How to fix click-to-call
If you don't want to go into details, contact us. We will quickly help you and answer all questions!
Contact us in any convenient way.
- 1.User makes a call from Bitrix24 (by clicking phone number in CRM).
- 2.Connector receives a command to make a call (originate).
- 3.Connector makes a request and sends it to Asterisk.
- 4.After that, the user's IP-phone or softphone will ring (phone with number which specified in the user profile).
- 5.When the user picks up the phone, the call goes to the customer's phone.
- 1.Whether the application is installed on the portal.
- 2.Whether Connector is running on the server.
- 3.Internal number in user's profile on the portal is specified and matches the one set up on the user's phone (SIP-phone).
- 4.“Asterisk-connector Itgrix (free)” must be selected as the default application for outgoing calls. How to check: on the portal, go to Telephony → Configure telephony → Telephony settings, field Default number for outgoing calls should contain “Application: Asterisk-connector Itgrix (free)”. For a specific user go to Telephony → Configure telephony → Telephony users. Setting up of this step is described in the instruction.
- 5.A handler must be installed in standalone (self-hosted) Bitrix24. Check if the code from the instruction has been added to the dbconn.php file:
if($_REQUEST['COMMAND'] === 'startCallViaRest' || $_SERVER['REQUEST_URI'] === '/rest/voximplant.call.startViaRest.json') { define('BITRIXREST_URL', '<Asterisk IP>:8077'); }
where<Asterisk IP>
is the address of your Asterisk server with the Connector installed on it. - 6.In the Asterisk console verify if that peer exists and it is online.
asterisk -rx ‘sip show peer 123’
where 123 is the internal phone number of the user. - 7.Verify that the request reaches the Connector. Search for event on ExternalCallStart (the last one) in the Connector log file
/var/log/itgrix_bx.log: grep "onExternalCallStart" /var/log/itgrix_bx.log --max-count=1
[2018-12-21 16:45:28.520 +05] debug Start onExternalCallStart event received from user [504] to phone [81234567890]
- 1.If nothing is found, then Connector has not received a call request. Then you need to check if the requests come to the server: tcpdump port 8077 After that, make a call with a click from Bitrix24.
- If the command output did not appear, then the problem may be as follows:
- Port is not forwarded (Connector behind NAT).
- Bitrix24 sends data to another address / port.
- Bitrix24 does not send a request.
- If there is an output, check firewall settings.
- 2.If such a log entry is found, check the customization result:
grep "Result of originate params customization" /var/log/itgrix_bx.log
[2018-12-21 16:41:18.202 +05] debug Result of originate params customization: QHash((caller_id, Звонок на 81234567890 <81234567890>)(extension, 81234567890)(priority, 1)(context, from-internal)(channel, SIP/123))
Try to execute originate with the received parameters directly in Asterisk (call from 123 to 81234567890):
# asterisk -rvvvvv
> channel originate SIP/123 extension
81234567890@from-internal
In case of success you will see:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 123
-- SIP/123-00001a5e is ringing
In case of failure you will see:- If the context is incorrect, you will see the message:
WARNING[9975][C-00004504]: pbx.c:4461 __ast_pbx_run: Channel 'SIP/123-00001a62' sent to invalid extension but no invalid handler: context,exten,priority=bad-context,81234567890,1
- If the number is missing:
[2018-12-21 17:47:52] ERROR[10555]: chan_pjsip.c:2458 request: Unable to create PJSIP channel - endpoint '123' was not found
- If the channel type is incorrect:
[2018-12-21 17:47:37] WARNING[10527]: channel.c:6083 request_channel: No channel type registered for 'PSSIP'
If this article does not solve the problem, please contact technical support.