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@outgoing:1] NoOp("DAHDI/22-1", """ <XXXXXX>") in new stack Executing
[XXXXXX@outgoing:2] Dial("DAHDI/22-1", "SIP/XXXXXX@ITSP1") in new stack
Called XXXXXX@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@outgoing:3] Dial("DAHDI/22-1", "SIP/XXXXXX@ITSP2") in new
stack Called XXXXXX@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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