Sorry to drag out an old thread, but this is exactly the problem I am having and I would like to know what feh had to do to solve it.
In depth, I am attempting to interface a VOIP gateway/server to my Nortel Norstar PBX to enable me to add WIFI SIP phones to our site. I have a few ATA2's to provide analog lines, and I am having two problems with them.
1) As with feh, the DTMF signal is NOT getting through. When I attach an old school analog phone to the ouput on the ATA2 and call it from a nortel phone, there is not audible DTMF. I have heard about feature 808 and "long tones" but I am confused. Do I have to dial "feature 808" before EVERY call that I want dtmf to get through? Is there no better way?
2)The ATA2 doesn't appear to provide any call disconnect supervision. Therefore, if a nortel phone user calls a SIP phone as below
[nortel ph.]->[Norstar]->[ATA2]->[VOIP]->[sip ph.]
and gets forwarded to voicemail (the VOIP has an integrated voicemail box), there is no indication to the VOIP server that the caller has hung up. You end up with a message that is about 10 minutes long - 15 seconds of voice follwed by a click and a dialtone.
Feh seemed to have found a solution - anybody care to explain what he did?
