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    Ken,

    You need to create dynamic variables. For example create variable row${counter}.
    If there are 3 records in result set, first variable that stores first record from database would be row1, second would be row2, and third would be row3. After each fetched row you need to increase counter for 1, counter = counter+1.

    I often use this approach with Integration Server, you may try IS it’s so much easier to retrieve and manipulate data from a database.

    Raman

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    Thanks Raman 🙂

    I managed to get it all working using ${CALLERID(num)} function.

    Ken

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    Loopyjoe7,

    UniqueID is composed of the epoch when a call starts, plus a monotonically incremental integer.
    Together, they will be unique for all calls originating from a single machine.

    You need to use ${CALLERID(num)} function instead of ${UNIQUEID}.

    Raman

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    Xygod,

    Do you have /etc/asterisk/globals_custom.conf file on your system and what version of FreePBX server do you use?

    You may also consider uploading a VDP log file and all /etc/asterisk/extension*.conf files so I could take a closer look at the configuration files.

    Raman

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    Arturo,

    Here is a link to Visual Dialplan and FreePBX tutorial page:
    http://www.apstel.com/visual-dialplan-tutorials/how-to-work-with-elastix-etc/

    What version of FreePBX are you using?
    Did you call 56781 extension for Hello-world sample?

    Raman

    • This reply was modified 10 years, 6 months ago by  admin.
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    in reply to: elastix plugin #5783 |

    I have the exact same problem with vdp for Elastix addon, finally I decided to download and try professional version from http://www.apstel.com.(external link)
    Problem solved 🙂

    Flux82

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    in reply to: Inbound SIP problems #5781 |

    The problem was actually a FreePBX issue (SIP values). I did a google search and found that adding insecure=port,invite to the outging section caused the incomming to work. I had read ‘Visual Dialplan and FreePBX’ and any modifications made were all done in FreePBX. Using Start extension was because I had copied one of the contexts from another Dialplan and it used Start Extension. All is well now and I’m on my way.

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    in reply to: Inbound SIP problems #5780 |

    Jim,

    The context should be started with Extension, not Start extension block.
    You can use ‘catch all’ extension (“_.”) to override all inbound or outbound calling rules in Free PBX or specific extensions (56781 or any other) to handle only those specific extensions

    Also, there is no need to set:
    context=vdp-inbound
    Just leave it as it is by default.
    BTW, the outgoing calls will be handled by the contex set in this field, not incoming calls.

    Visual Dialplan hooks code defined in vdp-outboung and vdp-inbound at appropriate places (designed for this) inside Free PBX.
    There is no need for you to make modifications like you did.
    Simply follow context structure defined in VDP and all will work just fine.

    Make sure to read this article:
    http://www.apstel.com/visual-dialplan-tutorials/how-to-work-with-elastix-etc/
    It is the best article I read about this.
    You will get it all once you read this.

    Best,
    Mark.

    • This reply was modified 10 years, 6 months ago by  admin.
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    in reply to: SIP Trunk Group #5778 |

    Finally solved! WILINE gave me 2 seperate SIP Trunks because I asked for two numbers. Turns out that the one trunk handles all calls and the second trunk isn’t needed. All I needed to do was set the Maximum Channels in FreePBX to be the number of calls we are authorized from our location and just use the first group.

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    in reply to: SIP Trunk Group #5777 |

    Another way that came to my mind was to use a separate Dial block for each outgoing line and use the busy of one to go to the in of the next. Does this way make any sense?

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    in reply to: SIP Trunk Group #5776 |

    I’ve called the trunk group “WILINE”. Would it be possible to use a “Set(CHANNEL=WILINE)” and then use Dial(SIP/${CHANNEL}). Also, GROUP_LIST()funtion looks as though it returns a count. Shouldn’t it just be the GROUP() function?

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    in reply to: SIP Trunk Group #5775 |

    Jim,

    Have you tried with GROUP_LIST() function?

    Bernard

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    in reply to: SwitchedonPBX #5773 |

    I’m currently using VDP 3.1.06 with Plain 1.8 Asterisk and it works like a charm biggrin. The truth is I haven’t had an opportunity to try Switchedon PBX … It’s a new distribution and the question is how stable it is.

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    Romerjm,
    Depending on the asterisk distribution it may or may not detect it correctly. Just manually select Freepbx and you should be fine.

    Kevin

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    Sure you can do this, there are even some dial plan examples bundled with Visual Dialplan that shows how you can collect digits and do database query.
    Check Single select and Multiselect database examples that comes with VDP.
    You may also want to check the following tutorial that describes how to work with databases in Asterisk:
    http://www.apstel.com/visual-dialplan-tutorials/database-connectivity/

    • This reply was modified 10 years, 6 months ago by  admin.
Viewing 15 posts - 241 through 255 (of 327 total)