SIP trunks do not fall through to next trunk

We have registered with two voip providers.
We have built an Ourbound Route with the Trunk Sequence containing these two SIP trunks.
When we place a call while the first voip (SIP) provider is busy, we get a message from the operator of the first provider, that we cannot use the account, which is correct.
How can we overcome this problem.
The problem does not appear when we use our ISDN trunks

As you mentioed you want to route call to another sip lin that u have when first sip out linne is busy
then in ur out going call context you sud do following arrangement

[out call-contex]
exten => _1XXXXXXXXXX,1,Dial(SIP/${EXTEN}@SWITCH-Sipline1,T)
exten => _1XXXXXXXXXX,n,goto(s-${DIALSTATUS},1)
exten => s-BUSY,1,Dial(SIP/${EXTEN}@SWITCH-Sipline2,T)
This will work for your requiremnt only thing you have 2 create 2 peers for 2 Sip provider in Sip.conf name is Swtich-siplin1 & swtich-siplin1
try this is will help