serfreeman1337 / asterlink Goto Github PK

PBX Version:16.0.40.4
PBX Distro:12.7.8-2306-1.sng7
Asterisk Version:18.19.0
SuiteCRM 8.4.0

When installing the module in a clean SuiteCRM, the module works fine.
I am making changes in my SuiteCRM and since then I get an EOF error in the asterlink module on FreePBX.
What could this be related to?

Deleting /extensions/asterlink, /cache and even completely reinstalling the module does not give results

First of all: The piece of work you did is awesome!!!

I'm using your software with asterisk 18 and AMI 7
The communication between SuiteCRM and asterlink is working, incomming and outgoing calls are beeing loged.

The problem begins with the updating of the created record. Here is the log:

msg="New incoming call" lid=1633609721.46
msg="map[asterlink_call_seconds_c:0 asterlink_cid_c:+1234567890 asterlink_did_c:987654321 asterlink_uid_c:1633609721.46 date_start:2021-10-07 12:28:41 direction:Inbound duration_hours:0 duration_minutes:0 name:+123456789 status:Planned]" method=POST suite=true url="http://URL/index.php?entryPoint=AsterLinkEntryPoint&action=create_call_record"
msg="201 Created" method=POST suite=true url="http://URL/index.php?entryPoint=AsterLinkEntryPoint&action=create_call_record"
msg="Call registred" id=a2d7359d-8d6d-7586-6e4b-615ee7708673 lid=1633609721.46 suite=true
msg=Answer ext=21 lid=1633609721.46
msg="map[data:map[assigned_user_id:1] id:a2d7359d-8d6d-7586-6e4b-615ee7708673]" method=POST suite=true url="http://URL/index.php?entryPoint=AsterLinkEntryPoint&action=update_call_record"
msg=Answer ext=21 lid=1633609721.46
msg="Call finished" lid=1633609721.46
msg="204 No Content" method=POST suite=true url="http://URL/index.php?entryPoint=AsterLinkEntryPoint&action=update_call_record"
msg="map[data:map[assigned_user_id:1] id:a2d7359d-8d6d-7586-6e4b-615ee7708673]" method=POST suite=true url="http://URL/index.php?entryPoint=AsterLinkEntryPoint&action=update_call_record"
msg="204 No Content" method=POST suite=true url="http://URL/index.php?entryPoint=AsterLinkEntryPoint&action=update_call_record"
msg="map[data:map[asterlink_call_seconds_c:0 date_end:2021-10-07 12:28:51 duration_hours:0 duration_minutes:0 status:Held] id:a2d7359d-8d6d-7586-6e4b-615ee7708673]" method=POST suite=true url="http://URL/index.php?entryPoint=AsterLinkEntryPoint&action=update_call_record"
msg="204 No Content" method=POST suite=true url="http://URL/index.php?entryPoint=AsterLinkEntryPoint&action=update_call_record"

Do you know how to get rid of this problem?

I'm trying to build from scratch but I still get this error: "../../sirupsen/logrus/terminal_check_unix.go:6:8: cannot finad package "golang.org/x/sys/unix" and "go get golang.org/x/sys/unix" fails with error: "no go-import meta tags"

Hi, I'm using the lastest source, outbound call is success on bitrix but call status on bitrix24 call details is always "The call was skipped".

Even when the call is successful and is answered by softphone, the lead is titled as "Пропущенный звонок" which means "Missed call". However, the employee in Bitrix24 is detected properly.

The readmes is are not really clear how this supposed to work if you are quite new. Currently I am using the odoo crap and that connects directly to asterisk for similar functionality (although as everything there, buggy). So I do not really get why it is necessary to have a process running, or is the zip file enough?

In asterisk ami there is also no empty space.

When you get free, can you check asterlink and if its compatible with suitecrm 7.12x+php73? Inbound popups are not working, the websocket connects, calls are being logged in the call module, but the popupcard hook doesnt seem to be triggered.. I even tried the previous old asterlink2 installation method which worked on php7.2/suitecrm 7.11.x and still cant get popup card to fire on 7.12.x/php73 ... Something else i found was the callerid/did were showing as the same when making outbound calls, i had to edit the ami_ago file and recompile for it to detect the callerid vs extension on asterisk 13x

I have currently a partial working setup, that is logging calls in the activies/calls section of suitecrm. What annoyed me is that if I am testing I have to wait a minute before time is registered. At first I thought when I saw the log entry below, seconds were just not submitted, but then I noticed that de database has duration stored in two separate fields (wtf ??? wtf ??? wtf ???)

When they change it to one field using something like seconds or interval, it would be nice to update this module for it.

time="2022-06-26T19:44:14+02:00" level=trace msg="map[data:map[asterlink_call_seconds_c:17 date_end:2022-06-26 17:44:14 duration_hours:0 duration_minutes:1 status:]

Later I will link the issue to request this change in the suitecrm database here.

During an incoming call, if I click on the call on the operator panel screen in Bitrix24 just after answered it via softphone, call go on without any problem. But if I just leave it there and answer it on the softphone, the call is dropped in random time.

The first thing I noticed is that the date_end is set to the date_start, when the record is being created. I do not think this is a correct way to enter data into database. It does not make sense setting by default an incorrect value. Just leave the field null.

date_end seems to be not included in the update of the call record

Thanks in advance.

Site is not loaded. white screen is displayed. Console display is pasted below.

https://crm.v4vs.com/dist/extensions/asterlink/remoteEntry.js file is missing

README does not describe how to install Asterlink on FreePBX. It is written in GoLang but FreePBX is based on PHP.

Hello.
I have installed the module and all works as intended. Calls are being logged on suitecrm and pop up window is popping with caller id information but I cannot match caller id phone number with contact, the phone number is being showed correctly on the logs and on the little window but nor the logs or the pop up window show contact information. I have tried different relationships with the modules and still no luck.

Any advise to make this work?

Thanks In advance
Luis

FW Console - FreePBX Utility 16.0.21.18
Asterisk 19.5.0 built by mockbuild @ jenkins7 on a x86_64 running Linux on 2022-07-07 06:25:53 UTC
Server IP:172.16.34.21

SuiteCRM Version 7.12.7
Sugar Version 6.5.25 (Build 344)
Server IP:172.16.34.23

When I used the command ./Asterlink I am getting the following message error:

time="2022-10-16T14:40:02+02:00" level=info msg=AsterLink
time="2022-10-16T14:40:02+02:00" level=info msg="Setting log level" fields.level=trace
time="2022-10-16T14:40:02+02:00" level=info msg="Using SuiteCRM Connector"
time="2022-10-16T14:40:02+02:00" level=trace method=GET suite=true url="http://172.16.34.23/index.php?entryPoint=AsterLinkEntryPoint&action=get_ext_users"
time="2022-10-16T14:40:02+02:00" level=trace msg="200 OK" method=GET suite=true url="http://172.16.34.23/index.php?entryPoint=AsterLinkEntryPoint&action=get_ext_users"
time="2022-10-16T14:40:02+02:00" level=info msg="User list updated" suite=true users="map[2002:1]"
time="2022-10-16T14:40:02+02:00" level=info msg="Enabling web server" addr="172.16.34.23:5678" suite=true
**time="2022-10-16T14:40:02+02:00" level=fatal msg="listen tcp 172.16.34.23:5678: bind: cannot assign requested address" suite=true**

Hi! I'm kind of confused on how to filter specific numbers for incoming calls. I get notifications on every incoming call that is being registered in my incoming context, but I want to be able to limit the numbers that are actually processed by Asterlink. Is there a way to make it work? I've seen the regexp example and tried it a little, but with no success. Thanks in advance!

Also I'm facing another issue right now and also I don't even know how to debug it, because I don't get log entries while performing click to call from Bitrix24 UI. The point is that when I manually perform outgoing call by typing a number I get the record a call uploaded but I'm not able to call from the Bitrix card straight.
Can you please give me some hints on resolving that issues? Thanks!

Maybe at the utils section[1] option to enable/disable debug logging on the suitecrm module

[1]
/index.php?module=AsterLink&action=config