ITSP failover for PRI

Home » Asterisk Users » ITSP failover for PRI
Asterisk Users 4 Comments

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.


exten => _1NXXNXXXXXX,1,NoOp(${CALLERID(all)=”” <>}) exten =>

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:


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

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)


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