[asterisk-users] Failover SIP logic

[Yusuf]

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Jeremy Mann wrote:
> I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence.  I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status
> 
> Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy)
> extensions.conf:
> 
> [globals]
> trunk_1 => SIP/trunk1
> trunk_2 => SIP/trunk2
> trunk_3 => SIP/trunk3
> 
> [macro-trunkdial]
> exten => s,1,Dial(${trunk_1}/${ARG1})
> exten => s,2,Hangup()
> exten => s,102,Dial(${trunk_2}/${ARG1})
> exten => s,103,Hangup()
> exten => s,203,Dial(${trunk_3}/${ARG1})
> exten => s,204,Hangup()
> 
> [from-internal]
> exten => _NXXNXXXXXX,1,Macro(trunkdial,+1${EXTEN})
> exten => _1NXXNXXXXXX,1,Macro(trunkdial,+${EXTEN})
> 
> sip.conf:
> 
> [trunk1]
> host=xxx.xxx.xxx.xxx
> port=5060
> type=peer
> allow=ulaw
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> nat=no
> fromuser=+xxxxxxxxxxx
> call-limit=1
> 
> [trunk2]
> host=xxx.xxx.xxx.xxx
> port=5060
> type=peer
> allow=ulaw
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> nat=no
> fromuser=+xxxxxxxxxxx
> call-limit=1
> 
> [trunk3]
> host=xxx.xxx.xxx.xxx
> port=5060
> type=peer
> allow=ulaw
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> nat=no
> fromuser=+xxxxxxxxxxx
> call-limit=1
> 
> Here's asterisk output when someone dials out:
> Executing [xxxxxxxxxx at from-internal:1] Macro("SIP/6001-007e2840", "trunkdial|+1xxxxxxxxxx") in new stack
>     -- Executing [s at macro-trunkdial:1] Dial("SIP/6001-007e2840", "SIP/trunk1/+1xxxxxxxxxx") in new stack
> [Sep 10 09:06:52] ERROR[16253]: chan_sip.c:3192 update_call_counter: Call to peer 'trunk1' rejected due to usage limit of 1
>     -- Couldn't call trunk1/+1xxxxxxxxxx
>   == Everyone is busy/congested at this time (0:0/0/0)
>     -- Executing [s at macro-trunkdial:2] Hangup("SIP/6001-007e2840", "") in new stack
> 
> I don't want the dialplan to cascade like:
> 
> exten => 1,dial...
> exten => 2,dial...
> 
> Because if the remote end hangs up I don't want it going to priority 2 to dial out again(in case my user doesn't hit hang-up on their end) so I need logic to detect a busy channel and jump to the next section..

If you have this:

exten => _X.,1,Dial(SIP/trunk1)
exten => _X.,2,Dial(SIP/trunk2)
exten => _X.,3,Dial(SIP/trunk3)

then, only if trunk is busy, will it go to trunk2, if thats busy, it will go to trunk 3. 
Reason is, is that control wont return to the dial plan(except h) if the call was 
successfull.  SO if the call went through on trunk 1, then it will exit, not dial trunk2 
or trunk3.  So this dial plan will work.  But its very sequential, i.e. will try trunk1, 
then trunk2, then trunk3.  If you want to replicate round-robin, r, then do this:

[globals]
IPt=trunk1-trunk2-trunk3
COUNTt=0

NoOfChannels=3


[just-an-idea]
exten => _XXXXXXXXX.,1,Gotoif($["${COUNTt}" = "${NoOfChannels}"] ? 2:3)
exten => _XXXXXXXXX.,2,SetGlobalVar(COUNTt=0])
exten => _XXXXXXXXX.,3,SetGlobalVar(COUNTt=$[${COUNTt}+1])
exten => _XXXXXXXXX.,4,Set(tr=${CUT(IPt,-,${COUNTt})})
exten => _XXXXXXXXX.,5,Dial(SIP/tr/${EXTEN})


modify at your leisure.  So if you get a few more trunks, you just change "NoOfChannels"


-- 

thanks,
Yusuf