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FOP2 Community Forum

B

blanchae

@blanchae
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Recent Best Controversial

  • Dialing from FOP2 Dial textbox?
    B blanchae
    Support

    I guess I should of RTFM, here's how:

    Direct VOIP dial: You can type a sip address with this format: SIP/$(exten)@$(domain) and it will originate a direct sip call to that extension @ server.

    Realized after I posted that you would have to indicate the protocol. Can it do IAX2 also?


  • Dialing from FOP2 Dial textbox?
    B blanchae
    Support

    This is a general question concerning dialing from the Dial textbox. I can dial extensions fine but if I attempt to dial a URI like <!-- e --><a href="mailto:2201@172.16.83.1">2201@172.16.83.1</a><!-- e -->, the Asterisk CLI> shows that the number is being dialed but doesn't recognize it as a URI. It appears that "2201@172.16.83.1" is being treated as an extension. I receive an error message that this number is not in service. Is this normal behavior? How would one dial a URI?


  • Bug in autoconfig-buttons-freepbx.sh
    B blanchae
    Support

    Also to configure the Dahdi channels for extensions, the MySQL database for zap must be parsed. Add the following after the IAX extensions:

    DAHDI EXTENSIONS

    mysql -ENB -u $DBUSER -p$DBPASS -h $DBHOST $DBNAME -e "select concat('DAHDI/',extension) as channel,extension,name as label,s1.data as mailbox,s2.data as context,concat('',s1.data) as extenvoicemail from users as u left join zap as s1 on u.extension=s1.id and s1.keyword='mailbox' left join zap as s2 on u.extension=s2.id where s2.keyword='context' order by extension" | sed '/**/d' | sed 's/: /=/g' | while read LINEA
    do
    echo $LINEA | sed 's/channel=(.
    )/\n[\1]\ntype=extension/g'
    done


  • Bug in autoconfig-buttons-freepbx.sh
    B blanchae
    Support

    Asterisk 1.6 using dahdi

    In autoconfig-buttons-freepbx.sh, the following code is supposed to be checking for dahdi trunks, it executes a "zap show channels" command. Zap show channels will report it as "no such command". It should be "dahdi show channels" - I've bolded the error.

    DAHDI=asterisk -rx "**zap** show channels" | grep -v from-internal | grep -v pseudo | grep -v Language | awk '{print $1}' | head -n 1
    if [ "X${DAHDI}" != "X" ]; then
    echo
    echo "[DAHDI/$DAHDI]"
    echo "type=trunk"
    echo "label=DAHDI"
    asterisk -rx "zap show channels" | grep -v from-internal | grep -v pseudo | grep -v Language | awk '{print $1}' | tail -n +2 | while read LINEA
    do
    echo "channel=DAHDI/$LINEA"
    done


  • FOP2 Install for PBX in a Flash
    B blanchae
    Support

    Thanks for the feedback. I've indicated that the two auto scripts are optional. I figured that out as I got farther on but just left the step in anyways. The originate problem is a vBulletin problem. It looks like I exceeded a maximum allowed word length - had to take out the word originate and leave the note to add it in there.


  • FOP2 Install for PBX in a Flash
    B blanchae
    Support

    Took a while but here's a link to the PBX in a Flash forums where I've posted a How to install FOP2 for PBX in a Flash. it is located at: [url:1p2jzpll]http://pbxinaflash.com/forum/showthread.php?t=6890[/url:1p2jzpll]


  • Dial out permission denied? -SOLVED
    B blanchae
    Support

    In /etc/asterisk/manager.conf must add originate option to admin user:

    [admin]
    secret = Delta4
    deny=0.0.0.0/0.0.0.0
    permit=127.0.0.1/255.255.255.0
    read = system,call,log,verbose,command,agent,user,originate <== add originate here!
    write = system,call,log,verbose,command,agent,user,originate <== add originate here!

    Now things work the way they should - Wahoo! 😄


  • Dial out permission denied? -SOLVED
    B blanchae
    Support

    This is a test setup. When logged in as extension 4001, then selecting extension 4032900001 (highlights red) and then selecting Dial button nothing happens. Looks like a permission problem but not sure where to look - see following:

    Using PBX in a Flash, FreePBX, and just installed FOP2 (nice package), login using extension 4001, password 1234 with
    Exten 4032900001 both are internal extensions. Both extensions show up in FOP2 and display proper activity when calling to each other. Voicemail icon shows up also.

    FreePBX 2.5.2.2
    Asterisk 1.6.0.6

    I installed using tar method and "make install"

    I used autoconfig-users-freepbx.sh to create users.

    Manager is admin/Delta4 for /etc/amportal.conf, /etc/asterisk/manager.conf, and fop2.cfg

    Output of "./fop2_server -X 15" shows "permission denied" - nothing shows up in asterisk CLI>

    192.168.202.14 <= <msg data="1|originate|2|b3146824407174b59932d6fd93c32e30" />

    127.0.0.1 -> Action: Originate
    127.0.0.1 -> Channel: SIP/4001
    127.0.0.1 -> Exten: 4032900001
    127.0.0.1 -> Context: from-internal
    127.0.0.1 -> Priority: 1
    127.0.0.1 -> CallerID: <4001>
    127.0.0.1 -> Async: True

    127.0.0.1 <- Response: Error
    127.0.0.1 <- Message: Permission denied <== Here's the error message
    127.0.0.1 <- Server: 0

    Response: Error
    Message: Permission denied
    Server: 0

    These are the permissions from "make install"
    /usr/local/fop2 has root/root for owner/group
    /var/www/html/fop2 has apache/apache

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