Hi Scott, I solved the problem. Early, I configured the gateway using full hostname.domain.name to the other gateway, even changed the address with IP address, but the result still the same. Then after I changed the gateway address using SRV, suddenly it work.
Well, I think that's why sipXecs very dependent with DNS SRV. Thanks for the help Scott. Next step, testing between 2 server with both server behind NAT. Wish me luck ;-) Regards, B. Aidil > -----Original Message----- > From: Boy Aidil Sjam [mailto:[email protected]] > Sent: Friday, June 12, 2009 12:13 PM > To: 'Scott Lawrence' > Cc: '[email protected]' > Subject: RE: [sipx-users] Interconnection 2 sipXecs > > > > > -----Original Message----- > > From: Scott Lawrence [mailto:[email protected]] > > Sent: Friday, June 12, 2009 1:43 AM > > To: Boy Aidil Sjam > > Cc: [email protected] > > Subject: RE: [sipx-users] Interconnection 2 sipXecs > > > > On Thu, 2009-06-11 at 11:49 +0700, Boy Aidil Sjam wrote: > > > > > > > -----Original Message----- > > > > From: Scott Lawrence [mailto:[email protected]] > > > > Sent: Wednesday, June 10, 2009 7:20 PM > > > > To: Boy Aidil Sjam > > > > Cc: [email protected] > > > > Subject: RE: [sipx-users] Interconnection 2 sipXecs > > > > > > > > On Wed, 2009-06-10 at 17:30 +0700, Boy Aidil Sjam wrote: > > > > > Scott, > > > > > I've setup lab environment before go live to public network > > > > (internet), and I like to see how it works. > > > > > There's no NAT involve in this lab testing. > > > > > The scenarios: > > > > > 1. I installed 2 sipXecs server, each server in their own > domain. > > > > > 2. Each server use the same dial-plan (I set up ext. 200 - 201) > > > > > 3. I set up prefix 9 to dial to another domain. > > > > > 4. User from server A able to dial server B autoattendant using > > 9100 > > > > and able to transfer to another extension and vice versa. > > > > > 5. But when one of the user want to dial directly to user > > extension > > > > in another server (dialed 9200) the call directly terminated. > > > > > > > > > > Is that the way how the sipXecs works or is there something > > > > > wrong > > > > with my configuration? > > > > > > > > That should work. > > > > > > > > > > > > To debug this, you'll need to trace the message flow and see > > > > what's going wrong. See: > > > > > > > > http://sipx- > > > > > > > wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer#Ge > > t > > > > ting_SIP_Messages_to_display > > > > > > > > when you get the trace data, take a look at it using sipviewer > > > > and/or post the trace with a description of your configuration > > > > (identify components by IP address), what you were doing, and > > > > which call in the trace you're talking about (by call-id or frame > > > > number in the trace, preferably). > > > > > > > > > > Server A > > > Domain: office.net > > > Hostname: voip.office.net > > > IP address: 10.10.12.50 > > > User Extension: 201 > > > User IP address: 10.10.12.169 > > > > > > Custom dial plan > > > Dialed Number --> Prefix 9 and 3 digits Required Permission --> > > > Local Dialing Resulting Call --> Dial (empty) and append Matched > > > suffix Schedule Always > > > Gateway: voip.net.lab > > > > > > Server B > > > Domain: net.lab > > > Hostname: voip.net.lab > > > IP address: 10.10.12.200 > > > User extension: 201 > > > User IP address: 10.10.12.170 > > > > > > Custom dial plan > > > Dialed Number --> Prefix 9 and 3 digits Required Permission --> > > > Local Dialing Resulting Call --> Dial (empty) and append Matched > > > suffix Schedule Always > > > Gateway: voip.office.net > > > > > > 1st test, extension [email protected] dialed 9100 ([email protected]) > > > Result: AA answered with default greeting Then [email protected] press > > > 201 > > > Result: Call transfer to [email protected], and the phone is ringing > > > > > > 2nd test, extension [email protected] dialed 9201 ([email protected]) > > > Result: Busy tone and followed by AA said, "The person you call > > > isn't available" > > > > > > I attached the 2nd test sipx-trace result from both server > > > > It looks like you don't have the clocks synchronized on these two > > servers; this makes the traces very difficult to combine and analyze. > > > > >From what I can see, there was no phone registered at [email protected] > > > Here another sipx-trace files. > I've synchronized the clocks to the same NTP server I used different > extension: > [email protected] = 10.10.12.30 > [email protected] = 10.10.12.31 > > Both registered to their own sipx server domain: > voip.net.lab = 10.10.12.200 > voip.office.net = 10.10.12.50 > > this time, called initiated from [email protected] to [email protected] using > 9205 and the result still the same like before. > > Before and after the test, I've check the phone registration page and > both phone still registered > > > > > DISCLAIMER: Information in this message is confidential and may be legally privileged. 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