Well this debug desn't show the bad call setup. And furthermore all commands are accepted by the asterisk/UA.
Martin On Mon, 1 Sep 2003, Andrew Joakimsen wrote: > There might be some other stuff mixed in there as well, 64.36.104.205 is > asterisk and 64.36.104.206 is the DTA > > 11 headers, 2 lines > Reliably Transmitting: > NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK471af161 > From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as17328ab1 > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > User-Agent: Asterisk PBX > Event: message-summary > Content-Type: application/simple-message-summary > Content-Length: 36 > > Messages-Waiting: no > Voicemail: 0/1 > (no NAT) to 64.36.104.203:5060 > Sip read: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK471af161 > From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as17328ab1 > To: <sip:[EMAIL PROTECTED]>;tag=6b0caf74-4936-bceb-18dc-3ee5469aa3f6 > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > User-Agent: Grandstream SIP UA 1.0.3.81 > Contact: <sip:[EMAIL PROTECTED];user=phone> > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE > Content-Length: 0 > > > 10 headers, 0 lines > Sip read: > SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 64.36.104.206;branch=z9hG4bK.2ddac.11a85c89;rport > From: <sip:[EMAIL PROTECTED]>;tag=t2d9e0a11a85c88g > To: sip:[EMAIL PROTECTED] > Call-ID: [EMAIL PROTECTED] > CSeq: 100 SUBSCRIBE > Contact: sip:[EMAIL PROTECTED] > Expires: 3600 > Max-Forwards: 70 > Event: traverse > User-Agent: DTA SIP/0.11.8 NNOS/VR30 > Content-Type: application/sdp > Content-Length: 156 > > v=0 > o=0403532579 0 0 IN IP4 64.36.104.206 > =-m3*CLI> > c=IN IP4 64.36.104.206 > t=0 0 > m=audio 8002 RTP/AVP 18 101 > a=ptime:10 > a=rtpmap:101 telephone-event/8000 > > 13 headers, 8 lines > Using latest SUBSCRIBE request as basis request > Sending to 64.36.104.206 : 5060 (non-NAT) > Looking for 9999 in international > Transmitting (no NAT): > SIP/2.0 200 OK > Via: SIP/2.0/UDP 64.36.104.206;branch=z9hG4bK.2ddac.11a85c89;rport > From: <sip:[EMAIL PROTECTED]>;tag=t2d9e0a11a85c88g > To: sip:[EMAIL PROTECTED];tag=as57545bcd > Call-ID: [EMAIL PROTECTED] > CSeq: 100 SUBSCRIBE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Expires: 3600 > Contact: <sip:[EMAIL PROTECTED]>;expires=3600 > Content-Length: 0 > > > to 64.36.104.206:5060 > Reliably Transmitting: > NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK5bb88bc0 > From: sip:[EMAIL PROTECTED];tag=as57545bcd > To: <sip:[EMAIL PROTECTED]>;tag=t2d9e0a11a85c88g > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > User-Agent: Asterisk PBX > Content-Type: application/xpidf+xml > Content-Length: 352 > > <?xml version="1.0"?> > <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" > "xpidf.dtd"> > <presence> > <presentity uri="sip:[EMAIL PROTECTED];method=SUBSCRIBE" /> > <atom id="9999"> > <address uri="sip:[EMAIL PROTECTED];user=ip" priority="0,800000"> > <status status="open" /> > <msnsubstatus substatus="online" /> > </address> > </atom> > </presence> > (no NAT) to 64.36.104.206:5060 > Sip read: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK5bb88bc0 > From: <sip:[EMAIL PROTECTED]>;tag=as57545bcd > To: <sip:[EMAIL PROTECTED]>;tag=t2d9e0a11a85c88g > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > Server: DTA SIP/0.11.8 NNOS/VR30 > Content-Length: 0 > > > 8 headers, 0 lines > Message is NOTIFY > hm3*CLI> > > > -----Original Message----- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Martin Pycko > > Sent: Saturday, August 30, 2003 12:30 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Packet8 DTA310 > > > > Post the sip debug .. maybe someone will help you. > > > > Martin > > > > On Sat, 30 Aug 2003, Andrew Joakimsen wrote: > > > > > Has anyone been successful in using the DTA310 as provided by > Packet8 to > > > work with asterisk? I have gotten it to register with Asterisk but > > > whenever I try to dial a call all I get is silence, when I dial an > > > invalid extension I get a fast busy signal. When looking at the SIP > > > debug it seems that it is transmitting XML. > > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
