Reddog4891@aol.com
2003-Nov-26 23:09 UTC
[Asterisk-Users] Attempting to get SJPhone configured for Asterisk- Help!
I recently setup an Asterisk Server- I was able to follow a tutorial from http://www.automated.it/guidetoasterisk.htm#_Toc49248752 Until it told me to call another line, let it ring until voice mail picks up. My problem is the tutorial left out how to configure a SJPhone so that it connects to my asterisk server not directly FWD. I've tried everything I can think of, I must be missing something simple. In the tutorial I got up to the part where I successfully added Voicemail boxes and started configuring the server to go to voice mail after "X" number of rings or if the line was busy. The next line in the tutorial instructed me to test it out, but the SJPhone software is not configured properly. All I?m trying to do is setup a simple voicemail server for 1-5 phones. If I can get this to work, I?m buying a few Budgetone 102?s. If anyone can point me in the right direction, I would greatly appreciate it! Thanks- Dan
Mark Johnston
2003-Nov-27 08:34 UTC
[Asterisk-Users] Attempting to get SJPhone configured for Asterisk- Help!
Reddog4891@aol.com wrote:> My problem is the tutorial left out how to configure a SJPhone so > that it connects to my asterisk server not directly FWD. I've tried > everything I can think of, I must be missing something simple.I'm using SJPhone with the following config: sip.conf: [markspc] type=friend host=dynamic dtmfmode=inband callerid="Mark's PC" <1> username=markspc SJPhone (SIP tab): Use local outbound proxy - checked. Proxy IP Address: 192.168.0.1 Caller ID: sip:markspc@192.168.0.1 Register - checked. Account: markspc Password blank. Everything else is default (haven't changed the Advanced SIP Options or anything.) Of course, in production you'd want to add a secret line to sip.conf and the corresponding password to SJPhone. When it's working, SJPhone shows: Status: no active calls Default protocol: SIP SIP Proxy: registered with 192.168.0.1 Host address: 192.168.0.2 and Asterisk's console says: Registered SIP 'markspc' at 192.168.0.2 Asterisk also periodically reports: Got SIP response 481 "Subscription does not exist" back from 192.168.0.2 which seems harmless. HTH, Mark