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madlac

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Posts posted by madlac

  1. Update, maybe it will help someone :)

     

     

    As I couldn't get my call-in to work with Acanac ( my home phone number), I registered a free account with Voxalot and thru a Sip Broker local PSTN access line I can get into my IVM now.

    Thx to the help I got here I was able to set up my IVM so I can call an extension on my Axon now, no matter where that extension is located, also I was able to call out to a variable number using my dial-out plan.

    Problem is resolved. :)) (hopefully)

     

     

    Thx to all who helped.

  2. Contacted my phone provider and they say it must be my PBX as their Asterisk is a standard one...

    So I assume is some setting inside Axon as others complain about similar issues. Could you guys make this "hold call ? signaling" a selectable option in the next update ?

     

    Thx

     

     

    P.s. here is a piece of Sip trace I got, just wondering if it would be something in the Allow line ??

     

     

    15:36:33 UDP Packet Received from **.**.255.42:5060 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

    OPTIONS sip:905******@*****.homeip.net:5060 SIP/2.0

    Via: SIP/2.0/UDP **.**.255.42:5060;branch=z9hG4bK565698f0;rport

    From: "asterisk" <sip:asterisk@6**.**.255.42>;tag=as35b4e0e0

    To: <sip:905*******@******.homeip.net:5060>

    Contact: <sip:asterisk@**.**.255.42>

    Call-ID: 40bb6c1c764e4f7c4e3a334874c7e48b@**.**.255.42

    CSeq: 102 OPTIONS

    User-Agent: Asterisk PBX

    Max-Forwards: 70

    Date: Thu, 13 Mar 2008 19:36:34 GMT

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

    Supported: replaces

    Content-Length: 0

     

     

    I only have VOIP numbers thru Axon, and yes, if I put the chimes on, it just hangs up at the 1st ring....

    Probably some signalling issue between Axon and my provider's Asterisk ( Acanac.ca)

     

     

    Thx

  3. I only have VOIP numbers thru Axon, and yes, if I put the chimes on, it just hangs up at the 1st ring....

    Probably some signalling issue between Axon and my provider's Asterisk ( Acanac.ca)

     

     

    Thx

     

     

     

    Are you referring to analogue calls (e.g. through a modem or telephony board)? Or what is meant by calls outside VoIP? Is the problem that incoming VoIP calls hang-up after getting transferred to an Axon group (when hold chimes is enabled)?
  4. 1st I thought it's a IVM problem but I've found out this :

    If I put the Group Hold Options while waiting on Basic Hold Chimes, calls coming in from outside VOIP just disconnect right away. If I leave it on Ringing only, it works ok...

    The log shows my provider uses Asterisk.

     

    CSeq: 103 BYE

    User-Agent: Asterisk PBX

    Max-Forwards: 70

    X-Asterisk-HangupCause: Normal Clearing

    X-Asterisk-HangupCauseCode: 16

    Content-Length: 0

     

    Any ideas ? Thx

  5. After reading thru forums, IVM?? or Axon?? might send something to my home phone provider when puts the call on hold so it disconnects the line ??????

     

     

    Update :

     

    Registered a free Voxalot account and thru Sip broker PSTN local access number works now. So its definitely my provider's Asterisk not working with Axon & IVM properly.

  6. and my work cell, and my wife cell :)))))

    I'm one step closer, thx to you : if i call from a voip to voip line it works.

     

    11:36:45 Incoming SIP call

    11:36:47 Ringing on Line 198 detected. Waiting until ring number 3.

    11:36:49 VoIP Call Attendant *******267 [Answered]

    11:36:49 Answering call...

    11:36:50 Answered line [1 "VoIP Call Attendant"] call number [110] cid [*******267] did[198] drn[0 (0ms)]

    11:36:50 Play file: C:\Documents and Settings\All Users\Application Data\NCH Swift Sound\IVM\OGMs\VoIP Call Attendant Start.wav

    11:36:52 Caller pressed key [*]

    11:36:52 Command - Go

    11:36:52 Open OGM: Variable number

    11:36:52 Play file: C:\Documents and Settings\All Users\Application Data\NCH Swift Sound\IVM\OGMs\Variable number.wav

    11:36:54 Caller pressed key [9]

    11:36:54 Caller pressed key [0]

    11:36:54 Caller pressed key [5]

    11:36:55 Caller pressed key [x]

    11:36:55 Caller pressed key [x]

    11:36:55 Caller pressed key [x]

    11:36:56 Caller pressed key [x]

    11:36:57 Caller pressed key [x]

    11:36:57 Caller pressed key [x]

    11:36:57 Caller pressed key [6]

    11:36:57 Caller pressed key [#]

    11:36:57 Variable numcall = 905xxxxxx6

    11:36:57 Command - Transfer

    11:36:57 Voip Blind Transfer To: 905xxxxxx6

    11:36:57 Play system prompt: Ringing

    11:36:57 Transferring sip call to: 905xxxxx6

    11:36:57 Call has disconnected

     

    and voila, it rings ! :))))

     

     

    If I call from my rogers (gsm) cell, it rings once then hangs up....

     

    11:48:56 Incoming SIP call

    11:48:58 Ringing on Line 198 detected. Waiting until ring number 3.

    11:49:00 VoIP Call Attendant 905******6 [Answered]

    11:49:00 Answering call...

    11:49:00 Call has disconnected

    11:49:00 Answered line [1 "VoIP Call Attendant"] call number [113] cid [905*****6] did[198] drn[0 (0ms)]

    11:49:00 Play file: C:\Documents and Settings\All Users\Application Data\NCH Swift Sound\IVM\OGMs\VoIP Call Attendant Start.wav

    11:49:00 Call disconnected

     

    Any ideas ? :)

     

     

    Yeah, I just added that assuming you only wanted your cell to have access to the system.
  7. I didn't put % around the variable number, will try it this way, thx so much

    would it work without the id validity check ?

     

     

    In the answer OGM's properties, go to the Date/Time ID tab and tick User Number ID Validity Checking and enter your cell number in the textbox and select the OGM to 'skip' to.

     

    In the second OGM's properties, under the Key Response select Data Entry (variable). Enter a variable name (e.g. numvar), set the maximum digits to the max number of digits to accept (e.g. you could set this to 10 to prevent most international calls). Under # or end select Transfer call and enter %numvar% in the textbox. I recommend a blind transfer for this application, but you can select a confirmed transfer if you want.

     

    Now just so long as one of your telephony devices / voip lines has Allow outbound and message forward calls on this line ticked (under the Options tab), IVM should be able to transfer the call. (Note that not all VoIP providers support call transfers).

  8. Somebody please help. It's driving me crazy that I can't figure it out.

    I've been using Axon for about 18 months now, I have 5 voip lines from different countries including my home phone and I'm happy with it.

    What I'd like to do is to be able to call from my cell my own number, enter a new number then somehow redirect that number to an outgoing one using the existing dial plan.

    I've been able to configure OGM-s to record the new number as a variable but it just hangs up.

    I'd like to buy IVM but if I can't figure this out, it's useless.

     

    Thx

  9. Yes, simple create a dialing plan in Axon that will use the appropriate line based on the dialed number (eg if number starts with 2, use line 2 and remove the "2" digit from the start of the number, otherwise use line 1).

     

    Then using your softphone or hardware phone / ata adapter's transfer feature (if it has one) to initiate a call transfer and use the dialing rules you have set in Axon to dial the number with the 2nd line (eg 2 NNN NNNN).

     

    If you want an automatic transfer (eg, you call in to your house, and then get a prompt where you are asked to enter the number you wish to dial and then have it transfer you so you don't pay outrageous cellphone prices for the long distance call, then you will need IVM.

     

    Thx, the 2nd one is exactly what I'd like to do : to be able to call home from my cell and be able to use one of my outbound lines to call my parents in Europe etc.

    I have IVM intalled, couldn't figure out yet how to do it, though.

    Any help to at least where to start would be much appreciated.

  10. Hi,

     

    I have multiple VOIP accounts from different countries and 1 Linksys 2102 extension in my house.

    They work perfect, routing them is very easy.

    I would like to configure my system, so when I call a VOIP line from my cell , I can transfer my call to another VOIP line. Is it possible ?

     

    Thx

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