Action buttons not working

  1. 13 years ago

    Hi all,

    I am new in FOP2 form.

    I have asterisk 1.8.2 and i have configured fop2 but somehow everything working except buttons like dial,transfer,pickup etc..

    I have full permission on manager.conf read,write "all"

    also

    I have full permission on fop2.cfg "all"

    Following my debug logs. -X 15 (when i am selecting on extension and press Dial button)

    I am logged in as 7658 extension and dialing 7623.

    172.30.254.222 <= <msg data="4|originate|6|a438e34b880a2b3660de52e726b34419" />

    127.0.0.1 -> Action: Originate
    127.0.0.1 -> Channel: SIP/7658
    127.0.0.1 -> Exten: 7623
    127.0.0.1 -> Context: from-sip
    127.0.0.1 -> Priority: 1
    127.0.0.1 -> CallerID: John <7658>
    127.0.0.1 -> Async: True

    127.0.0.1 <- Response: Success
    127.0.0.1 <- Message: Originate successfully queued
    127.0.0.1 <- Server: 0

    127.0.0.1 <- Event: Newchannel
    127.0.0.1 <- Privilege: call,all
    127.0.0.1 <- Channel: SIP/7658-00000018
    127.0.0.1 <- ChannelState: 0
    127.0.0.1 <- ChannelStateDesc: Down
    127.0.0.1 <- CallerIDNum:
    127.0.0.1 <- CallerIDName:
    127.0.0.1 <- AccountCode:
    127.0.0.1 <- Exten:
    127.0.0.1 <- Context: from-sip
    127.0.0.1 <- Uniqueid: 1297881733.29
    127.0.0.1 <- Server: 0

    127.0.0.1 <- Event: VarSet
    127.0.0.1 <- Privilege: dialplan,all
    127.0.0.1 <- Channel: SIP/7658-00000018
    127.0.0.1 <- Variable: SIPCALLID
    127.0.0.1 <- Value: 58eed36f6fbde6a46418d0bf2ea23707@[::1]:5060
    127.0.0.1 <- Uniqueid: 1297881733.29
    127.0.0.1 <- Server: 0

    127.0.0.1 <- Event: ChannelUpdate
    127.0.0.1 <- Privilege: system,all
    127.0.0.1 <- Channel: SIP/7658-00000018
    127.0.0.1 <- Uniqueid: 1297881733.29
    127.0.0.1 <- Channeltype: SIP
    127.0.0.1 <- SIPcallid: 58eed36f6fbde6a46418d0bf2ea23707@[::1]:5060
    127.0.0.1 <- SIPfullcontact:
    127.0.0.1 <- Server: 0

    127.0.0.1 <- Event: ChannelUpdate
    127.0.0.1 <- Privilege: system,all
    127.0.0.1 <- Channel: SIP/7658-00000018
    127.0.0.1 <- Channeltype: SIP
    127.0.0.1 <- SIPcallid: 58eed36f6fbde6a46418d0bf2ea23707@[::1]:5060
    127.0.0.1 <- SIPfullcontact:
    127.0.0.1 <- Peername:
    127.0.0.1 <- Server: 0

    127.0.0.1 <- Event: NewCallerid
    127.0.0.1 <- Privilege: call,all
    127.0.0.1 <- Channel: SIP/7658-00000018
    127.0.0.1 <- CallerIDNum: 7658
    127.0.0.1 <- CallerIDName: John
    127.0.0.1 <- Uniqueid: 1297881733.29
    127.0.0.1 <- CID-CallingPres: 0 (Presentation Allowed, Not Screened)
    127.0.0.1 <- Server: 0

    127.0.0.1 <- Event: NewAccountCode
    127.0.0.1 <- Privilege: call,all
    127.0.0.1 <- Channel: SIP/7658-00000018
    127.0.0.1 <- Uniqueid: 1297881733.29
    127.0.0.1 <- AccountCode:
    127.0.0.1 <- OldAccountCode:
    127.0.0.1 <- Server: 0

    127.0.0.1 <- Event: NewCallerid
    127.0.0.1 <- Privilege: call,all
    127.0.0.1 <- Channel: SIP/7658-00000018
    127.0.0.1 <- CallerIDNum: 7658
    127.0.0.1 <- CallerIDName: John
    127.0.0.1 <- Uniqueid: 1297881733.29
    127.0.0.1 <- CID-CallingPres: 0 (Presentation Allowed, Not Screened)
    127.0.0.1 <- Server: 0

    127.0.0.1 <- Event: Hangup
    127.0.0.1 <- Privilege: call,all
    127.0.0.1 <- Channel: SIP/7658-00000017
    127.0.0.1 <- Uniqueid: 1297881712.28
    127.0.0.1 <- CallerIDNum: 7658
    127.0.0.1 <- CallerIDName: John
    127.0.0.1 <- Cause: 0
    127.0.0.1 <- Cause-txt: Unknown
    127.0.0.1 <- Server: 0

    127.0.0.1 <- Event: OriginateResponse
    127.0.0.1 <- Privilege: call,all
    127.0.0.1 <- Response: Failure
    127.0.0.1 <- Channel: SIP/7658
    127.0.0.1 <- Context: from-sip
    127.0.0.1 <- Exten: 7623
    127.0.0.1 <- Reason: 3
    127.0.0.1 <- Uniqueid: <null>
    127.0.0.1 <- CallerIDNum: 7658
    127.0.0.1 <- CallerIDName: John
    127.0.0.1 <- Server: 0

    172.30.254.222 => { 'btn': '4@GENERAL', 'cmd': 'settimer', 'data': '0@STOP', 'slot': '1' }

    172.30.254.222 => { 'btn': '4@GENERAL', 'cmd': 'settext', 'data': '&inactive_line!1', 'slot': '1' }

    172.30.254.222 => { 'btn': '4@GENERAL', 'cmd': 'state', 'data': 'DOWN', 'slot': '1' }

    172.30.254.222 => { 'btn': '4@GENERAL', 'cmd': 'state', 'data': 'DOWN', 'slot': '0' }

    127.0.0.1 <- Event: JabberEvent
    127.0.0.1 <- Privilege: user,all
    127.0.0.1 <- Account: asterisk
    127.0.0.1 <- Packet:
    127.0.0.1 <- Server: 0

    172.30.254.222 <= <msg data="1|ping||" />
    172.30.254.222 => { 'btn': '0', 'cmd': 'pong', 'data': '0', 'slot': '' }

    127.0.0.1 <- Event: Hangup
    127.0.0.1 <- Privilege: call,all
    127.0.0.1 <- Channel: SIP/7658-00000018
    127.0.0.1 <- Uniqueid: 1297881733.29
    127.0.0.1 <- CallerIDNum: 7658
    127.0.0.1 <- CallerIDName: John
    127.0.0.1 <- Cause: 0
    127.0.0.1 <- Cause-txt: Unknown
    127.0.0.1 <- Server: 0

    ---------------------

    Here is my asterisk /var/log/asterisk/full logs

    [Feb 16 13:41:26] VERBOSE[5920] netsock2.c: == Using SIP RTP CoS mark 5
    [Feb 16 13:41:26] WARNING[5920] acl.c: Cannot connect
    [Feb 16 13:41:26] WARNING[5920] chan_sip.c: sip_xmit of 0x8b47428 (len 713) to 0.0.29.234:5060 returned -1: Invalid argument
    [Feb 16 13:41:26] WARNING[5838] chan_sip.c: sip_xmit of 0x8b3e5a0 (len 713) to 0.0.29.234:5060 returned -1: Invalid argument
    [Feb 16 13:41:27] WARNING[5838] chan_sip.c: sip_xmit of 0x8b14ba0 (len 715) to 0.0.29.234:5060 returned -1: Invalid argument
    [Feb 16 13:41:27] WARNING[5838] chan_sip.c: sip_xmit of 0x8b47428 (len 713) to 0.0.29.234:5060 returned -1: Invalid argument
    [Feb 16 13:41:27] WARNING[5838] chan_sip.c: sip_xmit of 0x8b271d8 (len 709) to 0.0.29.234:5060 returned -1: Invalid argument
    [Feb 16 13:41:27] WARNING[5838] chan_sip.c: sip_xmit of 0x8b34328 (len 715) to 0.0.29.234:5060 returned -1: Invalid argument
    [Feb 16 13:41:27] WARNING[5838] chan_sip.c: sip_xmit of 0x8b3e5a0 (len 713) to 0.0.29.234:5060 returned -1: Invalid argument

  2. admin

    17 Feb 2011 Administrator

    We talked on the live help yesterday... the problem was that you were logged as a different extension than your working one, right?

  3. Yes, It has been resolved. Problem was i forgot to comment "exec autoconfig-users-freepbx.sh" ;)

    Thank you so much for your help :idea:

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