tks, do i need to add prefix 55 in pots?

Actually I think your R2 config is incomplete. No R2 config will ever work with "national defaults" only.

I will try to collect an applicable example. The TAC should have some for the various countries too.

tks a lot.

Finally, we clarified that local PSTN send only 3 digits to us and our extension range is still 5XXX.

So, we successfully deployed the E1 R2 in vg and ccm. I would like to share the vg configuration as below.

================================

!

controller E1 0/0/0

framing NO-CRC4

ds0-group 0 timeslots 1-15 type r2-digital r2-compelled ani

cas-custom 0

country philippines use-defaults

!

dial-peer voice 401 voip

description CCM-subscriber

preference 1

destination-pattern ...

session target ipv4:

dtmf-relay h245-alphanumeric

codec g711alaw

ip qos dscp cs5 media

no vad

!

dial-peer voice 402 voip

description CCM-publisher

preference 2

destination-pattern ...

session target ipv4:

dtmf-relay h245-alphanumeric

codec g711alaw

ip qos dscp cs5 media

no vad

!

dial-peer voice 400 pots

destination-pattern [0-9]..

direct-inward-dial

port 0/0/0:0

!

================================

Since this trunk is for inbound only, so we haven't configure for outbound call.

In CCM, we created H323 gateway and enter vg ip addr in Domain Name field. In Inbound Calls -> Prefix DN, we enter 5 so that 5XXX is ensure to CCM. Besides, an approiate CSS is entered in the filed of CSS. Again, we haven't configure Outbound Calls.

I hope this can help others.

i've got another problem. We've found that there is dealy ring issue. Phone rings after callers hear 11 sec ringback tone. I've add "dnis-digits min 1 max 10", but it can only reduce 3 sec, i.e. dealy ring is 8 sec. Is there any other parameters can be set to further reduce it?

Since only 3 digits come in from the Telco, then use dnis-digits min 1 max 3 - this will route the call once 3 digits have come in.

Your dial peer configs need some find tuning as well. Currently you have the following dial peers -

!

dial-peer voice 401 voip

description CCM-subscriber

preference 1

destination-pattern ...

session target ipv4:

dtmf-relay h245-alphanumeric

codec g711alaw

ip qos dscp cs5 media

no vad

!

dial-peer voice 402 voip

description CCM-publisher

preference 2

destination-pattern ...

session target ipv4:

dtmf-relay h245-alphanumeric

codec g711alaw

ip qos dscp cs5 media

no vad

!

dial-peer voice 400 pots

destination-pattern [0-9]..

direct-inward-dial

port 0/0/0:0

!

The two VOIP dial peers have destination patterns of '...' which would cover the number range of 000 to 999. The POTS dial peer has a destination pattern of [0-9].., which covers the range of 000 to 999 as well. So all dial peers cover the same number range !

Using a destination pattern of '...' is bad - it is too generic and will catch ANY call coming in from the IP or POTS side. Always make your destination patterns specific to the number range you are using on the CallManager.

OK ... now this is just a theory and I would need debug traces to prove it, but since you have preference 1 and preference 2 on the VOIP dial peers, the default preference on the POTS dial peer will be 0, which means a call that comes in from the PSTN is very likely to set up a new POTS leg and the router will try to re-route the call straight back to the PSTN. Since you will only send three digits back out to the telephone exchange, chances are it will fail after the interdigit timeout and then hunt to the VOIP dial peers.

You said the CCM has extensions in the number range of 5XXX. The best thing to do here is to translate the numbers on the gateway and send the full number range to the CCM so you don't get this number overlap. You should also set up the dial peer/CCM redundancy as per this document -

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080094852.shtml

Due to the size of this reply, refer to the next post on what to do next !!!

Part 2 - following on from the last post !!!

Try adding the following config -

voice class h323 1

h225 timeout tcp establish 3

translation-rule 1 <---- prefix 5 to the incoming DNIS

rule 1 ^.* 5

voice-port 0/0/0:0

translate called 1 <---- apply the translation rule to the voice port called number

dial-peer voice 401 voip

description CCM-subscriber

preference 1 <--- first choice dial peer

voice-class h323 1 <---- 3 second timeout for calls to the Sub

destination-pattern 5... <--- 4 digit destination pattern in range 5XXX

session target ipv4:

dtmf-relay h245-alphanumeric

codec g711ulaw <---- use G711ulaw on IP side to ensure maximum interworking to IP devices

ip qos dscp cs5 media

no vad

!

dial-peer voice 402 voip

description CCM-publisher

preference 2 <---- 2nd choice dial peer

destination-pattern 5...

session target ipv4:

dtmf-relay h245-alphanumeric

codec g711ulaw

ip qos dscp cs5 media

no vad

!

no dial-peer voice 400 <---- remove dial peer 400

!

dial-peer voice 555 pots <---- dummy dial peer to enable DID

incoming called-number .

direct-inward-dial

port 0/0/0:0

For you outgoing calls, change the CCM route pattern so that you use an access code of 9, for example 9.! (or make individual specific route patterns)

click on provide outside dial tone (if not already enabled)

Under called party transforms, set it as discard digits and prefix digits

This means when a user makes an outside call, they will dial 9 , and this full digit string will be sent to the gateway.

On the H323 gateway, change your outgoing POTS dial peer to be this -

!

dial-peer voice 400 pots

destination-pattern 9

port 0/0/0:0

!

This will match on the call coming in from the CCM. The leading 9 of the digit string will match on the configured destination pattern and it will be stripped off, so the rest of the numbers the user dialed will be sent out to the PSTN.

Now if all this does not help , do the following -

turn off all debuging (undebug all)

set the timestamps to the msec (in config mode add 'service timestamps debug datetime localtime msec')

enable 'debug vpm signal'

make an inwards call and log the debugs

Turn off all debugs

Enable 'debug voip vtsp all' and make another inwards call and log these debugs.

Paste each file here, along with your full gateway configuration.

finally, i can fix the delay ring problem to 2 sec. pls find the following config

=================================

!

voice class h323 1

h225 timeout tcp establish 3

!

!

!

voice translation-rule 1

rule 15 // /5/

!

!

voice translation-profile inbound

translate called 1

!

!

controller E1 0/0/0

framing NO-CRC4

ds0-group 0 timeslots 1-15 type r2-digital r2-compelled ani

cas-custom 0

country philippines

dnis-digits min 1 max 3

!

voice-port 0/0/0:0

translation-profile incoming inbound

!

!

!

dial-peer voice 402 voip

description CCM-subscriber

preference 1

destination-pattern 5...

voice-class h323 1

session target ipv4:165.46.210.21

dtmf-relay h245-alphanumeric

codec g711ulaw

ip qos dscp cs5 media

no vad

!

dial-peer voice 400 pots

incoming called-number .T

direct-inward-dial

port 0/0/0:0

!

!

dial-peer voice 404 voip

description CCM-publisher

preference 1

destination-pattern 5...

voice-class h323 1

session target ipv4:165.46.67.21

dtmf-relay h245-alphanumeric

codec g711ulaw

ip qos dscp cs5 media

no vad

!

=================================