[asterisk-users] Fast busy after one digit dialled? - 7970 SIP 8.0.3

[Paul Duffy]

Hi All

I'm trying to get a 7970 working with SIP 8.0.3S and the latest build of
Asterisk (doing this as a new build before a replacement of my existing
system).

So far I've managed to get the phone upgraded successfully.

I can dial the phone from the console successfully.

However whenever I dial a number from the phone the phone goes into fast
busy mode after the first digit.

The SIP debug shows (in this instance the digit dialled was a 7):-

<-- SIP read from 10.131.111.51:49226:
INVITE sip:7 at 10.131.111.10;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.111.51:5060;branch=z9hG4bK8f6323d8
From: "301" <sip:301 at 10.131.111.10>;tag=000f3487e566000ed76e9557-dcf4e93a
To: <sip:7 at 10.131.111.10;user=phone>
Call-ID: 000f3487-e5660009-e2e36f89-af6ced7c at 10.131.111.51
Max-Forwards: 70
Date: Thu, 20 Jul 2006  GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7970G/8.0
Contact: <sip:301 at 10.131.111.51:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "301"
<sip:301 at 10.131.111.10>;party=calling;id-type=subscriber;privacy=off;screen=
yes
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 275
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 21184 0 IN IP4 10.131.111.51
s=SIP Call
t=0 0
m=audio 19796 RTP/AVP 0 8 18 101
c=IN IP4 10.131.111.51
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (19 headers 13 lines)---
Using INVITE request as basis request -
000f3487-e5660009-e2e36f89-af6ced7c at 10.131.111.51
Sending to 10.131.111.51 : 5060 (non-NAT)
Found user '301'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.131.111.51:19796
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing),  combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1  (telephone-event)
Looking for 7 in 7970SIP (domain 10.131.111.10)
Reliably Transmitting (no NAT) to 10.131.111.51:5060:
SIP/2.0 484 Address Incomplete
********************************************
Via: SIP/2.0/UDP
10.131.111.51:5060;branch=z9hG4bK8f6323d8;received=10.131.111.51
From: "301" <sip:301 at 10.131.111.10>;tag=000f3487e566000ed76e9557-dcf4e93a
To: <sip:7 at 10.131.111.10;user=phone>;tag=as17095de3
Call-ID: 000f3487-e5660009-e2e36f89-af6ced7c at 10.131.111.51
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7 at 10.131.111.10>
Content-Length: 0


---

<-- SIP read from 10.131.111.51:49227:
ACK sip:7 at 10.131.111.10;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.111.51:5060;branch=z9hG4bK8f6323d8
From: "301" <sip:301 at 10.131.111.10>;tag=000f3487e566000ed76e9557-dcf4e93a
To: <sip:7 at 10.131.111.10;user=phone>;tag=as17095de3
Call-ID: 000f3487-e5660009-e2e36f89-af6ced7c at 10.131.111.51
Date: Thu, 20 Jul 2006  GMT
CSeq: 101 ACK
Content-Length: 0


--- (8 headers 0 lines)---
Destroying call '000f3487-e5660009-e2e36f89-af6ced7c at 10.131.111.51'


The "Address Incomplete" that I've highlighted use ************ above is
interesting but having read the wiki and googled extensively I can't see a
reason for this problem.

As I'm now starting to get brain block from this can anyone make any
suggestions (it's probably something incredibly simple I'm missing, I just
can't see it).

Thanks