Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
New Member

Toll Bypass using SIP trunks at main site.

I have a main site with 10 SIP trunks.  I am trying to have long distance calls from the remote site to go across the VPN and call out the SIP trunks at the main site.  I started by just running a simple test to call a 7 digit phone number from the remote site through the main site.  Here is my configuration with example phone number:

Remote site:  (192.168.11.1)

dial-peer voice 2101 voip

destination-pattern 95551234

preference 1

session target ipv4:192.168.10.1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

Local Site: (192.168.10.1)

dial-peer voice 2101 voip

incoming called-number 95551234

voice-class h323 1

dtmf-relay h245-alphanumeric

fax protocol cisco

no vad

dial-peer voice 1030 voip

description **CCA*North American-7-Digit*7-Digit Local**

translation-profile outgoing PSTN_Outgoing

preference 1

destination-pattern 9[2-9]......

session protocol sipv2

session target sip-server

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

voice translation-profile PSTN_Outgoing

translate calling 1111

translate called 1112

translate redirect-target 410

translate redirect-called 410

voice translation-rule 410

rule 1 /^9\(.......)\)$/ /970\1/

rule 2 /^9\(.*\)/ /\1/

rule 15 /^...$/ /9702430727/

voice translation-rule 1111

rule 15 /^...$/ /9702430727/

!

voice translation-rule 1112

rule 1 /^9/ //

When I call from the remote site, the called phone rings, but there is no one there.  The remote caller hears a fast busy.  Any help would be appreciated!

Thank You.

Everyone's tags (5)
1 REPLY
Hall of Fame Super Silver

Toll Bypass using SIP trunks at main site.

Post "debug ccsip messages" and provide called and calling number.

Is there a reason you have H323 between the sites and SIP to SIP trunk, why not configure SIP between sites as well?

Can do you hard code everything to one codec? As your outbour leg from remote site  has no codec class or codec specified which means it will default to G729, yet your SIP trunk dial peer has codec class (which you did not post).

You mention VPN connection, Is there any NATing taking place? Are all needed ports permitted?

HTH,

Chris

399
Views
0
Helpful
1
Replies
CreatePlease to create content