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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>ITSP failover for PRI<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Hello All,<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>We’re using an Asterisk based SIP-T1 trunking gateway and would like to implement failover between two ITSPs.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>If we connect a soft phone to the gateway with the following lines in extensions.conf failover works. <o:p></o:p></p><p class=MsoNormal>If one ITSP is unavailable the call flow cascades to the second ITSP and connects with audio.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>[outgoing]<o:p></o:p></p><p class=MsoNormal>exten => _1NXXNXXXXXX,1,NoOp(${CALLERID(all)="" <>}) exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@ITSP1)<o:p></o:p></p><p class=MsoNormal>exten => _1NXXNXXXXXX,3,Dial(SIP/${EXTEN}@ITSP2)<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>If we attempt calls from the PBX over the PRI connected to the Astlinux Gateway the calls connects, but there is no audio.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> This is what we see:<o:p></o:p></p><p class=MsoNormal> ITSP1:<o:p></o:p></p><p class=MsoNormal>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<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>SIP/ITSP1-000000c6 is circuit-busy (This result is because the ITSP1 account is blocked for testing)<o:p></o:p></p><p class=MsoNormal>Everyone is busy/congested at this time (1:0/1/0)<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> ITSP2:<o:p></o:p></p><p class=MsoNormal>Executing [XXXXXX@outgoing:3] Dial("DAHDI/22-1", "SIP/XXXXXX@ITSP2") in new stack Called XXXXXX@ITSP2<o:p></o:p></p><p class=MsoNormal>SIP/ITSP2-000000c7 is making progress passing it to DAHDI/22-1<o:p></o:p></p><p class=MsoNormal>SIP/ITSP2-trunk-000000c7 answered DAHDI/22-1<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Can someone please make suggestions or point us in the right direction to resolve this no audio issue?<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Thank you<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='font-family:"Tahoma","sans-serif";color:#0037E6'><o:p> </o:p></span></p></div></body></html>