VOIP not showing incoming/outgoing call notifications

[khopper]

VOIP not showing incoming/outgoing calls notifications

I have enable this to the server my phone is connected to and connection was successful, I setup my user ID with my Ext and PW.
I setup the scheduled task to run every minute as instructed. I am not getting any notifications! What could be wrong?

[tarasm]

Did you follow this instruction https://www.espocrm.com/features/ast...ration-setup/?
In some cases, the Asterisk server sends to EspoCRM a user phone number instead of user Ext. Try to add additional phone numbers for your user https://www.espocrm.com/features/ast...-phone-numbers.

[khopper]

Yes, I followed the setup instructions.
My Ext has no other numbers associated to it, IB or OB.
This is however running on a windows server using the "Task Scheduler" and set to run every minute.
Has this been extension been tested by development to work on a Microsoft platform or is it specific to Linux?

[tarasm]

Linux server is preferred, but for Windows it should work fine as well.
To find a problem:
1. Enable a debug mode https://www.espocrm.com/documentatio...debugging-mode.
2. Make test calls
3. Check EspoCRM log file https://www.espocrm.com/documentatio...crm-error-logs. Here should be some information about call events.

[khopper]

I just installed VoIP Integration 1.9.1 and configured and connection show successful - however the i am not getting any notifications or pop ups.
The Cron job is scheduled but I do not see any logs in espo regarding the voip at all.

[tarasm]

Please let me know more details about your telephony server?

[khopper]

Asterisk 1.8 on a Vicidial server. I also confirmed my user account in the CRM is configured for it also.
VERSION: 2.12-565a
BUILD: 160827-0917

[tarasm]

Did you have configure this cron job (the correct command is inside Asterisk connector):

* * * * * /usr/bin/php -f /var/www/espocrm/voip.php Asterisk > /dev/null 2>&1

[khopper]

This is my code:

* * * * * cd /var/www/espocrm; /usr/bin/php -f cron.php > /dev/null 2>&1
* * * * * cd /var/www/espocrm; /usr/bin/php -f voip.php Asterisk > /dev/null 2>&1

[tarasm]

Yes, that's fine. Please debug a problem:
1. Enable a debug mode, https://www.espocrm.com/documentatio...e-for-a-logger.
2. Make test calls.
3. Check EspoCRM log file https://www.espocrm.com/documentatio...crm-error-logs. Here should be some information about call events or errors.

[khopper]

DEBUG Mode is already enabled and is always on in my environment.

I currently see this in the new log before attempting anything. I renamed the old log file and restarted the apache2 services after renaming so espo would generate a new file.
Anything involving VOIP does not appear to be creating any log files even in DEBUG mode.
See attached GIF.

My support team says they see it connecting when preforming the "Test Connection" but nothing else.

[2019-01-26 08:30:10] Espo.ERROR: Uncaught Exception PAMI\Client\Exception\ClientException: "Error reading" at /var/www/espocrm/application/Espo/Modules/Voip/Providers/Asterisk/Ami/vendor/marcelog/pami/src/PAMI/Client/Impl/ClientImpl.php line 238 {"exception":"[object] (PAMI\\Client\\Exception\\ClientException(code: 0): Error reading at /var/www/espocrm/application/Espo/Modules/Voip/Providers/Asterisk/Ami/vendor/marcelog/pami/src/PAMI/Client/Impl/ClientImpl.php:238)"} []
[2019-01-26 09:30:10] Espo.ERROR: Uncaught Exception PAMI\Client\Exception\ClientException: "Error reading" at /var/www/espocrm/application/Espo/Modules/Voip/Providers/Asterisk/Ami/vendor/marcelog/pami/src/PAMI/Client/Impl/ClientImpl.php line 238 {"exception":"[object] (PAMI\\Client\\Exception\\ClientException(code: 0): Error reading at /var/www/espocrm/application/Espo/Modules/Voip/Providers/Asterisk/Ami/vendor/marcelog/pami/src/PAMI/Client/Impl/ClientImpl.php:238)"} []

[tarasm]

Your Asterisk is closing the connection, you should check the Asterisk console for more information.

[khopper]

According to the AMI logs i do not see espo passing the Dialstring or Peername properly if at all.



Event: Newchannel
Privilege: call,all
Channel: SIP/2005-00001954
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName: Kim Hopper
AccountCode: 2005
Exten:
Context: default_wr
Uniqueid: 1565447825.21106

Event: ChannelUpdate
Privilege: system,all
Channel: SIP/2005-00001954
Uniqueid: 1565447825.21106
Channeltype: SIP
SIPcallid: 0c388836537745bf60b3f710397df9f2@192.168.5.249:506 0
SIPfullcontact: sip:2005@192.168.4.99:53200;rinstance=bb704ae0a8cc d16b

Event: ChannelUpdate
Privilege: system,all
Channel: SIP/2005-00001954
Channeltype: SIP
SIPcallid: 0c388836537745bf60b3f710397df9f2@192.168.5.249:506 0
SIPfullcontact: sip:2005@192.168.4.99:53200;rinstance=bb704ae0a8cc d16b
Peername: 2005

Event: NewAccountCode
Privilege: call,all
Channel: SIP/2005-00001954
Uniqueid: 1565447825.21106
AccountCode:
OldAccountCode: 2005

Event: NewCallerid
Privilege: call,all
Channel: SIP/2005-00001954
CallerIDNum: 2005
CallerIDName: Kim Hopper
Uniqueid: 1565447825.21106
CID-CallingPres: 0 (Presentation Allowed, Not Screened)

Event: Newstate
Privilege: call,all
Channel: SIP/2005-00001954
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum: 2005
CallerIDName: Kim Hopper
ConnectedLineNum: 2005
ConnectedLineName:
Uniqueid: 1565447825.21106

Event: ChannelUpdate
Privilege: system,all
Channel: SIP/2005-00001954
Channeltype: SIP
Uniqueid: 1565447825.21106
SIPcallid: 0c388836537745bf60b3f710397df9f2@192.168.5.249:506 0
SIPfullcontact: sip:2005@192.168.4.99:53200;rinstance=bb704ae0a8cc d16b
Peername: 2005

Event: Newstate
Privilege: call,all
Channel: SIP/2005-00001954
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 2005
CallerIDName: Kim Hopper
ConnectedLineNum: 2005
ConnectedLineName:
Uniqueid: 1565447825.21106

Event: Hangup
Privilege: call,all
Channel: SIP/2005-00001954
Uniqueid: 1565447825.21106
CallerIDNum: 2005
CallerIDName: Kim Hopper
ConnectedLineNum: 2005
ConnectedLineName: <unknown>
Cause: 16
Cause-txt: Normal Clearing

[tarasm]

Kim,
Thanks for detailed information. I do not see any errors based on your log information. These may cause the problems:
1. Check if "Dial Context" option is correct. This option should be set as defined in your Asterisk configuration. This option is very important for outgoing calls.
2. I see that you have 4 additional phone numbers in your user profile. And you are trying to make a call from one your phone number to another. It may cause a problem with displaying a call popup. Try to leave only your phone number, other phone numbers you can save in contacts for example.
After these changes, please try to make new tests.