alan@mousty.com

Member

Last active 11 years ago

  1. 11 years ago
    Mon Jul 15 14:21:43 2013
    alan@mousty.com started the conversation Users changing their own passwirds.

    Is there a way for FOP2 users to change their own passwords? If there is not then can this be added as a feature in the future.

  2. Fri Apr 12 16:59:16 2013
    alan@mousty.com started the conversation List of what each permission means.

    Is there a list describing what each permission does for FOP2? Some are explanatory but some seem not to be.

    What are the minimal permissions that a user needs? It would be nice if there were some built-in templates for a standard user, admin, power user.

  3. Mon Mar 18 23:51:25 2013
    alan@mousty.com posted in New FOP2 license not working.

    Nicolás

    Thank you. That worked. It even has the apostrophe so the customer will be happy.

    Thanks for all your hard work. I am looking forward to the next version of FOP2.

    Regards
    Alan

  4. Sun Mar 17 18:08:12 2013
    alan@mousty.com started the conversation New FOP2 license not working.

    Nicolás

    I just purchased a license code for FOP2 with the Voicemail Explorer. I received the code and registered. After the registration FOP2 does not work. After I log in the main page of extensions will not display. I think the problem is that when I registered I used the name "Woman's Health Center" with an apostrophe character ' in the name. Is it not possible to use the apostrophe character in the name? If I revoke the license and use the free version the system works? If it is not possible to use the apostrophe character in the name how to I re-register without it?

    Thanks
    Alan

  5. 12 years ago
    Mon Sep 10 02:44:37 2012
    alan@mousty.com posted in FOP2 debug help. Can't connect to server.

    That worked. Thank you.

  6. Thu Sep 6 13:03:00 2012
    alan@mousty.com started the conversation FOP2 debug help. Can't connect to server.

    I have a FreePBX 2.9 system and installed FOP2. When I try to log in I get the message connecting to server. After many attempts I get a message that the server is not available.

    I have disables iptables and still the same problem.

    Here are some debug logs.

    ** DIGEST_EVENT: start

    • * PROCESA_BLOQUE 0
    • * PROCESA_BLOQUE MESSAGEWAITING 0

    It's blessed into class Extension

    DIGEST canal SIP/501, comando voicemail, datos 0, slot

    DIGEST canal SIP/501, comando voicemailcount, datos &vmail_count!0!0, slot

    • * MAIN New client connection 192.168.250.77
    • * MAIN New client connection 192.168.250.77

    ** MAIN AMI event received...

    HTML5 Web Socket Connection

    websocket handshake version hiby v8

    HTTP/1.1 101 Switching Protocols
    Upgrade: WebSocket
    Connection: Upgrade
    Sec-WebSocket-Accept: UJ3Mzq/G3BJMXwV1dugyNWyLsyQ=

    192.168.250.77 => HTTP/1.1 101 Switching Protocols
    Upgrade: WebSocket
    Connection: Upgrade
    Sec-WebSocket-Accept: UJ3Mzq/G3BJMXwV1dugyNWyLsyQ=

    • * MAIN AMI event received...
    • * MAIN Processing command received from flash clients...

    192.168.250.77 <= <msg data="GENERAL|contexto|1|" />

    -- PROCESS_FLASH_COMMAND origen GENERAL accion contexto destino 1 password

    192.168.250.77 => { "btn": "0", "cmd": "key", "data": "XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", "slot": "" }

    192.168.250.77 => { "btn": "0@GENERAL", "cmd": "version", "data": "2.26!XXXXXXXXXXXXXXXXXXl, IL!1", "slot": "" }

    ** MAIN New client connection 127.0.0.1

    • * MAIN AMI event received...
    • * MAIN Processing command received from flash clients...

    127.0.0.1 <= <msg data="GENERAL|contexto|1" />

    -- PROCESS_FLASH_COMMAND origen GENERAL accion contexto destino 1 password

    127.0.0.1 => { "btn": "0", "cmd": "key", "data": "XXXXXXXXXXXXXXXXXXXXh", "slot": "" }

    127.0.0.1 => { "btn": "0@GENERAL", "cmd": "version", "data": "2.26!XXXXXXXXXXXXXXXXXXXXXXXXL!1", "slot": "" }

    • * MAIN AMI event received...
    • * MAIN Processing command received from flash clients...

    127.0.0.1 <= <msg data="GENERAL|checkauth|301|3cd6b224aae4a683d2ced1942e813762" />

    -- PROCESS_FLASH_COMMAND origen GENERAL accion checkauth destino 301 password 3cd6b224aae4a683d2ced1942e813762

    • * CLEAN SOCKET connection lost removing socket IO::Socket::INET=GLOB(0x111d5c54)
    • * CLEAN SOCKET flash client connection lost

    Flash clients connected: 2
    ---------------------------------------------------
    192.168.250.77:62837 IO::Socket::INET=GLOB(0x1114ec18), none
    192.168.250.77:62838 IO::Socket::INET=GLOB(0xeaccd80), none
    ---------------------------------------------------

    ** MAIN closing 192.168.250.77

    • * CLEAN SOCKET connection lost removing socket IO::Socket::INET=GLOB(0x1114ec18)
    • * CLEAN SOCKET flash client connection lost

    Flash clients connected: 1
    ---------------------------------------------------
    192.168.250.77:62838 IO::Socket::INET=GLOB(0xeaccd80), none
    ---------------------------------------------------

    ** MAIN End of block from localhost

    127.0.0.1 <- Event: PeerStatus
    127.0.0.1 <- Privilege: system,all
    127.0.0.1 <- ChannelType: SIP
    127.0.0.1 <- Peer: SIP/800
    127.0.0.1 <- PeerStatus: Registered
    127.0.0.1 <- Address: 10.253.181.59:5060
    127.0.0.1 <- Server: 0

    • * MAIN AMI event received...
    • * MAIN There's an 'Event' in the event block
    • * MAIN Event detected block_count = 1 (Event: PeerStatus)
    • * MAIN There are 1 blocks for processing
    • * MAIN Answer block cleared

    ** DIGEST_EVENT: start

    • * PROCESA_BLOQUE 0
    • * PROCESA_BLOQUE PEERSTATUS 0

    SET QUALIFY para SIP/800 lo dejo como estaba ok

    ** MAIN End of block from localhost

    127.0.0.1 <- Event: PeerStatus
    127.0.0.1 <- Privilege: system,all
    127.0.0.1 <- ChannelType: SIP
    127.0.0.1 <- Peer: SIP/8009
    127.0.0.1 <- PeerStatus: Registered
    127.0.0.1 <- Address: 10.253.181.59:5060
    127.0.0.1 <- Server: 0

    • * MAIN AMI event received...
    • * MAIN There's an 'Event' in the event block
    • * MAIN Event detected block_count = 1 (Event: PeerStatus)
    • * MAIN There are 1 blocks for processing
    • * MAIN Answer block cleared

    ** DIGEST_EVENT: start

    • * PROCESA_BLOQUE 0
    • * PROCESA_BLOQUE PEERSTATUS 0

    SET QUALIFY para SIP/8009 lo dejo como estaba ok

    Exiting...

  7. Sun Aug 26 03:25:00 2012
    alan@mousty.com started the conversation USE FOP to spy on extensions on remote system.

    I have a customer with 2 store locations. Both locations have their own dedicated FreePBX systems each with a FOP2 license. The store locations are connected via a VPN. I have IAX2 trunk setup between the two stores and routing in place to allow the systems to call over the IAX2 trunk,

    The owner of the business wants the ability to monitor calls on both systems from a single location. The owner is usually at store 1 and uses FOP2 to spy on calls. Is there a way to configure FOP2 so that he can also spy on the calls at store 2?

    My current plan is to program the Polycom phone to register an extension from each server. This would give the ability for the owner to access the individual FOP2 instances at each store and listen to calls. The only problem is that he will have to log into two different FOP2 instances (one at each store) to monitor calls.

  8. Sun Jul 29 14:20:33 2012
    alan@mousty.com posted in Does FOP work with Asterisk 1.8 and above.

    I am running FreePBX. If I download the version that you provide on this site will I have problems? What are the issues?

  9. Sat Jul 28 22:37:32 2012
    alan@mousty.com started the conversation Does FOP work with Asterisk 1.8 and above.

    I was curious if the original FOP works with Asterisk 1.8 and greater? Do all the features of FOP work correctly?

  10. Tue Jul 17 04:01:29 2012
    alan@mousty.com posted in Do not see recordings in user portal.

    I did not realize that you had to look at the phone book to see the call recordings.

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