Openvox a400p01 dahdi channels


I am a new member and lack of knowledge brought me here…
I recently bought an openvox a400p01 card with 2 fxo modules and i use it with Freepbx Distro SNG7.I have one line(number) with 2 channels and i lead these 2 channels in the 2 fxo ports.
So,when pbx receives an incoming call(i have an inbound route set to an IVR),
both channels answer the call.I have an option in IVR that leads to miscellaneous destination(call to a mobile number)
My problem is that,when the caller chooses that option ,asterisk hangs up the call,because the second channel is busy too.It takes about 30 seconds for the second channel to “realize” that the first channel has answered the call.
My question is why is this happening?Is there a way to answer only one channel the call,so the second one will be “free”?
Sorry for my english…
Thank you in advance!


Generally questions about circuit switched hardware would be redirected to the vendor, but, in this case, the problem seems t be with the service provider. Why on earth is the service provide presenting the call on two analogue lines, on the same trunk, at the same time? What you actually require from your provider is a hunt group, not parallel ringing of two lines.

Asterisk will have to time out the cessation of ringing, before it can know that the second line is clear, but, even if that is done quickly, there is no guarantee that the provider will allow the line to be seized immediately.

Are the lines ground or loop start?

Also,any answers here will be in terms of directly editing the configuration files. Such edits may be incompatible with the use of FreePBX.


Hi David551,
thank you for your quick reply!

I have to be more specific…The line is voip…Saying that,i mean that there is a trunk between the provider and the modem/router.So the router has two outputs,phone 1and phone 2.From these 2 ports, the output is the same number(that’s why 1 line-2 analogue channels)
What i have done,is to create one dahdi trunk,group 0 round robin ascending,between these two outputs and asterisk server…
If there is a need to provide my configuration,please let me know…


You are using a service that, even if it does allow two simultaneous calls to external parties, is not intended for automated use. It is very unlikely that your service provider will even understand when you ask the sort of question you need to ask. It is also unlikely that the terminal device you have has documentation of an appropriate level, which is available end users.

It might be possible to get something to work, but you are likely to need quite a strong telephony background and have to do a lot of reverse engineering of the interface.