ITSP failover for PRI

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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”, “”” “) 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

4 thoughts on - ITSP failover for PRI

  • 2011/6/19 Claude Hayn

    What about incoming calls ?
    Do you have a way to have calls that normally comes from ITPS1 to comes from
    ITSP2 ?

  • 2011/6/20 Alex Balashov

    Yes, that’s what I thought but you never know 😉
    (Maybe SS7 offers such redundancy but I’ve got no experience of any king in
    this domain).

  • No audio is usually a NAT issue. Verify you have the proper NAT settings on
    your ITSP2 account settings and try again. A SIP debug trace would be very
    useful for debugging this (sip set debug on on the asterisk CLI or “tcpdump