Again you are not saying how asterisk is connected to the panasonic, stop using pictures.
On 8/1/06, Pablo Mora <[EMAIL PROTECTED]> wrote:
Ok Ok, the figure doesn't help. Here we go again… ----- ---------- ----------- ------ | SIP | ----- | ASTERISK | ------ | PANASONIC | ------- | PSTN | ----- ---------- ----------- ------ | | Ext1 Ext2 Here is my dialplan [incoming] exten => s,1,Answer exten => s,2,Background(prueba-pbx) exten => s,3,Set(TIMEOUT(response)=5) exten => 1001,1,Dial,SIP/1001|20 exten => 1001,2,Hangup exten => 1001,102,Congestion,3 exten => 1002,1,Dial,SIP/1002|20 exten => 1002,2,Hangup exten => 1002,102,Congestion,3 [sip] include => outgoing exten => 1001,1,Dial(SIP/1001,20) exten => 1001,2,Hangup exten => 1001,102,Congestion,3 exten => 1002,1,Dial(SIP/1002,20) exten => 1002,2,Hangup exten => 1002,102,Congestion,3 [outgoing] exten => 0,1,Dial,Zap/g1 exten => 0,2,Congestion exten => 0,102,Congestion exten => 9,1,Dial,Zap/g1/9 exten => 9,2,Congestion exten => 9,102,Congestion When I make a call from PSTN to SIP, first Answer the Panasonic, after this I digit an Extension and the call goes to asterisk, then I dial to sip and the call goes on successfully. When I make a call from Ext1 or Ext2 to SIP, I dial an extension and the call goes to asterisk, then I dial to sip and the call goes on. When I make a call from SIP to PSTN, first dial 9 and asterisk gets a Zap sending 9 to get PSTN line, the dial the PSTN number and the call goes on. When I make a call from SIP to Ext1 (Ext2… ExtN), the Sip phone keeps ringing and user behind Ext1 doesn't hear anything. Your help will be appreciated. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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