Access to the PSTN from CUCM

[Labanelamia]

can someone help me please to configure an access form the ip phone ( CUCM (192.168.1.101)) to the PSTN

CUCM h323 ----------- (H323 gateway ) FXO PORT 0/0/0 -----------------------PSTN
                                               CISCO 2921
         1002  ---------------->       H323 gatway         ----------------------------->  0xxxxxxxx

                                                voice service voip
                                                allow-connections h323 to h323

                                                interface Loopback0
                                                ip address 20.20.20.20 255.255.255.255
                                                h323-gateway voip interface
                                               h323-gateway voip bind srcaddr 20.20.20.20


                                              voice-port 0/0/0
                                              supervisory disconnect dualtone mid-call
                                               pre-dial-delay 2
                                               disc_pi_off
                                               cptone FR
                                               timeouts call-disconnect 3
                                               timeouts wait-release 3
                                               connection plar opx 1002
                                               impedance complex2
                                               caller-id enable
                                               !

                                               dial-peer voice 4 voip
                                               destination-pattern 1002
                                               session target ipv4:192.168.1.101
                                                voice-class codec 1
                                               !

The call from pstn to the ip phone is ok.
what route pattern i have to add in CUCM to reach pstn (all number beginning with 0)
and what dial-peer i have to add in gateway ??

more details are in the file oploaded.

thank you.

[Rajan · Accepted solution]

Route pattern can be 0xxxxxxxx pointing to H323 gateway or create a route group, route list and point the RL in the Route pattern where RG has H323 gateway

use the below dial-peer in H323 gw to route the call out

dial-p v x pots
destination-pattern  0xxxxxxxx
incoming called number .T
forward-digits all
port 0/0/0

HTH

Rajan

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[Rajan · Accepted solution]

my bad Correction in dial-peer

dial-p voice x pots
destination-pattern  0T (or) 0........
incoming called number .T
forward-digits all
port 0/0/0

[Rajan · Accepted solution]

if prefix 0 need not be sent to PSTN, then tweak the forward-dgits command accordingly like 8 or whatever digit you will send out

[Ayodeji Okanlawon · Accepted solution]

Firs of all before you add a route pattern you need to configure a route group. Then add the h323 gateway to this new route group. You then need to configure a route list, and add the route group you configured into the route group.

Once all this is done, you will then proceed to configure your route pattern. However before you configure your route pattern you may need to configure partiton and css, should you need to implement calling privilieges.

So you create a partition, call it something meaningful eg PT-PSTN or PSTN_PT

Then create your rooute pattern. Your route pattern depends on your users dialling behaviour. Some users diall access code before they dial their numbers..The access code can be 9 or 0..If yours dont..

then just configure a Route pattern (RP) that matches the different numbers to be dialled..

You will need to know the different types of numbers that needs to be dialled.

eg local. long distance, international etc..

Eg..

012XXXXXXXX will match all numbers that begin with 012. You will then assign this RP to the RL you created.

On your gateway you will need to configure inbound dial-peer for calls coming in from cucm and also outbound dial-peers to the PSTN

eg..

dial-peer voice 10 voip

incoming called .

dtmf-r h245-alpha

voice-class codec 1

no vad

dial-peer voice 11 p

destination-p 0T

prefix 0

port 0/00

[Labanelamia]

I have 21 phones (8945 )and B6000 (cucm 10 and UC), I need to enable vidéo conference, can you give me a simple way to configuré that;
Thanks for your help.