Steve-Alloy Posted September 12, 2008 Report Share Posted September 12, 2008 Is there anyway to automatically pass an incoming sip call to a CS410 and have the dial plan configured to pass the call out an FXO port? Reason is i am testing faxing s/w on a Windows PC and have configured a registration in the CS410 with the IP address of the PC which allows the call to come into the CS410 (which i can see from the system logs) but from there the CS410 does not know what to do with the call, i dont want to send it to an extension but rather send it out an FXO port as a T.38 fax. Is this possible? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 12, 2008 Report Share Posted September 12, 2008 Is there anyway to automatically pass an incoming sip call to a CS410 and have the dial plan configured to pass the call out an FXO port? Reason is i am testing faxing s/w on a Windows PC and have configured a registration in the CS410 with the IP address of the PC which allows the call to come into the CS410 (which i can see from the system logs) but from there the CS410 does not know what to do with the call, i dont want to send it to an extension but rather send it out an FXO port as a T.38 fax. Well you can add a static registration to an extension that looks like this: "sip:0@127.0.0.1:5062;line=4" (where 4 would be the FXS 4). Quote Link to comment Share on other sites More sharing options...
Steve-Alloy Posted September 15, 2008 Author Report Share Posted September 15, 2008 Well you can add a static registration to an extension that looks like this: "sip:0@127.0.0.1:5062;line=4" (where 4 would be the FXS 4). I have added a static registration with the following "sip:0@127.0.0.1:5062;line=1". Its still not working and all faxes are failing. Does the "sip:0" need to be changed or is that what is needed for a static route to work? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 15, 2008 Report Share Posted September 15, 2008 I have added a static registration with the following "sip:0@127.0.0.1:5062;line=1". Its still not working and all faxes are failing.Does the "sip:0" need to be changed or is that what is needed for a static route to work? Does the FXS extension actually ring? Keep in mind this is FXO, and the gateway will DTMF the destination number out. You might have to put the real extension number in there, e.g. sip:123@127.0.0.1:5062;line=1. Quote Link to comment Share on other sites More sharing options...
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