Hi,
Recently I migrated one of our branches to CUCM/SRST deployment. Just like any other CUCM/SRST deployment , I expected to finish smoothly. During the migration the IT personal discovered they had three different Departments with three diffrent Phone System working on this branch.
(Someone didnt do the survey properly, I guess. )
Which meant resulted me migrating these systems on the single Voice gateway avaiable for the branch.
Each department had their own POTS lines. This setup had to be maintained after the migration. i.e lines for the department had to be only used by their respective department.
While this was easy with CUCM and voice translation rules , the problem I am facing is with SRST Mode, I am unable to succeed.
What have already been deployed is Calls from specific extensions to be forwarded to specific FXO port while the system is registered with CUCM.
For example,
| dial-peer voice 999 voip |
| description ### 6042210 #### |
| answer-address 6042210 | | voice-class h323 1 | | dtmf-relay h245-alphanumeric | | codec g711ulaw | | no vad |
|
| dial-peer voice 998 pots | | port 0/0/1 |
|
| description *** MOBILE-CALLS *** |
| destination-pattern 7905........ |
| forward-digits 10 |
Voice translation rule used :
voice translation-rule 6042210
rule 1 /^9/ /79/
voice translation-profile XYZ-LINES
translate called 60422210
Everything works smooth with CUCM but when the system goes to SRST it does not work. Which is how it should behave. The problem is with SRST mode the ephone create their own dial-peer. The possiblity I am left with is to use Cor-List under call-manager-fallback or translation rules.
With Both I am unable to succeed, cannot seem to figure out a standard workable resolution.
I hope someone can help me out on it.
Thanks,
Mazin
