asked on

need to add area code when dialing pots dial peers not working anymore

Hello

The PSTN have now added new area codes and we must now add the area code when dialing. Since then I am unable to make outbound calls locally. I keep getting a message to dial the area code but when I do it the message received is "the call cannot be completed as dialed".
I can however still receive calls no problem.
I can also place long distance calls no problem but unable to place local calls to PSTN.
As for over the SIP trunk it's down for the moment so it probably doesn't work either. If I can at least fix the PSTN side it would be ok for the moment.

We used to dial only 7 digits but now have to include the area code so we now have to dial 10digits to place local calls.
ex: used to dial: 365-6667 but now have to dial 705-365-6667

Here are my configurations for my dial peers which were working no problem before:

!
dial-peer voice 6097 pots
description [-[ telephone_fax_machine ]-]
preference 2
destination-pattern 6097
port 0/0/0
no sip-register
!
dial-peer voice 101 pots
description [-[ LONG_DISTANCE calls ]-]
preference 2
destination-pattern 1[2-9]..[2-9]......
port 0/1/0
forward-digits all
no sip-register
!
dial-peer voice 411 pots
description [-[ YellowPages directory ]-]
preference 2
destination-pattern 411
no digit-strip
port 0/1/0
forward-digits 3
no sip-register
!
dial-peer voice 911 pots
description [-[ Emergency 911 calls ]-]
preference 2
destination-pattern 911
port 0/1/0
forward-digits 3
no sip-register
!
dial-peer voice 9911 pots
description [-[ 9911 calls to EMERGENCY ]-]
preference 2
destination-pattern 9911
port 0/1/0
forward-digits 3
no sip-register
!
dial-peer voice 6665 voip
description [-[ Incoming dial-peer Incoming calls from SIP ]-]
preference 1
destination-pattern 17052696665
voice-class codec 100
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
no vad
!
dial-peer voice 6666 voip
description [-[ 7 Digit Local calls to SIP ]-]
translation-profile outgoing local
preference 1
destination-pattern [2-9]......
voice-class codec 100
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
no vad
!
dial-peer voice 6667 voip
description [-[ 11 Digit Long Distance calls to SIP ]-]
translation-profile outgoing SIPout
preference 1
destination-pattern 1[2-9]..[2-9]......
voice-class codec 100
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
no vad
!
dial-peer voice 100 pots
description [-[ LOCAL_DIAL plan FXO port 0/1/0 connected to PSTN wall jack]-
preference 2
destination-pattern [2-9]......
port 0/1/0
no sip-register
!

********THIS IS THE DIAL-PEER i ADDED BUT STILL NOT WORKING*****
dial-peer voice 102 pots
description [-[ Local_Dial plan FXO port 0/1/0 connected to PSTN wall jack ]
preference 2
destination-pattern [2-9]..[2-9]......
port 0/1/0
forward-digits all
no sip-register

Any chance someone can help me with this issue.

Thanks

ASKER CERTIFIED SOLUTION

ibanez7

membership

This solution is only available to members.

To access this solution, you must be a member of Experts Exchange.

Start Free Trial

ibanez7

ASKER

As stated in my last response to my own question.

Here's the fix.

dial-peer voice 100 pots
description [-[ LOCAL_DIAL plan FXO port 0/1/0 connected to PSTN wall jack]-
preference 2
destination-pattern [2-9]......
port 0/1/0
prefix 705 ********Added this line. All is now working correctly
no sip-register

Reason why this works:
On a POTS dial peer, only wildcard-matched digits are forwarded by default.
Use the prefix command to send the prefix numbers 705 before forwarding the four wildcard-matched digits.

********THIS DIAL-PEER IS NOT REQUIRED SO BEEN REMOVED*****
dial-peer voice 102 pots
description [-[ Local_Dial plan FXO port 0/1/0 connected to PSTN wall jack ]
preference 2
destination-pattern [2-9]..[2-9]......
port 0/1/0
forward-digits all
no sip-register

Have figured it out by myself so no points will be awarded.

Thanks all the same