Or at least a username and password. How would * be able to differentiate between the two clients? Try ading: Username= Secrect=
For both... -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Primoz Kragelj Sent: Monday, May 02, 2005 2:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] SIP problems Hi all, I'm newbie to VOIP/SIP/asterisk... and I having problems with SIP on local network. I have Freebsd server 5.3 running asterisk and two x-lite clients. I added following lines to sip.conf [tina] type=friend host=dynamic dtmfmode=inband context=sip [primozz] type=friend host=dynamic dtmfmode=inband context=sip And following to extensions.conf [sip] exten => 1000,1,Dial,SIP/tina exten => 2000,1,Dial,SIP/primozz *CLI> sip show users Username Secret Accountcode Def.Context ACL NAT primozz sip No RFC35 tina sip No RFC35 I have X-Lite clinet on Win XP and while trying to make call to "tina" I got 404 error - not found. Same for vice versa...Both users are local. >From debug below following line: To: <sip:[EMAIL PROTECTED]>;tag=as1283188b is very strange to me. Instead od 192.168.1.3 there should be 192.168.1.1. Do I need to put some ware static IP for each client ? And following is debug from asterisk: Peer audio RTP is at port 192.168.1.3:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for tina in sip list_route: hop: <sip:[EMAIL PROTECTED]:5060> Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA From: Primoz <sip:[EMAIL PROTECTED]>;tag=1716760483 To: <sip:[EMAIL PROTECTED]>;tag=as1283188b Call-ID: [EMAIL PROTECTED] CSeq: 22324 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.1.3:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA From: Primoz <sip:[EMAIL PROTECTED]>;tag=1716760483 To: <sip:[EMAIL PROTECTED]>;tag=as1283188b Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 22324 ACK Max-Forwards: 70 Content-Length: 0 Thanks for help ! Regards, Primoz _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
