11-07-2007 06:03 AM - edited 03-15-2019 07:06 AM
I have the following topology
CME Site A -----IP WAN ------CME Site B
I would like to initiate a call off-net from a cell phone to CME Site A get a dial tone and call a off-net number at Site B
I have the Toll by pass working between site A and Site B, and got as far as calling from a cell phone at site A, and get dial tone, by not including the direct inward dial command for the pots number. I can dial extensions on CME Site A, but cannot dial anything else.
I know this is not the most effecient way to do this, does any one have any ideas.
Customer requirement is to be able to call from their cell phones in site A to a cell Phone at Site B, Using Site B CME Gateway.
Is this possible
11-07-2007 08:19 AM
Hi,
sure it's possible, it's just a matter of configuring the DP correctly.
For example, with the digit 9 followed by variable of fixed length,
site A conf
dial-peer voice XXX voip
destination-pattern 9.......
session-target ipv4:x.x.x.x
codec g711
Site B conf
dial-peer voice XXX voip
incoming called-number 9.......
codec g711
dial-peer voice ZZZ voip
destination-pattern 9.......
port YYY
hope this helps, please rtae post if it does!
11-07-2007 11:32 AM
ok i see what you are saying. 1st DP ZZZ at site B should be a pots DP, this was a typo...right?
as i mentioned i can dial a DID number from my cell into site A, and i get dial tone because i left out direct-inward-dial for that incoming called number.
From my cell phone dialing into site A and then trying to dial an extension at site B. I cannot dial an extension at siteB, and I have this VoIP DP in place, and it is working, as someone in site A can dial someone st site B.
Also I cannot dial any other numbers, except phones registered to site A CME, when i call into site A from my cell phone, using the method desribed above.
If I can call from my cell into site A, get a dial tone aand be able to ring an extension at site B, or even dial a local call out site A gateway then I move ahead and work on calling a off-net number at site B.
Am i making any sense?
Thanks
11-07-2007 12:59 PM
Hi,
yes DP ZZZ should be pots, a typo.
One should look at debug to understand why you can't call even site B phones. perhaps a codec mismatch.
I suggest you make all your voip DP with "session protocol sipv2" that is easy to debug, then enable "debug ccsip message" and "term mon" when trying the call.
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