1. Brekeke Product Name and version:2.3.6.0
2. Java version: 4.1.31
3. OS type and the version: Fedora 8
4. UA (phone), gateway or other hardware/software involved: Linksys PAP2
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : Pattern 4
6. Your problem: Make call from a pstn--------->CiscoGW--------->SIP server-------->my sip phone. Cisco GW, been configured to use DID, and I can trace my call going to the right dial-peer, but always get a busy signal. Below, are my dial-peers and sip dial plan, any idea why this not working? Any help greatly appreciated.
Let's assume the following:
974444444 is my DID,
215.8.160.20 is my Cisco GW
215.8.163.17 is my sip server IP
dial-peer voice 1 pots
incoming called-number 974444444
direct-inward-dial
dial-peer voice 3 voip
destination-pattern 974444444
session target ipv4:215.8.160.17
Matching Pattern
$request=^INVITE
$addr=215\.8\.160\.20
To=sip:97444444@
Deploy Pattern
$auth=off
To=sip:974444444@
2. Java version: 4.1.31
3. OS type and the version: Fedora 8
4. UA (phone), gateway or other hardware/software involved: Linksys PAP2
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : Pattern 4
6. Your problem: Make call from a pstn--------->CiscoGW--------->SIP server-------->my sip phone. Cisco GW, been configured to use DID, and I can trace my call going to the right dial-peer, but always get a busy signal. Below, are my dial-peers and sip dial plan, any idea why this not working? Any help greatly appreciated.
Let's assume the following:
974444444 is my DID,
215.8.160.20 is my Cisco GW
215.8.163.17 is my sip server IP
dial-peer voice 1 pots
incoming called-number 974444444
direct-inward-dial
dial-peer voice 3 voip
destination-pattern 974444444
session target ipv4:215.8.160.17
Matching Pattern
$request=^INVITE
$addr=215\.8\.160\.20
To=sip:97444444@
Deploy Pattern
$auth=off
To=sip:974444444@