FreePBX yumwrapper error

 I encountered the same thing after automatic module upgrades. I had two files located in: /dev/shm/yumwrapper/ I deleted those two files and things were happy again.

I don’t mean to offend, but I don’t know your level of Linux skill. Maybe this will help:
After logging into the server’s console (command line) enter the following two commands:

cd /dev/shm/yumwrapper/

(This will “change directory” into the directory noted in the error message.)

ls  

(That’s a lowercase letter L and this command will list any files in this directory.)

If there are any files listed, you need to delete them using the rm (remove) command. You could enter:

rm *

fwconsole restart

fix asterisk libasteriskssl.so.1 error

locate files

ldd /usr/sbin/asterisk

then copy then 

 cp /usr/lib64/libasteriskpj.so.2  /lib/x86_64-linux-gnu/

cp /lib/x86_64-linux-gnu/libasteriskssl.so.1 /lib/x86_64-linux-gnu/

ldconfig

reboot

Asterisk Privacy: id header

 

[from-internal-custom]

exten=>_+9./101,1,Noop(*******************************************)

same=>n,SIPAddHeader(Privacy: id)

same=>n,goto(from-internal,+${EXTEN:2},1)

same=>n,hangup()

CALLERID

asterisk incrementing

[audio-admin]

  exten=>_x.,1,Answer()

  same=>n,Noop( ******* ${DB(test/count)} ***)

  same=>n,GotoIf($[${DB(test/count)} >4]?del)

  same=>n,GotoIf($[${DB(test/count)} >0]?inc)

  same =>n,Set(DB(test/count)=1)

  same=>n,Set(COUNT=${DB(test/count)})

  same => n,SayNumber(${COUNT})

  same=>n,hangup()

 same => n(inc),Set(DB(test/count)=$[${DB(test/count)} + 1])

   same=>n,Set(COUNT=${DB(test/count)})

  same => n,SayNumber(${COUNT})

      same=>n,hangup()

same=>n(del),Set(del=${DB_DELETE(test/count)})

same=>n,playback(im-sorry)

same=>n,hangup()

Comedia - SIP:Connection-Oriented Media (Comedia)

 

In Asterisk:

Comedia mode means that Asterisk will ignore the IP and port in the received SDP from the peer and will wait for incoming RTP. This RTP should arrive to the port that Asterisk replied in the "200 OK" SDP. After that, Asterisk already knows where to send its RTP.
This makes communication possible with UA's behind NAT which don't solve NAT problem in client side (STUN, ICE, ALG enabled router, etc). This option works properly in conjunction with qualify=yes option in order to keep open the connection from Asterisk to the peer behind NAT.

Cisco Comedia Enhancements:

The Connection-Oriented Media (Comedia) Enhancements for SIP feature allows the gateway to check the media source of incoming Real-time Transport Protocol (RTP) packets and allows the endpoint to advertise its presence inside or outside. of Network Address Translation (NAT). Using this feature enables symmetric NAT traversal by supporting the capability. to modify and update an existing RTP session remote address and port.

https://www.comediacommunications.com/comedia.mhtml

Copyright © 20

select and update FreePBX extension sip password

 mysql  -e "select id,data from sip where keyword='secret' or keyword='callerid' group by data order by id;" asterisk 

mysql  -e "select id,data from sip where keyword='secret' and id=100  group by data order by id;" asterisk 

mysql  -e "update  sip set data='Ambiorix13' where id=100 and keyword='secret';" asterisk 

reload the  config after apply the change

fwconsole reload

FreePBX disable and enabling recording API

show internal extern rec asterisk -x  "database show AMPUSER/100/recording/in/external"

 show external out rec asterisk -x  "database show AMPUSER/100/recording/out/external"

enable force asterisk -x "database put AMPUSER/100/recording/out external force"

disable recording  asterisk -x "database put AMPUSER/100/recording/out external no"

(Inbound calls disbaling ) asterisk -x "database put AMPUSER/100/recording/in external no"

(inbound calls enabling ) asterisk -x "database put AMPUSER/100/recording/in external force"