[Asterisk-Users] Packet8 DTA310

[Martin Pycko]

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.
> > >
>
>
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