[Asterisk-Users] Packet8 DTA310

[Andrew Joakimsen]

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.
> >