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Configuring FXO port for inbound calls

Hi dear community,

I have a problem when configuring a fxo port in a ISR4331 for receiving inbound calls, I´m new in this kind of configurations.

I have this scenario:

I connect my fxo port to a telephone strip (you know a cable with only 2 pairs), and in theory we received this number 7282851300, next I tried to send this number to my CUCM, I configured an IP phone with only the last four digits (1300) and tried to make a call.

This is the configuration I used:

voice-port 0/1/0
ring number 3
signal groundStart
cptone MX
connection plar 1300
caller-id enable

I used connection plar command because a peer told me the gateway will create this number in order to send it to cucm because is an analog signal.

!
dial-peer voice 100 pots
description Incoming Calls
incoming called-number .
port 0/1/0
!
dial-peer voice 110 voip
description Calls To CUCM
destination-pattern 1300
session protocol sipv2
session target ipv4:10.1.130.247
dtmf-relay rtp-nte
codec g711ulaw
no vad

I activated the "debug voice ccapi inout" command in order to see if something is being sent to my gateway but when I make calls no messages are displayed, I introduced the "sh voice port summary" command and could see that port status change to ringing and connect adn ringing again, I suppose something is happening here.

I created a SIP trunk to connect CUCM and ISR4331.

As I said I´m new in this kind of configurations, so I don´t know if I am forgetting something, so I expect you can help me whit this problem.

Regards.

3 Replies 3

Hi,

Assuming that you completed your bind commands for SIP, try this config

voice translation-rule 1
rule 1 /.*\(....$\)/ /\1/
!
voice translation-profile incomingTo4
translate called 1
!
voice-port 0/1/0
translation-profile incoming incomingTo4
no connection plar 1300
!
no dial-peer voice 100 pots

Hi Mohhamed,

Thanks for your answer, this is the SIP configuration I used, would you consider it is enough for what I want to do?

voice service voip
ip address trusted list
ipv4 10.1.130.0 255.255.255.0
ipv4 10.1.130.247     ---------------- CUCM Subscriber
dtmf-interworking rtp-nte
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
registrar server expires max 600 min 60
!
voice class h323 1
h225 timeout tcp establish 1

Regards.

The above config is good just add it with the commands I mentioned earlier.  also add the following

Sip-ua

Retry invite 3

Timer trying 200