[Resolved]SIP register failover

I got a sip register account, for my VOIP provider
but I’d like to know if it’s possible to put a failover on this account, and if this account is logoff, I use my FXO card to make the call ?

I 'm not sure that is really clear…
(but I’m french be cool with me plz :laughing: )

up ? I’m sure it’s possible… I found something like that using the last gui

but I like to understand, trunk_1 is a sip user (users.conf) that is my VOIP provider
trunk_2 is a FXO user (users.conf) plug on my traditionnal line but I don’t understand where are the Dail(ZAP … and the Dial(SIP/

what trunkdial-failover-0.3 mean???

Plz help me[/code]

For people interrested in Here 's the solution
here the macro.

[macro-trunkdial-failover-0.3] exten = s,1,GotoIf($[${LEN(${FMCIDNUM})} > 6]?1-fmsetcid,1) exten = s,2,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} > 1]?1-setgbobname,1) exten = s,3,Set(CALLERID(num)=${IF($[${LEN(${CID_${CALLERID(num)}})} > 2]?${CID_${CALLERID(num)}}:)}) exten = s,n,GotoIf($[${LEN(${CALLERID(num)})} > 6]?1-dial,1) exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${CID_${ARG3}})} > 6]?${CID_${ARG3}}:${GLOBAL_OUTBOUNDCID})}) exten = s,n,Goto(1-dial,1) exten = 1-setgbobname,1,Set(CALLERID(name)=${GLOBAL_OUTBOUNDCIDNAME}) exten = 1-setgbobname,n,Goto(s,3) exten = 1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM}) exten = 1-fmsetcid,n,Set(CALLERID(name)=${FMCIDNAME}) exten = 1-fmsetcid,n,Goto(1-dial,1) exten = 1-dial,1,Dial(${ARG1}) exten = 1-dial,n,Gotoif(${LEN(${ARG2})} > 0 ?1-${DIALSTATUS},1:1-out,1) exten = 1-CHANUNAVAIL,1,Dial(${ARG2}) exten = 1-CHANUNAVAIL,n,Hangup() exten = 1-CONGESTION,1,Dial(${ARG2}) exten = 1-CONGESTION,n,Hangup() exten = 1-out,1,Hangup()

and here the line I use

first argument of the macro is your first way to call so here you can put SIP/anithing… all the thing can be put in in instruction Dial
Second argument it’s your rescue way it’s the same you can put any argument that can be put in instruction Dial
The third and fourth meant to be your callerID if you 're from the first or the second way…But i’m not sure 100% sure so if you can confirm?
I got a sip provider, and when my sip porvider don’t answer (no login, no network) the call is going through my FXO card

Thanks you and I hope my explanation 's clear enough

Good bye