I have a small problem that has so far eluded me. I have incoming calls coming into Asterisk via PSTN lines to a FXO port on a TDM400P card. The incoming call causes another Zaptel port to be dialled, i.e., an analogue phone connected to a FXS port. My predicament is that when the analogue phone rings I want to be able to pickup the call on a SIP handset. Any suggestions would be greatly appreciated. Many thanks, Stephen. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040727/c7ae489a/attachment.htm
> My predicament is that when the > analogue phone rings I want to be able to pickup the call on a SIP > handset. Any suggestions would be greatly appreciated. >How about configuring your extensions.conf to ring BOTH the analogue phone and the SIP as well? something akin to: exten => s,1,Answer exten => s,2,Dial(Zap/1&SIP/user) That might work... Cheers, Faiz
Hi, Was just wondering if a few people could send me a few ring1.bin etc ring tones for the Grandstream phone. I attempted to make my own, and it appeared to all work, and I even logged the phone tftping the ring1.bin without errors from my tftpserver, however, no joy.. If some one can send me a file they know works, so I can see if the phones are the right rev,B type phones, I would appreciate it, Thanks, James