Modem Calls - Transfer Capability


I am currently experimenting with trying to get a modem to dial out through our system.

The analog line is provided by a Vega 50 analog to IP gateway and is all set up for Data transfer however when the call is initiated it comes up with the following in Asterisk:

– Requested transfer capability: 0x00 - SPEECH

The modem handshake therefore fails, sometimes it connects but not for long. Sometimes Asterisk detects a CED tone and disables the echo canceller however not every time.

Looking at the below link suggests there are different settings specifically for fax and data calls but I have no idea on where to start with this. I don’t want to change anything that could impact our SPEECH calls as that side of things is working beautifully.

In our set up, FreePBX is connected to an Alcatel system via a Q-SIG E1 trunk which then gives us our ISDN30 outside lines.

Can anyone advise please?

FreePBX -
Asterisk -
PBX Firmware -

I’ve been doing a bit of digging with this but getting no-where fast

Is there context that I can create that either every call from a particular extension or to a particular number changes the transfer capability?

exten => 123,1,Set(CHANNEL(transfercapability)=SPEECH)

Any guidance appreciated

Modem calls are not data calls they are voice calls where the original digital signal is processed through a MOdulatorDEModulator, DAHDI will turn off echocancelling when it detect the 2100Hz carrier, I can’t speak to your Vega setup of it’s FXO ports used for moddem calls but do you have your PRI timing source set correctly?

I was on the phone to Vega support for almost 2 hours last week with remote support and they could not find any issue with the Vega unit. PRI timing is correct as this was confirmed. The guy came to the conclusion that Asterisk was not picking up the tone properly as it only disabled the echo canceller 5 times out of 10 but even with it did none of the calls would sync properly.

Perhaps you where misadvised, asterisk takes no part in the echo cancellation, on your asterisk box that is done by the DAHDI channel driver, you should check the levels with the milliwatt() application and adjust rx and txgain and dahdi_monitor to optimize the dahdi part of the path, but as you have an added complication of a SIP path to your gateway then other checks such as rtp integrity and jitter buffer disabling would also be necessary, I will also state from experience that connections above 9600 will almost never work over a SIP channel and that SIP channel MUST be g711. I would build and deploy an iaxmodem on the Asterisk box to isolate the current path into two paths, with that device you can connect in either direction to troubleshoot.

interesting. I’ll investigate and see how I get on.

Thanks for the info