Transfer Calls not working after FreePBX upgrade

mjsouth

My FreePBX server was a few versions behind, so I updated it. It all went very smooth, but now FOP won't transfer calls. I upgraded to 2.28, erased the management module from FreePBX. (love the web based config) It functions well, I feel like there is a disconnect between FOP and Asterisk that I can't put my finger on. The config files all have transfers turned on, as I have read older threads with similar problems.

I also upgraded Asterisk to version 13. Not sure if that caused some of the problems.

What can I do to debug this problem?

filippov

Same problem for me. Asterisk 12.6.1 FreePBX 12.0.13 Ubuntu 14.10 FOP 2.28. Not working transfer (blind and attended) and no sound (and not working mic) in 'listen&whisper' mode for conference (the button listen and 'listen&whisper' are working).

nicolas

FOP 2.29 will address some changes in Asterisk 12 and 13. It is available for download, but in beta.

FreerPBXer

You can roll back to Asterisk 11 by running asterisk-version-switch and choosing 11. I just did that after having this problem and the buttons started working.

filippov

OK, pls give me URL to FOP 2.29 Beta (Debian/Ubuntu x64). And what to do with the activation code.
I have the key to 2.27. Two weeks ago I bought a key for upgrade to 2.28 (on main page fop2.com claimed that supported Asterisk 12).
How to deal with the new version 2.29?

nicolas

http://download.fop2.com/fop2-2.29-debian-x86_64.tgz

Just download, extract and run make. The license will upgrade itself, you are free to upgrade for ONE year after purchase.

svan

Hi, I'm having the same issue: call transfer and pickup do not work with fop2 2.28 and FreePBX 12. However after upgrading to 2.29 beta (the CentOS version, I get the following error when running "/usr/local/fop2/fop2_server --test":

Flash Operator Panel 2 - White Label Version.
unable to initialize libusb: -99
Flash Operator Panel 2 - Valid License (7)

ERROR 1102 (42000) at line 1: Incorrect database name ''
ERROR 1102 (42000) at line 1: Incorrect database name ''
Connection to manager OK!

And no buttons display in fop2.
Any ideas?

nicolas

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,

svan

The autoconfig-users runs fine, but the autoconfig-buttons doesn't:

./autoconfig-buttons.sh
# Problem connecting to mysql

! Cannot connect to Fo2 Manager database

This 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`

nicolas

Can you send me privately your /etc/freepbx.conf file ? Or please contact me via the live help. I am online now.

filippov

Hello, transfer is now working, but no sound in listen/whisper mode. If cal to 555 (freepbx spy channel) + number extension - all good.

nicolas

asterisk -rx "core show channels concise"

When you are listening to a call, look for the chanspy session and the parameter delimiter, is it a comma or a pipe, if a pipe, in fop2.cfg set force_parameter_delimiter=','

filippov

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

nicolas

You cannot spy on conferences? Only extensions. If you want to hear a conference, just join it.

Best regards,

filippov

No sounds in spy mode (listen / whisper) only in conferences. Button listen / whisper work well with other internal numbers. If i call 555 + number conference sounds is good, but it is inconvenient for the user.

nicolas

You cannot spy on a conference in fop2, you can join the conference and listen in (same effect).

filippov

About spy in conference understand.
I have error with blind transfer from user in queues to conference - after transfer incoming call to conference, sip line of user form queues is not released. The line of user remains busy until incoming call not complet.

nicolas

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,

filippov

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.

nicolas

Your description fits almost exactly on the case I described above. How do your queue members look like? Are you using Local/xxx@yyy/n devices?

asterisk -rx "queue show"

Best regards,