Remote Office with H323 Voice Gateway + FXO Ports + CUCM

[VOIP323ps]

I am trying to setup a new voice gateway for the remote office and it will be using SRST later on. The voice gateway is a CISCO ISR 2911 with couple of FXO ports (the remote office will have 5 analog lines with totally different numbers other than the area code). It is a H323 voice gateway as the same  gateway as the head office. Head office is currently using H323 voice gateway 2851 with CUCM 8.6 and need to dial 8 to dial out to local PSTN. Everything works fine in the head office and all internal extensions are in 3 digital xxx formats from ext 100 to ext 700.

Now I have a test analog line with phone number 1(234)567-8900 and need to dial 9 to connect to the local PSTN. The analog phone line is currently connected to the 2911 FXO port 0/0/0. I am trying to do some basic setup and the first thing I want to do is to tie the analog phone number 1(234)567-8900 to CUCM ext 500 so that ext 500 can receive phone calls from the outside and also dial out locally. I did a lot of research and reading and I know there are so many ways to do it such as creating translation-rule/ plar opx immediate on the gateway directly or translation patterns directly on CUCM. I tired it all but nope of the configurations seem working after all. I have a feeling I might miss something on the CUCM side.

Can anyone tell me what is the best practice or the easier way to set that up with my current scenario?

Here is what I tried so far:

On CUCM

• Setup New Gateway under Device -> Gateway
• Setup New Calling Search Space under Call Routing -> Class of Control -> Call Search Space
• Setup New Partition under -> Call Routing -> Partition
• Pointed x500 to the new Route Partition and calling search space

On Voice Gateway 2911

voice-port 0/0/0
description line 2345678900
caller-id enable

dial-peer voice 1 pots
description *** For incoming call matching ***
incoming called-number.
direct-inward-dial
port 0/0/0

dial-peer voice 9 ots
description *** for outgoing 9 Traffic to PSTN ***
destination-pattern 9T
tone reingback alert-no-PI
direct-inward-dial
port 0/0/0

dial-peer voice 11 voip
description *** CUCM Peer ***
incoming called-number.
Voice-class codec 1
Voice-class h323 1
Session target ipv4: 10.0.0.2
dtmf-relay h245-signal h245-alphanumeric
ip qos dscp cs3 signaling
no vad

My goal for now:

• connect extension 500 to 1(234)567-8900 so if someone call 1(234)567-8900 it will ring the extension 500
• extension 500 can dial 9 and it can call outside phone number such as 1(213)567-8888 etc
• internal extension calls are working fine since I added calling search space from the head office to the new calling search space as extension 500 can call extension 600 etc

Thank you very much in advance!!

[VOIP323ps]

Islam,

Thanks!!

Remember the follwoing:

-  8 is the number we need to dial for all exteranl calls (Internal Extension are in 100~700 range)

- 9 is the PSTN number for all exteranl calls.

- 9 PSTN number is only for my test line environment right now. The acutal production branch office does not need to dial 9 to get out to PSTN.

Scenario 1 ----------> 1(234)567-8900

1. Dial 8 and 912345678900

2. Phone shown 8912345678900 while dialing

3. Phone shown only 9123456 when it is connected

Scenario 2 -----------> 323-5678

1. Dial 8 and 93235678

2. Phone shown 893235678 while dialing

3. Phone shown only 9323567 when it is connected

Please see below for my configurations.

# dial-peer voice 9 pots
# description *** For Outgoing 9 Traffic to PSTN ***
# destination-pattern 1..........
# tone ringback alert-no-PI
# forward-digits 10
# port 0/0/0
CUCM Route Pattern 81.[2-9]XXXXXXXXX

Provide Outside Dial Tone/discard digits choose predot

dial-peer voice 10 pots
# description *** For 7 digit local calls ***
# destination-pattern [2-9]......
# tone ringback alert-no-PI
# forward-digits 7
# port 0/0/0
CUCM Route Pattern 8.[2-9]XXXXXX

Provide Outside Dial Tone/discard digits choose predot

# dial-peer voice 11 pots
# description *** For 911 Calls ***
# destination-pattern 911
# tone ringback alert-no-PI
# forward-digits 3
# port 0/0/0
CUCM Route Pattern 8.911

Provide Outside Dial Tone/discard digits choose predot

# dial-peer voice 12 pots
# description *** For 011 International Calls ***
# destination-pattern 011T
# no digit-strip
# tone ringback alert-no-PI
# port 0/0/0
CUCM Route Pattern 8.011!

Provide Outside Dial Tone/discard digits choose predot

[islam.kamal]

hello

Again , scinario 1 & 2 incorrect 89 will match local calls and you will get wrong dialed numbers , other are perfect good work. I am sure that you have somethig start with 9 100% , look we can avoid this headache , and you can use "7 or 88" as outgoing code instead of "9". Please do not forget to rate all useful information and correct answers to be useful for users in future.

Thanks

please rate all useful information

[VOIP323ps]

Islam,

Thanks again! I'm ok with the dial pear and RT now. I do have more question. I will need to add 4 more lines to the gateway and should I do the follwoing?

1. New x2 translation-rule for each line?

2. New x2 voice translation-profile for each line?

3. New dial-peer for each lines? such as

dial-peer voice 11 voip

destination-pattern 500$

dial-peer voice 22 voip

destination-pattern 501$

dial-peer voice 23 voip

destination-pattern 502$

and so on?

-------------------------------------------------------------------

Phone 2345678900 to ext500 port 0/0/0
Phone 2346789000 to ext501 port 0/0/1
Phone 2344567890 to ext502 port 0/0/2
Phone 2343214789 to ext503 port 0/0/3
Phone 2342345678 to ext504 port 0/0/4
------------------------------------------
voice translation-rule 1
rule 1 /2345678900/ /500/

voice translation-rule 2
rule 2 /500/ /2345678900/

voice translation-rule 3
rule 3 /2346789000/ /501/

voice translation-rule 4
rule 4 /501/ /23456789000/

voice translation-rule 5
voice translation-rule 6
voice translation-rule 7
voice translation-rule 8
voice translation-rule 9

and so on
-------------------------------------------
voice translation-profile INCOMING_8900
translate called 1

voice translation-profile OUTOGING_8900
translate calling 2

voice translation-profile INCOMING_9000
translate calling 3

voice translation-profile OUTOGING_9000
translate calling 4

and so on
--------------------------------------------
voice-port 0/0/0
translation-profile incoming INCOMING_8900
translation-profile outgoing OUTGOING_8900
description line 2345678900
caller-id enable
connection plar opx 500
trunk-group FXO1

voice-port 0/0/1
translation-profile incoming INCOMING_9000
translation-profile outgoing OUTGOING_9000
description line 2346789000
caller-id enable
connection plar opx 501
trunk-group FXO1

voice-port 0/0/2
voice-port 0/0/3
voice-port 0/0/3

and so on
----------------------------------------------

dial-peer voice 11 voip
description *** CUCM Peer ***
destination-pattern 500$
Voice-class codec 1
Voice-class h323 1
Session target ipv4: 10.0.0.2
dtmf-relay h245-signal h245-alphanumeric
ip qos dscp cs3 signaling
no vad

dial-peer voice 21 voip
description *** CUCM Peer ***
destination-pattern 501$
Voice-class codec 1
Voice-class h323 1
Session target ipv4: 10.0.0.2
dtmf-relay h245-signal h245-alphanumeric
ip qos dscp cs3 signaling
no vad

dial-peer voice 22 voip
description *** CUCM Peer ***
destination-pattern 502$
Voice-class codec 1
Voice-class h323 1
Session target ipv4: 10.0.0.2
dtmf-relay h245-signal h245-alphanumeric
ip qos dscp cs3 signaling
no vad

and so on

[islam.kamal]

Hello

Q:-connect extension 500 to 1(234)567-8900 so if someone call 1(234)567-8900 it will ring the extension 500          

voice-port 0/0/0

caller-id enable

connection plar opx 500  or  connection plar opx immediate 500

Q-extension 500 can dial 9 and it can call outside phone number such as 1(213)567-8888 etc

dial-peer voice 9 pots
destination-pattern 9.T
tone reingback alert-no-PI
direct-inward-dial
port 0/0/0

3rd question is not clear for me .

thanks

please rate all useful information