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!

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

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kavinagpurAsked:
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arnoldCommented:
Which Application are you using Asterisk, Cisco, OpenPBX, etc.?

You may need to define a context for the line as inbound only
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kavinagpurAuthor Commented:
We have CIsco  UC520 Voip Router & I have used cisco

Thanks
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Adam PedderDirectorCommented:
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.
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kavinagpurAuthor Commented:
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
http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/VoIP/Q_25889261.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

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Pro4iaCommented:
can you paste more of your config?  not enough information for us to help you with.
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kavinagpurAuthor Commented:
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
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Pro4iaCommented:
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?
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kavinagpurAuthor Commented:
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

0
arnoldCommented:
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.
0
arnoldCommented:
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.
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Pro4iaCommented:
kavinagpur

easiest way to tell whether or not 0/1/1 is being used to make outbound calls is to look at the LED of the FXO port 0/1/1 on the UC520 device while making an outbound call.  Active port lights up green so when you make outbound calls and you only see 0/1/0 lighting up, you know 0/1/1 is not being used.

if there is no outbound dial-peer for that port, it simply cannot dial out of it.

you can also run a "debug voip ccapi inout" which will tell you when you make an outbound call, which dial-peer it uses.  i'm going to assume that you will always see outbound dial-peer 1 there.
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kavinagpurAuthor Commented:
Hello Pro4ia,


Ok Thanks for helping, really great
0
kavinagpurAuthor Commented:
0
kavinagpurAuthor Commented:
Great
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