Multiple SIP Trunks - Failover setup

[jepps3]

I know this has been discussed several times on this forum but maybe I can get a little detailed help on this one. I've done the search but still confused.

I have 5 different sip trunks, each with its own user and password logins. Only one call at a time on each trunk allowed by the carrier. I have 5 agents and I want the campaign to use all trunks available, in a failover fashion when dialing out. So if trunk 1 is busy, it will try trunk 2, and if thats busy it will go on till it hits the last one.

Ive seen mflorell show a FAILOVER dailplan such as the following:

exten => _91NXXNXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_1.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_2.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_3.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_4.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_5.com)
exten => _91NXXNXXXXXXXX,n,Hangup


Is this correct for a failover type system? If so, I just dont understand where the @sbc.provider_1.com is defined. Is it in the sip.conf?

Can you show me the correct way to add SIP trunks in the [globals] section in the extension.conf?

Also, can you tell me how to properly input the trunks in the SIP.CONF so it will work correctly with the EXTENSION.CONF??

[mflorell]

admin.php version and build?

[jepps3]

Vicidialnow 1.2

[gardo]

The admin.php version can be found on the lower left corner of your admin page. Something like:

VERSION: 2.0.5-174
BUILD: 90522-0506

[jepps3]

Oh, here it is.

VERSION: 2.0.5-173
BUILD: 90320-0424

[Op3r]

Yes this can be used as a fail over if the first dial fails.

exten => _91NXXNXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_1.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_2.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_3.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_4.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_5.com)
exten => _91NXXNXXXXXXXX,n,Hangup

[jepps3]

How would you properly setup the "sbc.provider_5.com" part in the SIP.CONF?

Also, do I have to do anything special for the [global] trunks section in the EXTENSION.CONF

Maybe you can show me some examples of how that looks.

Thanks

[jepps3]

In the [globals] section of the EXTENSIONS.CONF would this be acceptable:

SIPtrunk1=provider_1
SIPtrunk2=provider_2
SIPtrunk3=provider_3
SIPtrunk4=provider_4
SIPtrunk5=provider_5


And then have the trunks in SIP.CONF as this:

[provider_1]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=testSIPtrunk
secret=test
host=10.10.10.16
dtmfmode=inband
qualify=1000


[provider_2]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=testSIPtrunk
secret=test
host=10.10.10.16
dtmfmode=inband
qualify=1000


[provider_3]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=testSIPtrunk
secret=test
host=10.10.10.16
dtmfmode=inband
qualify=1000

And so on...............

Is this OK for a Failover solution?

[theorians]

Hi: I hope all is well.

I'm using 2 GoAutoDial Servers. I was having no issues with below mentioned failover settings. Call was automatically going to carrier2 but now I have got a new carrier which offers me better rate but it doesn't support my current failover. Please guide:

Dialers:
VERSION: 2.4-309a
BUILD: 110430-1642
Asterisk 1.4.39.1-vici

FailOvers:
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,n,Dial(SIP/Car1/${EXTEN:1},,tTor)
exten => _91XXXXXXXXXX,n,Dial(SIP/Car2/${EXTEN:1},,tTor)
exten => _91XXXXXXXXXX,n,Dial(SIP/Car3/${EXTEN:1},,tTor)
exten => _91XXXXXXXXXX,5,Hangup

this dial plan doesn't work but if I remove Car1 entry it works perfect. Please guide. Thanks