search for: busy_max

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, Asterisk currently uses the timestamps in incoming RTP packets to build outgoing voice frames. Is this true? Would it be possible for me to use i.e. zaprtc as a timing source for the outgoing stream? I.e. in a setup like below I'd like to use zaprtc timing on Ast1 because I don't trust the timestamps coming from

Thanks for the suggestion. I have busydetect=yes in zapata.conf. You refer to "Martin's algorithm". Can you provide more details please? ----- Original Message ----- From: "Soner Tari" <list@kulustur.org> Subject: Re: [Asterisk-Users] FXO module in TDM400P (UK, BT) - Hangup detection failing To: "Asterisk Users Mailing List - Non-Commercial Discussion"

So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither