Transfer Calls not working after FreePBX upgrade
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Hi Svan,
Try editing /usr/local/fop2/autoconfig-users.sh and remove the caret from the AMPDBNAME, AMPDBUSER and AMPDBPASS variables, so it looks like this (at the top of the file).
if [ -e /etc/freepbx.conf ]; then DBNAME=`cat /etc/freepbx.conf | sed 's/ //g' | grep AMPDBNAME | cut -d= -f2 | tail -n1` DBUSER=`cat /etc/freepbx.conf | sed 's/ //g' | grep AMPDBUSER | cut -d= -f2 | tail -n1` DBPASSLINE=`cat /etc/freepbx.conf | grep AMPDBPASS | tail -n1`Then do the same in autoconfig-buttons.sh
To test it out, run the script in the command line and inspect the ouptut
[fixed]
/usr/local/fop2/autoconfig-users.sh
[/fixed]Best regards,
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The autoconfig-users runs fine, but the autoconfig-buttons doesn't:
./autoconfig-buttons.sh # Problem connecting to mysql ! Cannot connect to Fo2 Manager databaseThis is what it looks like now:
if [ -e /etc/freepbx.conf ]; then DBNAME=`cat /etc/freepbx.conf | sed 's/ //g' | grep AMPDBNAME | cut -d= -f2 | tail -n1` DBUSER=`cat /etc/freepbx.conf | sed 's/ //g' | grep AMPDBUSER | cut -d= -f2 | tail -n1` DBPASSLINE=`cat /etc/freepbx.conf | grep AMPDBPASS | tail -n1` DBSTRIP=`echo $DBPASSLINE | cut -d= -f1` DBPASS=`echo $DBPASSLINE | sed "s/$DBSTRIP=//g"` DBHOST=`cat /etc/freepbx.conf | sed 's/ //g' | grep ^AMPDBHOST | cut -d= -f2 | tail -n1` -
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All the same, there is no sound. Output for asterisk -rx "core show channels concise" with and without force_parameter_delimiter=','
DAHDI/pseudo-922601214!default!s!1!Rsrvd!(None)!!!!!3!47!!1418758260.206057 DAHDI/i2/889658064166-31d!from-internal!STARTMEETME!4!Up!MeetMe!201,oTMr,!89658064166!!!3!30!!1418758277.206075 DAHDI/pseudo-2103507193!default!s!1!Rsrvd!(None)!!!!!3!47!!1418758260.206059 DAHDI/i2/88162280360-31c!from-internal!STARTMEETME!4!Up!MeetMe!201,oTMr,!88162280360!1418758252.206045!1418758252.206045!3!56!!1418758252.206045 SIP/1002-00000317!from-internal-increase-vol!!1!Up!ChanSpy!CONFERENCE/201,dq!1002!!!3!12!!1418758295.206085 -
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I am not sure if I understand correctly the "flow" of the problem. Not sure who are you transferring nor how into the conference?
A line being busy after a transfer is made, and remaining busy until the transferred call finishes, is a classic asterisk "feature" when using Local/xxxx@yyyy/n channels. It is not a FOP2 problem, is a byproduct of /n in a Local channel inside Asterisk, and as such, cannot be fixed within FOP2, nor it is a FOP2 bug.
However, I am not sure if your description of the problem has to do with that or not.
Best regards,
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Incomming call (from dahdi) -> direct to SIP extension -> blind transfer to conference = all ok - external call in conference and SIP extension line released.
Incomming call (from dahdi) -> queue -> SIP extension -> blind transfer to conference = problem - external call in conference but SIP extension not released line.
Screen in attach.
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