Question about AudioCodes MP118 Analog Modem

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Audiocodes MP118 8 FXO early media to Freeswitch SIP extension

I have one Audiocodes MP118 8FXO media gateway, between PSTN and internal SIP PBX Freeswitch, when calling from PSTN to IP, in this case, FXO automatic dialing ext 5001 on SIP server which is supposed to play IVR prompt, the problem is 1) many times when PSTN calls get answered some random media inserted before the right IVR prompt

2) some times the IVR prompt got cut off for the initial part

Audiocodes MP118 setup with not registering with SIP server, early media disabled. I have no single issue when calling from any SIP phones inside the network to the ext 5001 which is a main IVR.

could you please shed some light on this as you guys are Audiocodes experts?

Chris Chen

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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.


Posted on Nov 04, 2009

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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.


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