hi Claude
are there nat router between the astlinux gateway to yours ISTPs?
Graziano
Il 18/06/2011 20.31, Claude Hayn ha scritto:
> ITSP failover for PRI
>
> Hello All,
>
> We're using Astlinux as a 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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--
Graziano Brioschi
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