[Asterisk-Users] Packet8 DTA310

[Andrew Joakimsen]

Asterisk recognizes and interprets the XML correctly?


> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of Martin Pycko
> Sent: Tuesday, September 02, 2003 12:26 PM
> To: asterisk-users at lists.digium.com
> 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:318 at 64.36.104.203 SIP/2.0
> > Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK471af161
> > From: "asterisk" <sip:318 at 64.36.104.205>;tag=as17328ab1
> > To: <sip:318 at 64.36.104.203>
> > Contact: <sip:318 at 64.36.104.205>
> > Call-ID: 524b4aeb2b1988b25d1a458c74c543f5 at 64.36.104.205
> > 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:318 at 64.36.104.205>;tag=as17328ab1
> > To: <sip:318 at 64.36.104.203>;tag=6b0caf74-4936-bceb-18dc-3ee5469aa3f6
> > Call-ID: 524b4aeb2b1988b25d1a458c74c543f5 at 64.36.104.205
> > CSeq: 102 NOTIFY
> > User-Agent: Grandstream SIP UA 1.0.3.81
> > Contact: <sip:318 at 64.36.104.203;user=phone>
> > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO,
SUBSCRIBE
> > Content-Length: 0
> >
> >
> > 10 headers, 0 lines
> > Sip read:
> > SUBSCRIBE sip:9999 at hm5.joako.com SIP/2.0
> > Via: SIP/2.0/UDP 64.36.104.206;branch=z9hG4bK.2ddac.11a85c89;rport
> > From: <sip:0403532579 at 64.36.104.206>;tag=t2d9e0a11a85c88g
> > To: sip:9999 at hm5.joako.com
> > Call-ID: 50ff4-11a85c88-1410aaf-40247c92 at hm5.joako.com
> > CSeq: 100 SUBSCRIBE
> > Contact: sip:0403532579 at 64.36.104.206
> > 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:0403532579 at 64.36.104.206>;tag=t2d9e0a11a85c88g
> > To: sip:9999 at hm5.joako.com;tag=as57545bcd
> > Call-ID: 50ff4-11a85c88-1410aaf-40247c92 at hm5.joako.com
> > CSeq: 100 SUBSCRIBE
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > Expires: 3600
> > Contact: <sip:9999 at 64.36.104.205>;expires=3600
> > Content-Length: 0
> >
> >
> >  to 64.36.104.206:5060
> > Reliably Transmitting:
> > NOTIFY sip:0403532579 at 64.36.104.206 SIP/2.0
> > Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK5bb88bc0
> > From: sip:9999 at hm5.joako.com;tag=as57545bcd
> > To: <sip:0403532579 at 64.36.104.206>;tag=t2d9e0a11a85c88g
> > Contact: <sip:9999 at 64.36.104.205>
> > Call-ID: 50ff4-11a85c88-1410aaf-40247c92 at hm5.joako.com
> > 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:0403532579 at 64.36.104.206;method=SUBSCRIBE" />
> > <atom id="9999">
> > <address uri="sip:9999 at hm5.joako.com;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:9999 at hm5.joako.com>;tag=as57545bcd
> > To: <sip:0403532579 at 64.36.104.206>;tag=t2d9e0a11a85c88g
> > Call-ID: 50ff4-11a85c88-1410aaf-40247c92 at hm5.joako.com
> > 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: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-
> > > admin at lists.digium.com] On Behalf Of Martin Pycko
> > > Sent: Saturday, August 30, 2003 12:30 PM
> > > To: asterisk-users at lists.digium.com
> > > 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
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users