Hi All I have an issue with SIP SRST on a branch router, the branch in question has a single number coming into the gateway which is translated to 13500, this is sent to the CUCM's dial-peers and to a hunt pilot in CUCM.
If i set the 13500 as the pilot on the voice hunt-group, then the inbound call failed to go to the CUCM dial-peers and tried to go to the pilot on the voice hunt-group instead so i had to remove the 13500 pilot for calls to work. In SRST, if i added the 13500 back on the voice hunt-group then the inbound calls work fine and ring round the hunt group, to try and resolve this in both SRST and normal mode, i have tried to translate 13500 to 7016499 and set 7016499 as the pilot then added an alias to the voice register pool, however in SRST the call still does not get to the hunt pilot, if i change the translation for 13500 to go to the DN of one of the numbers of the phones in SRST this works fine. Any idea how I can get the pilot to recieve the inbound call? Relevant config below ! voice register global system message Backup SRST Active max-dn 35 max-pool 35 ! voice register pool 1 translation-profile outgoing SIP-TRANSLATE id network 10.5.16.0 mask 255.255.255.0 alias 1 13500 to 7014699 voice-class codec 1 ! ! voice hunt-group 1 parallel final 7016450 list 7016411,7016412,7016413,7016430,7016450 timeout 120 pilot 7016499 ! ! ! ! voice translation-rule 100 rule 1 /290290/ /13500/ rule 2 /241430/ /13500/ ! ! voice translation-rule 110 rule 1 /13500/ /7016499/ ! ! voice translation-profile SIP-TRANSLATE translate called 110 ! ! dial-peer voice 3 voip tone ringback alert-no-PI description *** FIRST CCM NODE *** destination-pattern 13... progress_ind setup enable 3 progress_ind alert enable 8 session target ipv4:10.10.10.10 voice-class codec 1 voice-class h323 1 dtmf-relay h245-alphanumeric fax rate disable ip qos dscp cs3 signaling no vad ! Regards Nick
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