Have SRST configured to use alias commands as a bit of a hunt group. Works fine no matter how many extensions I put in, it flips through them exactly as I'd expect.
It fails however if I list a CUE extension which points to my AA. If I have ext 123, 124, 777123, with the last being my AA, the call fails when it gets there (so from the user it rings the first 2, then drops).
In SRST, I can reach my CUE/AA just fine, so I know the dial-peer is good.
translation points to 6565
alias 1 6565 to 6564 cfw 777001 timeout 15 alias 2 777001 to 6566 cfw 777002 timeout 15 alias 3 777002 to 770060
dial-peer voice 22 voip description Local NM-CUE (SRST) Greeting Management System destination-pattern 770060 session protocol sipv2 session target ipv4:10.164.48.3 dtmf-relay sip-notify codec g711ulaw no vad
Is it perhaps the SIP protocol that the Alias command cant use?
SIP traces provide key information in troubleshooting SIP Trunks, SIP
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Please find the attached HTML document, download and open it on your PC.
This provides an easy to use form where you simply answer a few
questions and it will render the proper jabber-config.xml file for you
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