Hi, I am configuring an audiocodes Medant1000 to talk to my asterisk box. So far I have successfull in landing a single call from mediant to my *box. my sip conf is as follows:
[general] context=sip bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [3911700] type=friend host=dynamic dtmfmode=info secret=blah context=sip where 3911700 is my E1 telephone no. in my extensions.conf I have exten => 3911700,1,Dial(SIP/100) When I dial from outside to my E1 number calls are coming like the following: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.9.230;branch=z9hG4bKac806223297 Max-Forwards: 70 From: <sip:[EMAIL PROTECTED]>;tag=1c806218385 To: <sip:[EMAIL PROTECTED];user=phone> Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: <sip:[EMAIL PROTECTED]> Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Remote-Party-ID: <sip:[EMAIL PROTECTED]>;party=called;npi=1;ton=4 Remote-Party-ID: <sip:[EMAIL PROTECTED]>;party=calling;privacy=off;screen=yes;screen-ind=1;npi=1;ton=0 User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 348 and the call get's connected to SIP/100 via the line in extensions.conf But what I am expecting is that the calls to come to the context's 's' extension. I am not sure if the changes are to be done in Asterisk or to Mediant. Any help in this will be much appreciated. raj _______________________________________________ --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
