[asterisk-users] ITSP failover for PRI

[Claude Hayn]

ITSP failover for PRI

 

Hello All,

 

We're using an Asterisk based SIP-T1 trunking gateway and would like to
implement failover between two ITSPs.

 

If we connect a soft phone to the gateway with the following lines in
extensions.conf failover works. 

If one ITSP is unavailable the call flow cascades to the second ITSP and
connects with audio.

 

[outgoing]

exten => _1NXXNXXXXXX,1,NoOp(${CALLERID(all)="" <>}) exten =>
_1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@ITSP1)

exten => _1NXXNXXXXXX,3,Dial(SIP/${EXTEN}@ITSP2)

 

If we attempt calls from the PBX over the PRI connected to the Astlinux
Gateway the calls connects, but there is no audio.

 

           This is what we see:

            ITSP1:

Accepting call from 'XXXXXX' to 'XXXXXX' on channel 0/22, span 1 Executing
[XXXXXX at outgoing:1] NoOp("DAHDI/22-1", """ <XXXXXX>") in new stack Executing
[XXXXXX at outgoing:2] Dial("DAHDI/22-1", "SIP/XXXXXX at ITSP1") in new stack
Called XXXXXX at ITSP1

 

SIP/ITSP1-000000c6 is circuit-busy     (This result is because the ITSP1
account is blocked for testing)

Everyone is busy/congested at this time (1:0/1/0)

 

         ITSP2:

Executing [XXXXXX at outgoing:3] Dial("DAHDI/22-1", "SIP/XXXXXX at ITSP2") in new
stack Called XXXXXX at ITSP2

SIP/ITSP2-000000c7 is making progress passing it to DAHDI/22-1

SIP/ITSP2-trunk-000000c7 answered DAHDI/22-1

 

Can someone please make suggestions or point us in the right direction to
resolve this no audio issue?

 

Thank you

 

 

 

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