search for: _0z

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not answering (in 3sec) or send "50x" error? Next idea is to use both peers in round-robin, if they are working. Could someone help? Regards Thomas

...th a few friends. It is connected to PSTN, and various SIP Providers. I offer Free Calls to my friends, but myself I would like to be able to make calls to non free destinations via my PSTN Line. Now I do this in my dialplan: ----------------------- [myself] ; National Destinations exten => _0z.,1,Dial(SIP/someisp/${EXTEN}); exten => _0z.,n,Dial(Zap/g1/${EXTEN}); ; International Destinations exten => _00z.,1,Dial(SIP/someisp/${EXTEN}); exten => _00z.,n,Dial(Zap/g1/${EXTEN}); include => freedestinations; [freedestinations] ; Local Free Destionations exten => _0800.,1,Dial...

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant