[Asterisk-Users] two phones one host

[Chris Lee]

I have a sip box with two FXS ports (Draytek 2600v adsl router)
I have had very little luck getting the two talking together.
For a very short time I did have calls originating on my FXO card routed 
to the phone working.

Phone1/2 on router 	---> handytone  works
handytone 		---> router phone1/2 works
Phone1/2 on router 	---> asterisk  broken*
asterisk 		---> router phone1/2 broken**

* Can see asterisk receive the number and start following calling plan 
but no sound comes through, phone eventualy times out.
** Worked for a short time for no apparent reason, now sees phone as 
bussy, phone does not ring.

Router:
	hostname: gateway-2.cybericom.co.uk
	phone 1: p3000
	phone 2: p3001
	IP: 10.10.10.2

Asterisk:
	hostname: babybell.cybericom.co.uk
	IP 10.10.10.3

-If I use a phone connected to the router, I see asterisk receive the 
number and start following dial plan, but I dont hear anything and 
asterisk retries sending the following packet:

Retransmitting #3 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ors-20101
From: p3000 <sip:p3000 at 10.10.10.2:5060>;tag=fSd-1369
To: <sip:8 at 10.10.10.3>;tag=as07550202
Call-ID: MGH-26709 at 10.10.10.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8 at 10.10.10.3>
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 16230 16230 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 13984 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

  to 10.10.10.2:5060

-If I get a call via my fxo and it is supposed to be routed to a phone 
on the router I get a 404 not found but it seems asterisk is not asking 
for a specific phone by name, here is the first packet:

     -- Executing Dial("Zap/1-1", "SIP/p3000|10|tr") in new stack
We're at 10.10.10.3 port 13934
Answering with preferred capability 2
Answering with preferred capability 4
Answering with preferred capability 8
Answering with non-codec capability 1
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:gateway-2.cybericom.co.uk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK04efdbfe
From: "Cybericom" <sip:Call at 10.10.10.3>;tag=as179b70c5
To: <sip:gateway-2.cybericom.co.uk>
Contact: <sip:Call at 10.10.10.3>
Call-ID: 4e8639721c227f13006076e208db0854 at 10.10.10.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 10 Feb 2004 15:16:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 16258 16258 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 13934 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
  (no NAT) to 10.10.10.2:5060
     -- Called p3000

is there not supposed to be a p3000@ in the sip: line?

here is my sip.conf if something is wrong please let me know, thanks:
[general]
port=5060                       ; Port to bind to
bindaddr=0.0.0.0                ; Address to bind to
context=in-sip          ; Default for incoming calls
callerid=Call
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
maxexpirey=1800
defaultexpirey=600
tos=throughput
 

[p3000]
type=friend
host=dynamic
user=p3000
;secret=
dtmfmode=rfc2833
mailbox=3000
callerid="Reception" <3000>
qualify=yes
context=wellingborough-road
 

[p3001]
type=friend
host=dynamic
user=p3001
;secret=
dtmfmode=rfc2833
mailbox=3001
callerid="Reception" <3001>
qualify=yes
context=wellingborough-road

Router has box for registrar set to 10.10.10.3
place for naming phone set to p3000 and p3001
place for port set to 5060



Thank you for any help

Regards
Chris Lee