1. Brekeke Product Name and version: Brekeke SIP Server 2.0
2. Java version:
3. OS type and the version: Windows 2003
4. UA (phone), gateway or other hardware/software involved: SJ Phone, Cisco Callmanager, Cisco VOice GW
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
6. Your problem:
I am testing a configuration where I can dial ISN style numbers by creating a dial rule and pointing the destination to the freenum.org/tello.com resolver. The issue I am having is when I start a call, the rule is matched but the deployed rule does not act as expected:
From any SIP softphone not registerd to any SIP server:
I can dial 612@fwd.pulver.com and call is OK. (FWD test number)
I can dial 612*262@freenum.org and call is OK. (FWD test number)
When I try to dial jesse@lightspeedtech.net, I have a NAPTR record pointing that SIP request to my SIP server and a rule matching that username. The rule is matched but in the deploy pattern, I have "TO:sip:612*262@freenum.org". I see the 302 reply from the sip server but in my logs for the SJ Phone software, it is redirected to "612*262@74.212.22.251" which is the address of my sip server and receive a "Not Found" error. I have tried other options to set the host with no success.
For additional testing, I created a new rule that all INVITE or just calls to jesse@lightspeedtech.net should point to 612@fwd.pulver.com and they work OK.
2. Java version:
3. OS type and the version: Windows 2003
4. UA (phone), gateway or other hardware/software involved: SJ Phone, Cisco Callmanager, Cisco VOice GW
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
6. Your problem:
I am testing a configuration where I can dial ISN style numbers by creating a dial rule and pointing the destination to the freenum.org/tello.com resolver. The issue I am having is when I start a call, the rule is matched but the deployed rule does not act as expected:
From any SIP softphone not registerd to any SIP server:
I can dial 612@fwd.pulver.com and call is OK. (FWD test number)
I can dial 612*262@freenum.org and call is OK. (FWD test number)
When I try to dial jesse@lightspeedtech.net, I have a NAPTR record pointing that SIP request to my SIP server and a rule matching that username. The rule is matched but in the deploy pattern, I have "TO:sip:612*262@freenum.org". I see the 302 reply from the sip server but in my logs for the SJ Phone software, it is redirected to "612*262@74.212.22.251" which is the address of my sip server and receive a "Not Found" error. I have tried other options to set the host with no success.
For additional testing, I created a new rule that all INVITE or just calls to jesse@lightspeedtech.net should point to 612@fwd.pulver.com and they work OK.