Question about AudioCodes MP118 Analog Modem

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AUDIOCODES MP-118 FXO

Hi, I am using Audiocodes MP-118 FXO.I cam make the calls from IP to PSTN.The only problem I have now is the call duration which is wrong.Since the call duration I takes is added with the ring time.So even if some one does not pick up the phone I get some 'x' call duration.

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Hi, I am using Audiocodes MP-118 FXO too
Man, check the Reverse Polarity.

1st.- Go to Protocol Management
2nd.-Go to Advance Parameter
3rd.- Go to General Parameter
4th.- Go to Enable Polarity Reversal and select ENABLE
5th.- BURN and RESET

You have FAS calls, because Reverse Polarity is disable.

Lucky

Posted on Nov 04, 2009

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Posted on Jan 02, 2017

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The easiest way (if enabled) is to use the voice menu as if you would assign a new IP to the device.

  1. Connect a telephone to one of the FXS ports. Lift the handset and dial ***12345 (three stars followed by the digits 1, 2, 3, 4, 5).
  2. Wait for the 'configuration menu' voice prompt to be played.
  3. To change the IP address, press 1 followed by the pound key (#).
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  • Dial the new IP address; use the star (*) key instead of dots ("."), e.g., 192*168*0*4 and press # to finish.
  • Review the new IP address, and press 1 to save it.

Of course you can try the default IP of 10.1.10.10 if the device is set to it's default standards.

If not, you can try a network scanner like angryip (link) or network scanner (link) - this might take a long time, since the IP can be virtually anything.

If you have physical access to the device, you can also use a straight RS-232 cable to connect the AudioCodes to your PC and use HyperTerminal to connect (Baud Rate: 115,200 bps (MP-124), 9,600 bps (MP-11x), Data bits: 8, Parity: None, Stop bits: 1, Flow control: None)

If you need the manual (and it never hurts ;-) here is the link:

http://www.audiocodes.com/filehandler.ashx?fileid=36320

If you have any more questions, just come back here and leave a comment - I'm happy to help.

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Hi, man.
There are many things to consider in this type of application.
when calling from PSTN to IP using analog lines from time to time you will have this type of event, I suggest working with H323 protocol, this will tone down a bit your problem.
If you want optimum performance, you should use a DID with the channels you need and address to your PBX SIP FreeSwitch. The DIDs are cheaper and are digital which eliminates random connectivity problems in the incoming call.
Once you have directed the DID you tell your server to transfer the call to the IVR using your internal dialing plan.

Lucky

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