05-10-2017 10:06 AM - edited 03-17-2019 10:17 AM
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.
05-10-2017 10:39 PM
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
05-11-2017 09:38 AM
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.
05-11-2017 10:19 AM
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
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