kavinagpur
asked on
How to Disable Outgoing Call from FXO Port
Hello Expert,
We have two Analog line 1st plugged in Voice-port 0/1/0 & 2nd is Voice-port 0/1/1
Im just thinking here, We need to use Voice-port 0/1/0 for all outbound calls at this stage, as we don’t get charged any extra for calls on this number, but we do on the Voice-port 0/1/1
so I want to disable all Outgoing call from Voice-port 0/1/1
When anyone will make a call from any extantion all outgoing call should make from Voice-port 0/1/0, but it should not affect on incoming calls.
I'd really like to get these issues sorted if anyone can point me in the right direction!
We have two Analog line 1st plugged in Voice-port 0/1/0 & 2nd is Voice-port 0/1/1
Im just thinking here, We need to use Voice-port 0/1/0 for all outbound calls at this stage, as we don’t get charged any extra for calls on this number, but we do on the Voice-port 0/1/1
so I want to disable all Outgoing call from Voice-port 0/1/1
When anyone will make a call from any extantion all outgoing call should make from Voice-port 0/1/0, but it should not affect on incoming calls.
I'd really like to get these issues sorted if anyone can point me in the right direction!
voice-port 0/1/0
output attenuation -6
cptone BR
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 202
caller-id alerting dsp-pre-allocate
voice-port 0/1/1
output attenuation -6
cptone BR
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 206
caller-id alerting dsp-pre-allocate
ephone-dn 11 dual-line
number 202
label 202
description India Nagpur
name India Nagpur
call-forward busy 206
call-forward noan 206 timeout 20
ephone-dn 15 dual-line
number 206 no-reg primary
label 206
description Lindsay Clark
name Lindsay Clark
call-forward busy 0411069447
call-forward noan 0411069447 timeout 10
!
dial-peer voice 4 pots
service stcapp
port 0/0/3
no sip-register
!
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 400
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 1000 voip
description gg gg
service aa
destination-pattern 1000
voice-class codec 1
session target ipv4:10.1.1.1
incoming called-number 1000
dtmf-relay rtp-nte
!
dial-peer voice 100 voip
dtmf-relay h245-alphanumeric
!
dial-peer voice 2 pots
service stcapp
port 0/0/1
no sip-register
!
dial-peer voice 3 pots
service stcapp
port 0/0/2
no sip-register
!
dial-peer voice 999 pots
incoming called-number .
direct-inward-dial
!
dial-peer voice 50 pots
destination-pattern 0000
port 0/1/0
forward-digits 3
no sip-register
!
dial-peer voice 1 pots
destination-pattern .T
port 0/1/0
!
!
no dial-peer outbound status-check pots
sip-ua
ASKER
We have CIsco UC520 Voip Router & I have used cisco
Thanks
Thanks
Use CCA to configure this - much simpler, the UC520 is designed to be configured with it.
Create a PSTN trunk group under Dial Plan/Outgoing with only 0/1/0 in it, and then adjust the Outgoing call plans to use this trunk (under configure priority.
Obviously it means you will be restricted to a single outgoing call. You may be better off setting the Trunk to use the lines sequentially (which it is by default), so it will try and use 0/1/0 first and only use 0/1/1 if it is busy.
Create a PSTN trunk group under Dial Plan/Outgoing with only 0/1/0 in it, and then adjust the Outgoing call plans to use this trunk (under configure priority.
Obviously it means you will be restricted to a single outgoing call. You may be better off setting the Trunk to use the lines sequentially (which it is by default), so it will try and use 0/1/0 first and only use 0/1/1 if it is busy.
ASKER
I already configured dial peer Under voice-port, i don't have that much knowledge about voip, i did it with the help of EE, i attached my dial peer along with this comment, can u tell me as per my current dial peer
All calls will be from whence ?
It will be a great if u will give me a solution of below link
https://www.experts-exchange.com/questions/25889261/voice-port-is-show-OFF-HOOK-after-disconnect-previous-call.html
Thanks
Vikrant
All calls will be from whence ?
It will be a great if u will give me a solution of below link
https://www.experts-exchange.com/questions/25889261/voice-port-is-show-OFF-HOOK-after-disconnect-previous-call.html
Thanks
Vikrant
dial-peer voice 4 pots
service stcapp
port 0/0/3
no sip-register
!
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 400
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 1000 voip
description gg gg
service aa
destination-pattern 1000
voice-class codec 1
session target ipv4:10.1.1.1
incoming called-number 1000
dtmf-relay rtp-nte
!
dial-peer voice 100 voip
dtmf-relay h245-alphanumeric
!
dial-peer voice 2 pots
service stcapp
port 0/0/1
no sip-register
!
dial-peer voice 3 pots
service stcapp
port 0/0/2
no sip-register
!
dial-peer voice 999 pots
incoming called-number .
direct-inward-dial
!
dial-peer voice 50 pots
destination-pattern 0000
port 0/1/0
forward-digits 3
no sip-register
!
dial-peer voice 1 pots
destination-pattern .T
port 0/1/0
can you paste more of your config? not enough information for us to help you with.
ASKER
Thanks for your reply,
I want to use voice-port(0/1/0) for outgoing & incoming call But I want to use voice-port(0/1/1) are only for incoming, if anyone will make call from any extantion they should make from voice-port(0/1/0)
I used as per above on my router but let me know if there is some wrong
Thanks
New.txt
I want to use voice-port(0/1/0) for outgoing & incoming call But I want to use voice-port(0/1/1) are only for incoming, if anyone will make call from any extantion they should make from voice-port(0/1/0)
I used as per above on my router but let me know if there is some wrong
Thanks
New.txt
I see two outbound FXO dialpeers -
dial-peer voice 50 pots
destination-pattern 0000
port 0/1/0
forward-digits 3
no sip-register
!
dial-peer voice 1 pots
destination-pattern .T
port 0/1/0
!
I don't see any outgoing dialpeers using 0/1/1 so it should not dial out of 0/1/1. What happens when you dial out currently? Does it dial out of 0/1/1 also?
dial-peer voice 50 pots
destination-pattern 0000
port 0/1/0
forward-digits 3
no sip-register
!
dial-peer voice 1 pots
destination-pattern .T
port 0/1/0
!
I don't see any outgoing dialpeers using 0/1/1 so it should not dial out of 0/1/1. What happens when you dial out currently? Does it dial out of 0/1/1 also?
ASKER
Hi Pro4ia,
Thanks for your reply,
when anybody make calls on 08 9377 3097 (voice-port0/1/1) means All incoming call which are coming on 08 9377 3097 (voice-port0/1/1) they diverted to 206 (ext.). If anybody are unable to receive call on 206 within 15 sec. then call Forward to my Mobile from 08 9377 0540 (voice-port0/1/0)
I know All is going ok, But How to confirm whether it forward from which port
Thanks
Thanks for your reply,
when anybody make calls on 08 9377 3097 (voice-port0/1/1) means All incoming call which are coming on 08 9377 3097 (voice-port0/1/1) they diverted to 206 (ext.). If anybody are unable to receive call on 206 within 15 sec. then call Forward to my Mobile from 08 9377 0540 (voice-port0/1/0)
I know All is going ok, But How to confirm whether it forward from which port
Thanks
Your best bet is to monitor the cisco and make an outgoing call which should lock up port 0/1/0.
Then using another phone place another outgoing call and see whether you get an error/busy type of issue i.e. no available lines or your call makes it out. You can check the cisco for status of the lines.
Then using another phone place another outgoing call and see whether you get an error/busy type of issue i.e. no available lines or your call makes it out. You can check the cisco for status of the lines.
Oh, the forwarding might have to go through the 0/1/1 since this is the port through which the call came in and might explain why the line gets tied up. Not sure whether when a call is forwarded whether the connection from the caller to the end recipient has to remain as it came in or whether it is redirected and the 0/1/1 line becomes free once the forward is complete i.e. the final remote recipient picked up the line.
ASKER CERTIFIED SOLUTION
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ASKER
Hello Pro4ia,
Ok Thanks for helping, really great
Ok Thanks for helping, really great
ASKER
Hello Pro4ia,
can you help me on below issue
https://www.experts-exchange.com/questions/25831317/Cisco-7921-IP-phone-unable-to-connect-router-on-Wifi.html
can you help me on below issue
https://www.experts-exchange.com/questions/25831317/Cisco-7921-IP-phone-unable-to-connect-router-on-Wifi.html
ASKER
Great
You may need to define a context for the line as inbound only