09:28:56] a=setup:actpass
[Jun 12 09:28:56] a=fingerprint:SHA-256 A5:F5:5B:36:D1:74:DD:85:B3:D8:44:0D:ED:D3:E9:A0:4B:34:C7:E7:8F:B0:CA:AF:34:DB:44:BA:BB:83:04:05
[Jun 12 09:28:56] a=sendrecv
[Jun 12 09:28:56]
[Jun 12 09:28:56] ---
[Jun 12 09:28:56]     -- Called 2222
[Jun 12 09:28:56]
[Jun 12 09:28:56] <--- SIP read from WS:xx1.171.xx.x:60751 --->
[Jun 12 09:28:56] SIP/2.0 100 Trying
[Jun 12 09:28:56] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK3ce6f173;rport
[Jun 12 09:28:56] From: "ACagcW15919685352222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as2e59afb1
[Jun 12 09:28:56] To: <sip:18s70scg@192.0.2.218;transport=wss>
[Jun 12 09:28:56] CSeq: 102 INVITE
[Jun 12 09:28:56] Call-ID: 5a1745f518bc2fca26ac8cf21c854eda@xx1.171.xx.x:5060
[Jun 12 09:28:56] Supported: outbound
[Jun 12 09:28:56] User-Agent: VICIphone 2.0
[Jun 12 09:28:56] Content-Length: 0
[Jun 12 09:28:56]
[Jun 12 09:28:56] <------------->
[Jun 12 09:28:56] --- (9 headers 0 lines) ---
[Jun 12 09:28:56]
[Jun 12 09:28:56] <--- SIP read from WS:xx1.171.xx.x:60751 --->
[Jun 12 09:28:56] SIP/2.0 180 Ringing
[Jun 12 09:28:56] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK3ce6f173;rport
[Jun 12 09:28:56] From: "ACagcW15919685352222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as2e59afb1
[Jun 12 09:28:56] To: <sip:18s70scg@192.0.2.218;transport=wss>;tag=n9g11ribc3
[Jun 12 09:28:56] CSeq: 102 INVITE
[Jun 12 09:28:56] Call-ID: 5a1745f518bc2fca26ac8cf21c854eda@xx1.171.xx.x:5060
[Jun 12 09:28:56] Supported: outbound
[Jun 12 09:28:56] User-Agent: VICIphone 2.0
[Jun 12 09:28:56] Contact: <sip:18s70scg@192.0.2.218;transport=wss>
[Jun 12 09:28:56] Content-Length: 0
[Jun 12 09:28:56]
[Jun 12 09:28:56] <------------->
[Jun 12 09:28:56] --- (10 headers 0 lines) ---
[Jun 12 09:28:56] sip_route_dump: route/path hop: <sip:18s70scg@192.0.2.218;transport=wss>
[Jun 12 09:28:56]     -- SIP/2222-00000006 is ringing
[Jun 12 09:28:58]
[Jun 12 09:28:58] <------------->
[Jun 12 09:28:59] Really destroying SIP dialog 'umejua6b0p2bs8li5plnol' Method: REGISTER
[Jun 12 09:28:59] ERROR[4869]: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor
[Jun 12 09:28:59] ERROR[4869]: tcptls.c:488 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor
[Jun 12 09:28:59]   == WebSocket connection from 'xx1.171.xx.x:60747' forcefully closed due to fatal write error
[Jun 12 09:29:00]
[Jun 12 09:29:00] ---
[Jun 12 09:29:00]
[Jun 12 09:29:00]
[Jun 12 09:29:06]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/2222-00000006
[Jun 12 09:29:06] Scheduling destruction of SIP dialog '5a1745f518bc2fca26ac8cf21c854eda@xx1.171.xx.x:5060' in 6400 ms (Method: INVITE)
[Jun 12 09:29:06] Reliably Transmitting (NAT) to xx1.171.xx.x:60751:
[Jun 12 09:29:06] CANCEL sip:18s70scg@192.0.2.218;transport=wss SIP/2.0
[Jun 12 09:29:06] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK3ce6f173;rport
[Jun 12 09:29:06] Max-Forwards: 70
[Jun 12 09:29:06] From: "ACagcW15919685352222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as2e59afb1
[Jun 12 09:29:06] To: <sip:18s70scg@192.0.2.218;transport=wss>
[Jun 12 09:29:06] Call-ID: 5a1745f518bc2fca26ac8cf21c854eda@xx1.171.xx.x:5060
[Jun 12 09:29:06] CSeq: 102 CANCEL
[Jun 12 09:29:06] User-Agent: Asterisk PBX 13.32.0-vici
[Jun 12 09:29:06] Content-Length: 0
[Jun 12 09:29:06]
[Jun 12 09:29:06]
[Jun 12 09:29:06] ---
[Jun 12 09:29:06] Scheduling destruction of SIP dialog '5a1745f518bc2fca26ac8cf21c854eda@xx1.171.xx.x:5060' in 6400 ms (Method: INVITE)
[Jun 12 09:29:06]
[Jun 12 09:29:06] <--- SIP read from WS:xx1.171.xx.x:60751 --->
[Jun 12 09:29:06] SIP/2.0 200 OK
[Jun 12 09:29:06] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK3ce6f173;rport
[Jun 12 09:29:06] From: "ACagcW15919685352222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as2e59afb1
[Jun 12 09:29:06] To: <sip:18s70scg@192.0.2.218;transport=wss>;tag=ep91a2imar
[Jun 12 09:29:06] CSeq: 102 CANCEL
[Jun 12 09:29:06] Call-ID: 5a1745f518bc2fca26ac8cf21c854eda@xx1.171.xx.x:5060
[Jun 12 09:29:06] Supported: outbound
[Jun 12 09:29:06] User-Agent: VICIphone 2.0
[Jun 12 09:29:06] Content-Length: 0
[Jun 12 09:29:06]
[Jun 12 09:29:06] <------------->
[Jun 12 09:29:06] --- (9 headers 0 lines) ---
[Jun 12 09:29:06]
[Jun 12 09:29:06] <--- SIP read from WS:xx1.171.xx.x:60751 --->
[Jun 12 09:29:06] SIP/2.0 487 Request Terminated
[Jun 12 09:29:06] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK3ce6f173;rport
[Jun 12 09:29:06] From: "ACagcW15919685352222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as2e59afb1
[Jun 12 09:29:06] To: <sip:18s70scg@192.0.2.218;transport=wss>;tag=n9g11ribc3
[Jun 12 09:29:06] CSeq: 102 INVITE
[Jun 12 09:29:06] Call-ID: 5a1745f518bc2fca26ac8cf21c854eda@xx1.171.xx.x:5060
[Jun 12 09:29:06] Supported: outbound
[Jun 12 09:29:06] User-Agent: VICIphone 2.0
[Jun 12 09:29:06] Content-Length: 0
[Jun 12 09:29:06]
[Jun 12 09:29:06] <------------->
[Jun 12 09:29:06] --- (9 headers 0 lines) ---
[Jun 12 09:29:06] Transmitting (NAT) to xx1.171.xx.x:60751:
[Jun 12 09:29:06] ACK sip:18s70scg@192.0.2.218;transport=wss SIP/2.0
[Jun 12 09:29:06] Via: SIP/2.0/WS xx1.171.xx.x:5060;branch=z9hG4bK3ce6f173;rport
[Jun 12 09:29:06] Max-Forwards: 70
[Jun 12 09:29:06] From: "ACagcW15919685352222222222222222" <sip:202xxx8707@xx1.171.xx.x>;tag=as2e59afb1
[Jun 12 09:29:06] To: <sip:18s70scg@192.0.2.218;transport=wss>;tag=n9g11ribc3
[Jun 12 09:29:06] Contact: <sip:202xxx8707@xx1.171.xx.x:5060;transport=ws>
[Jun 12 09:29:06] Call-ID: 5a1745f518bc2fca26ac8cf21c854eda@xx1.171.xx.x:5060
[Jun 12 09:29:06] CSeq: 102 ACK
[Jun 12 09:29:06] User-Agent: Asterisk PBX 13.32.0-vici
[Jun 12 09:29:06] Content-Length: 0
[Jun 12 09:29:06]
[Jun 12 09:29:06]
[Jun 12 09:29:06] ---
[Jun 12 09:29:06] Scheduling destruction of SIP dialog '5a1745f518bc2fca26ac8cf21c854eda@xx1.171.xx.x:5060' in 6400 ms (Method: INVITE)
[Jun 12 09:29:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 12 09:29:06]     -- Called 55558600051@default
[Jun 12 09:29:06]     -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000002;2", "8600051,K") in new stack
[Jun 12 09:29:06] WARNING[8760][C-00000002]: app_meetme.c:5261 admin_exec: Conference number '8600051' not found!
[Jun 12 09:29:06]     -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000002;2", "") in new stack
[Jun 12 09:29:06]   == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000002;2'
[Jun 12 09:29:06] WARNING[8760][C-00000002]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Jun 12 09:29:06]     -- Executing [h@default:1] AGI("Local/55558600051@default-00000002;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Jun 12 09:29:06]     -- <Local/55558600051@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Jun 12 09:29:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 12 09:29:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 12 09:29:07]   == Manager 'sendcron' logged off from 127.0.0.1
cyburity*CLI> exit
[Jun 12 09:29:09] Asterisk cleanly ending (0).