Asterisk recognizes and interprets the XML correctly?
> -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Martin Pycko > Sent: Tuesday, September 02, 2003 12:26 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Packet8 DTA310 > > 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
