vicidial.org

Posted: **Sun Jan 12, 2020 11:02 am**

hi friends

i have a fresh install of vicibox 9.01 cluster. versión: VERSION: 2.14-730a
BUILD: 191121-2256

I performed the procedure for the installation for the webrtc viciphone and it works very well, except something ...

after 10 minutes of connecting to the agent the asterisk call hangs up and is left in registered state

Look at the traces sip .. and I notice that when disconnected the error appears:

481 Subscription does not exist

I think this is due to a timer called:

registerExpires: 600

I am behind a static nat

Has anyone had something similar?

Posted: **Thu Jan 16, 2020 11:12 pm**

Posted: **Fri Jan 17, 2020 4:07 am**

Maybe your SIP config is trying to re-invite after 10 minutes? Have you tried with canreinvite=no and directmedia=no inside your SIP template?

Posted: **Fri Jan 17, 2020 11:35 am**

Try zypper -ref and zypper up to see if there are any updates with asterisk 13.27

Posted: **Fri Jan 17, 2020 11:35 pm**

hi...

yes canreinvite and directmedia = no ...but no working in my sip template

i realized zypper ref and up

i upgrade the last vici trunk too

Posted: **Mon Jan 27, 2020 4:44 pm**

does your firewall have an internal port restriction ? if it does try opening internal ports 25000 - 65000 on udp. also make sure that internal tcp 8089 port is open to all subnets internally.

Posted: **Mon Jan 27, 2020 11:32 pm**

i dont have firewall in my vicidial cluster

Posted: **Sun Feb 09, 2020 12:13 am**

I managed to solve the problem temporarily.

edit the sip.js file and change the value of the register expires to a value of 600 to 36000 and the session vicidial call is no longer cut

but I don't know if this is a bad thing

what do you think?

Posted: **Fri Feb 14, 2020 11:15 pm**

I tried the PBXWebPhone webphone and it works without problem I don't have call cuts from the agent session

but i want viciphone

someone can help me?

Posted: **Wed Feb 26, 2020 5:38 am**

Keep using PBXWebPhone

Posted: **Wed Feb 26, 2020 12:05 pm**

cvillarreal77 wrote:I tried the PBXWebPhone webphone and it works without problem I don't have call cuts from the agent session

but i want viciphone

someone can help me?

We have zero information about your system, what you tried, and what failed. Thus my advice is ...

chornyi_taras wrote:Keep using PBXWebPhone

Posted: **Wed Feb 26, 2020 11:56 pm**

Hello William ..

what I am trying to do is implement the famous viciphone ..

I already performed all the steps and I think the viciphone has a bug ...

1.-log in with phone login

2.-Log in with user login

3.- then the green screen "No one is in your session: 8600051
Go back , Call Agent Webphone -> "and the viciphone appears with the registered legend

4.- then click on call agent webphone "

5.- then the viciphone receives the call of the session and the audio "you are only person in the conference" is heard and the status of the call changes to "incall didxxx" show in the viciphone status

6.- up to here everything well, i can receive and make calls, but after 10 exact minutes the status of the call "incall did" change a "registered"

I thought the call was being cut but it is not like that, when i check with the "sip show channels" command ,i can see that the call is alive ... only that the viciphone only changed the status to registered after 10 minutes

Is it normal for the status of the viciphone to be changed after 10 minutes?

Posted: **Thu Feb 27, 2020 9:27 am**

cvillarreal77 wrote:6.- up to here everything well, i can receive and make calls, but after 10 exact minutes the status of the call "incall did" change a "registered"

I'll need details of this. Since this is apparently your problem, this one tiny entry needs expansion. Where do you see this message/value/status? Are there any asterisk CLI message that coincide with this? Is it precisely 10 minutes and reliable? You may have a firewall timeout of some sort.

Posted: **Thu Feb 27, 2020 6:24 pm**

first we login in the page with phone login and user login:

then ... the status of viciphone is registered,
From here we start counting the 10 minutes

then we pick in the link " call agent webphone"

an the status of viciphone change :

....

and when the 10 minutes end returns to registered , this are exacts

and in the cli asterisk with "sip set debug " we have when the time ends:

then

install the webphone on the server and modify the file sip.js the parameter "registerExpires: 600" and increase the value to another 600 seconds "

and the 10 minute count changed to 20 minutes too

do you need to see all the cli sip set debug text ?

Posted: **Thu Feb 27, 2020 6:42 pm**

Excellent post: Now for the million dollar question: Does this adversely affect the phone call in any way?

Posted: **Thu Feb 27, 2020 7:30 pm**

no, this dont affect , ...... but I worry ... I'll just put it into production

do you know this is it normal ?
In the future can I have problems?

Do you also see this effect in your implementations?

Posted: **Fri Feb 28, 2020 9:08 am**

As I understand you are complaining that viciphone does not display call status ( in-call) after a reregistering timeout but an agent is still connected. if yes I would say this is a simple UI issue that can be ignored.

PS
I've workaround this issue in pbxwebphone by introducing 2 states( phone state and call state)

Posted: **Fri Feb 28, 2020 1:26 pm**

yes, chorny , it is what is happening ..

after a reregistration the error appears ...

exist a new version of pbxwebphone? or the actual versión already has phone state and call state corrections?

Posted: **Fri Feb 28, 2020 1:44 pm**

chornyi_taras wrote:As I understand you are complaining that viciphone does not display call status ( in-call) after a reregistering timeout but an agent is still connected. if yes I would say this is a simple UI issue that can be ignored.

PS
I've workaround this issue in pbxwebphone by introducing 2 states( phone state and call state)

You should switch the "phone state" (which I assume is the registration on/off indication) to a wifi symbol/icon with a tool tip spelling out the registration status.

Posted: **Thu Jun 11, 2020 11:27 am**

When I am using PBXwebphone I am getting this option only
Ready
Terminated
Mute Button

But when I am using Viciphone I am able to dial out but as soon as call answered it disconnects

any idea?

Posted: **Thu Jun 11, 2020 11:29 am**

virtualsky:

can you paste your asterisk CLI output as you login and attempt to make a call

Posted: **Thu Jun 11, 2020 11:46 am**

Hi,
I have captured sip trace and asterisk CLI too

SIP TRACE
===============
<--- SIP read from WS:49.X.X.X:52767 --->
ACK sip:1234567890@1.1.1.177:50601;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.117;branch=z9hG4bK2304122
Max-Forwards: 70
To: <sip:1234567890@1.1.1.177>;tag=as2850c18f
From: "110011" <sip:110011@1.1.1.177>;tag=9o1o6ltgud
Call-ID: tvdfns1k8o4ger26ph67
CSeq: 1836 ACK
Supported: outbound
User-Agent: VICIphone 1.0-rc1
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from WS:49.X.X.X:52767 --->
BYE sip:1234567890@1.1.1.177:50601;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.117;branch=z9hG4bK4567744
Max-Forwards: 70
To: <sip:1234567890@1.1.1.177>;tag=as2850c18f
From: "110011" <sip:110011@1.1.1.177>;tag=9o1o6ltgud
Call-ID: tvdfns1k8o4ger26ph67
CSeq: 1837 BYE
Reason: SIP ;cause=488 ;text="Not Acceptable Here"
Supported: outbound
User-Agent: VICIphone 1.0-rc1
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Scheduling destruction of SIP dialog 'tvdfns1k8o4ger26ph67' in 27648 ms (Method: BYE)

<--- Transmitting (no NAT) to 49.X.X.X:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.117;branch=z9hG4bK4567744;received=49.X.X.X
From: "110011" <sip:110011@1.1.1.177>;tag=9o1o6ltgud
To: <sip:1234567890@1.1.1.177>;tag=as2850c18f
Call-ID: tvdfns1k8o4ger26ph67
CSeq: 1837 BYE
Server: Asterisk PBX 13.21.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
-- Channel SIP/110011-00000027 left 'simple_bridge' basic-bridge <126418e2-54b9-42d8-93ed-b88de7b69500>
-- Channel SIP/SIPAC-00000028 left 'simple_bridge' basic-bridge <126418e2-54b9-42d8-93ed-b88de7b69500>
== Spawn extension (default, 1234567890, 2) exited non-zero on 'SIP/110011-00000027'
-- Executing [h@default:1] AGI("SIP/110011-00000027", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----2-----0-----SIP 200 OK)") in new stack

==================
Asterisk CLI STARTS BELOW
===================

- Executing [1234567890@default:1] AGI("SIP/110011-00000024", "agi://127.0.0.1:4577/call_log") in new stack
-- <SIP/110011-00000024>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [1234567890@default:2] Dial("SIP/110011-00000024", "SIP/SIPAC/1234567890,,To") in new stack
[Jun 11 16:37:26] ERROR[30939][C-00000074]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("1.1.1.177.static.coresite.com", "(null)", ...): Name or service not known
[Jun 11 16:37:26] WARNING[30939][C-00000074]: acl.c:835 resolve_first: Unable to lookup '1.1.1.177.domain.com'
== Using SIP RTP CoS mark 5
-- Called SIP/SIPAC/1234567890
> 0x7f389800ced0 -- Strict RTP learning after remote address set to: 1.2.2.2:43860
-- SIP/SIPAC-00000025 is making progress passing it to SIP/110011-00000024
> 0x7f389800ced0 -- Strict RTP switching to RTP target address 1.2.2.2:43860 as source
-- SIP/SIPAC-00000025 answered SIP/110011-00000024
-- Channel SIP/SIPAC-00000025 joined 'simple_bridge' basic-bridge <a9771679-2306-4ba9-ac13-d529b4f6acf6>
-- Channel SIP/110011-00000024 joined 'simple_bridge' basic-bridge <a9771679-2306-4ba9-ac13-d529b4f6acf6>
-- Channel SIP/110011-00000024 left 'simple_bridge' basic-bridge <a9771679-2306-4ba9-ac13-d529b4f6acf6>
-- Channel SIP/SIPAC-00000025 left 'simple_bridge' basic-bridge <a9771679-2306-4ba9-ac13-d529b4f6acf6>
== Spawn extension (default, 1234567890, 2) exited non-zero on 'SIP/110011-00000024'
-- Executing [h@default:1] AGI("SIP/110011-00000024", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----3-----0-----SIP 200 OK)") in new stack
-- <SIP/110011-00000024>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -0-----SIP 200 OK) completed, returning 0

What is the reason of Not Acceptable Here ...is it codec issue?

Posted: **Thu Jun 11, 2020 11:55 am**

its not staying in the vicidial conference, i needed to see after the last line of cli output to see exactly what happened but can you make calls via zoiper without a problem?

Posted: **Thu Jun 11, 2020 12:00 pm**

Zoiper ..do u mean zoiper webphone? no i didnt try as it require plugin but I am able to dial from agent screen via manual dial..and when I try from xlite it says you r only person in the conf..

Posted: **Thu Jun 11, 2020 12:05 pm**

ok so xlite is working, i meant zoiper softphone but if xlite is working then its something with your setup, how does your viciphone template look? you have your ssl cert in there?

Posted: **Thu Jun 11, 2020 12:12 pm**

Yes I have SSL cert and I am using Viciphone and I tried this one too https://phone.viciphone.com/viciphone.php and still same then I came to know something issue in my setup but couldnt figure it out so I am debugging more and found that below issues ..may be it will help you to understand the exact issue

======================================================================

== WebSocket connection from '2.2.2.2:53845' for protocol 'sip' accepted using version '13'
2.*CLI>
[Jun 11 17:04:47] ERROR[970]: tcptls.c:447 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Broken pipe
== WebSocket connection from '2.2.2.2:53664' forcefully closed due to fatal write error
===================================
And I am logging out of agent screen then it shows me below error conf not founf

===================================================

r 'sendcron' logged on from 127.0.0.1
-- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000011;2", "8600051,K") in new stack
-- Called 55558600051@default
[Jun 11 17:03:40] WARNING[875][C-00000078]: app_meetme.c:5253 admin_exec: Conference number '8600051' not found!
-- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000011;2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000011;2'
[Jun 11 17:03:40] WARNING[875][C-00000078]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@default:1] AGI("Local/55558600051@default-00000011;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
-- <Local/55558600051@default-00000011;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0

Posted: **Thu Jun 11, 2020 12:26 pm**

I actually get this same error on the server i just got that uses viciphone. I am awaiting a response from the programmer to see what he thinks for that error. I will keep you updated.

Posted: **Thu Jun 11, 2020 12:30 pm**

carpenox wrote:I actually get this same error on the server i just got that uses viciphone. I am awaiting a response from the programmer to see what he thinks for that error. I will keep you updated.

But is your setup working fine?

Posted: **Thu Jun 11, 2020 12:33 pm**

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.

Posted: **Thu Jun 11, 2020 12:37 pm**

williamconley wrote:
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.

FYI as I mentioned above -> I am logging out of agent screen then it shows me below error conf not found which is normal in that case..

Posted: **Thu Jun 11, 2020 12:41 pm**

virtualsky wrote:
williamconley wrote:
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.

FYI as I mentioned above -> I am logging out of agent screen then it shows me below error conf not found which is normal in that case..

Hi William, sorry I think i misunderstood your reply...so what can be the issue.. and I am using asterisk 13.21 on centos 6.10

Posted: **Thu Jun 11, 2020 12:42 pm**

wshat is the first error in the output?

Posted: **Thu Jun 11, 2020 12:46 pm**

williamconley wrote:wshat is the first error in the output?

There is no error as soon as I dial from webphone, it answered on the CLI and disconnects immediately below are the complete logs

======================================================================================================================
= DTLS ECDH initialized (secp256r1), faster PFS enabled
[Jun 11 17:02:00] ERROR[526][C-00000077]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("1.1.1.177.static.domain.com", "(null)", ...): Name or service not known
[Jun 11 17:02:00] WARNING[526][C-00000077]: acl.c:835 resolve_first: Unable to lookup '1.1.1.177.static.domain.com'
== Using SIP RTP CoS mark 5
-- Executing [73991234567890@default:1] AGI("SIP/110011-0000002c", "agi://127.0.0.1:4577/call_log") in new stack
-- <SIP/110011-0000002c>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [73991234567890@default:2] Dial("SIP/110011-0000002c", "SIP/SIPAC/1234567890,,To") in new stack
[Jun 11 17:02:00] ERROR[702][C-00000077]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("1.1.1.177.static.domain.com", "(null)", ...): Name or service not known
[Jun 11 17:02:00] WARNING[702][C-00000077]: acl.c:835 resolve_first: Unable to lookup '1.1.1.177.static.domain.com'
== Using SIP RTP CoS mark 5
-- Called SIP/SIPAC/1234567890

== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/SIPAC-0000002d is making progress passing it to SIP/110011-0000002c
-- SIP/SIPAC-0000002d answered SIP/110011-0000002c
-- Channel SIP/SIPAC-0000002d joined 'simple_bridge' basic-bridge <03b8c17f-9379-44ec-892c-2e8b34807cdc>
-- Channel SIP/110011-0000002c joined 'simple_bridge' basic-bridge <03b8c17f-9379-44ec-892c-2e8b34807cdc>
-- Channel SIP/110011-0000002c left 'simple_bridge' basic-bridge <03b8c17f-9379-44ec-892c-2e8b34807cdc>
-- Channel SIP/SIPAC-0000002d left 'simple_bridge' basic-bridge <03b8c17f-9379-44ec-892c-2e8b34807cdc>
== Spawn extension (default, 73991234567890, 2) exited non-zero on 'SIP/110011-0000002c'
-- Executing [h@default:1] AGI("SIP/110011-0000002c", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----1-----SIP 200 OK)") in new stack
-- <SIP/110011-0000002c>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -1-----SIP 200 OK) completed, returning 0
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

Posted: **Thu Jun 11, 2020 1:03 pm**

viewtopic.php?f=8&t=38919

Posted: **Thu Jun 11, 2020 1:15 pm**

williamconley wrote:http://www.vicidial.org/VICIDIALforum/viewtopic.php?f=8&t=38919

Ok let me go through this meanwhile I debug the webphone log it says
BYE Reason: SIP ;cause=488 ;text="Not Acceptable Here"

Posted: **Thu Jun 11, 2020 11:31 pm**

williamconley wrote:http://www.vicidial.org/VICIDIALforum/viewtopic.php?f=8&t=38919

HI William, I have gone through the doc and configured FQDN properly but the real issue I guess is that it is not coming into the meet me conference

Posted: **Fri Jun 12, 2020 8:48 am**

Yes I am having the same issue and here is the CLI output during the webphone user logging into vici:

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).

Posted: **Fri Jun 12, 2020 12:04 pm**

that's the cancel. what happened before that cancel, which would presumably be the cause for the cancel.

Posted: **Thu Jun 18, 2020 12:31 am**

once ccabrera fixed this bug, it has been working now. Thx Christian

Posted: **Tue Jun 23, 2020 12:47 am**

carpenox wrote:once ccabrera fixed this bug, it has been working now. Thx Christian

How did u fix the issue?

vicidial.org

webrtc problems

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