FIXED: Cisco 7940 drops calls - retrans_pkt no reply

Hi.

I can’t reliably make calls from my 7940 with SIP 8.12(0) software image. If I make a call, nine times in ten Asterisk drops it after around 20 seconds. I’ve read sip-retransmit.txt and I am none the wiser.

The phone (172.16.3.245) is in the same subnet as the Asterisk box (172.16.3.2), there is no NAT or firewall between the two.

I’m completely out of ideas, any suggestions would be welcomed.

Here’s a debug:

pabx*CLI> 
<--- SIP read from 172.16.3.245:50462 --->
INVITE sip:917070@pabx.spruce SIP/2.0
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK205c7f3a
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
Max-Forwards: 70
Date: Tue, 22 Sep 2009 21:01:13 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:203@172.16.3.245:5061;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "Karen" <sip:203@pabx.spruce>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 276
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 16187 0 IN IP4 172.16.3.245
s=SIP Call
t=0 0
m=audio 27906 RTP/AVP 0 8 18 101
c=IN IP4 172.16.3.245
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (18 headers 13 lines) ---
Sending to 172.16.3.245 : 5061 (no NAT)
Using INVITE request as basis request - 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245

<--- Reliably Transmitting (no NAT) to 172.16.3.245:5061 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK205c7f3a;received=172.16.3.245
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as19cb83a5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0afa8f28"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245' in 32000 ms (Method: INVITE)
Found user '203'
pabx*CLI> 
<--- SIP read from 172.16.3.245:50492 --->
ACK sip:917070@pabx.spruce SIP/2.0
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK205c7f3a
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as19cb83a5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
Date: Tue, 22 Sep 2009 21:01:13 GMT
CSeq: 101 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
pabx*CLI> 
<--- SIP read from 172.16.3.245:50462 --->
INVITE sip:917070@pabx.spruce SIP/2.0
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK6db4a65a
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
Max-Forwards: 70
Date: Tue, 22 Sep 2009 21:01:14 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:203@172.16.3.245:5061;transport=udp>
Proxy-Authorization: Digest username="203",realm="asterisk",uri="sip:917070@pabx.spruce",response="f386fe3f3ff1e519e646e56f0b2da5d2",nonce="0afa8f28",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "Karen" <sip:203@pabx.spruce>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 276
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 16187 0 IN IP4 172.16.3.245
s=SIP Call
t=0 0
m=audio 27906 RTP/AVP 0 8 18 101
c=IN IP4 172.16.3.245
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (19 headers 13 lines) ---
Sending to 172.16.3.245 : 5061 (no NAT)
Using INVITE request as basis request - 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
Found user '203'
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 172.16.3.245:27906
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Got unsupported a:fmtp in SDP offer 
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer 
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.3.245:27906
Looking for 917070 in my-phones (domain pabx.spruce)
list_route: hop: <sip:203@172.16.3.245:5061;transport=udp>
pabx*CLI> 
<--- Transmitting (no NAT) to 172.16.3.245:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK6db4a65a;received=172.16.3.245
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:917070@172.16.3.2>
Content-Length: 0


<------------>
    -- Executing [917070@my-phones:1] Dial("SIP/203-103eb538", "ZAP/1/17070|60|r") in new stack
    -- Called 1/17070
pabx*CLI> 
<--- Transmitting (no NAT) to 172.16.3.245:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK6db4a65a;received=172.16.3.245
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as4793fff5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:917070@172.16.3.2>
Content-Length: 0


<------------>
    -- Zap/1-1 answered SIP/203-103eb538
Audio is at 172.16.3.2 port 16356
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.16.3.245:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK6db4a65a;received=172.16.3.245
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as4793fff5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:917070@172.16.3.2>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 17089 17089 IN IP4 172.16.3.2
s=session
c=IN IP4 172.16.3.2
t=0 0
m=audio 16356 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (no NAT) to 172.16.3.245:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK6db4a65a;received=172.16.3.245
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as4793fff5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:917070@172.16.3.2>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 17089 17089 IN IP4 172.16.3.2
s=session
c=IN IP4 172.16.3.2
t=0 0
m=audio 16356 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #2 (no NAT) to 172.16.3.245:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK6db4a65a;received=172.16.3.245
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as4793fff5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:917070@172.16.3.2>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 17089 17089 IN IP4 172.16.3.2
s=session
c=IN IP4 172.16.3.2
t=0 0
m=audio 16356 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #3 (no NAT) to 172.16.3.245:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK6db4a65a;received=172.16.3.245
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as4793fff5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:917070@172.16.3.2>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 17089 17089 IN IP4 172.16.3.2
s=session
c=IN IP4 172.16.3.2
t=0 0
m=audio 16356 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Reliably Transmitting (no NAT) to 172.16.3.103:5060:
OPTIONS sip:205@172.16.3.103:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.3.2:5060;branch=z9hG4bK17363aa4;rport
From: "asterisk" <sip:asterisk@172.16.3.2>;tag=as7d4b1cca
To: <sip:205@172.16.3.103:5060;user=phone;transport=udp>
Contact: <sip:asterisk@172.16.3.2>
Call-ID: 0de0a0362467a32a1d0689796f5763cc@172.16.3.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 Sep 2009 21:10:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
pabx*CLI> 
<--- SIP read from 172.16.3.103:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.3.2:5060;branch=z9hG4bK17363aa4;rport
From: "asterisk" <sip:asterisk@172.16.3.2>;tag=as7d4b1cca
To: <sip:205@172.16.3.103:5060;user=phone;transport=udp>;tag=741489567
Call-ID: 0de0a0362467a32a1d0689796f5763cc@172.16.3.2
CSeq: 102 OPTIONS
Server: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 279
Content-Type: application/sdp

v=0
o=205 352304 352304 IN IP4 172.16.3.103
s=Cisco 7912 SIP Call
c=IN IP4 172.16.3.103
t=0 0
m=audio 16384 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (11 headers 12 lines) ---
Really destroying SIP dialog '0de0a0362467a32a1d0689796f5763cc@172.16.3.2' Method: OPTIONS
Retransmitting #4 (no NAT) to 172.16.3.245:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK6db4a65a;received=172.16.3.245
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as4793fff5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:917070@172.16.3.2>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 17089 17089 IN IP4 172.16.3.2
s=session
c=IN IP4 172.16.3.2
t=0 0
m=audio 16356 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Reliably Transmitting (no NAT) to 172.16.3.101:5060:
OPTIONS sip:200@172.16.3.101:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.3.2:5060;branch=z9hG4bK6546f979;rport
From: "asterisk" <sip:asterisk@172.16.3.2>;tag=as7d005511
To: <sip:200@172.16.3.101:5060;user=phone;transport=udp>
Contact: <sip:asterisk@172.16.3.2>
Call-ID: 49a9ec1553e474b649d9c5cf7a21227d@172.16.3.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 Sep 2009 21:10:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
pabx*CLI> 
--- (11 headers 12 lines) ---
Really destroying SIP dialog '49a9ec1553e474b649d9c5cf7a21227d@172.16.3.2' Method: OPTIONS
Retransmitting #5 (no NAT) to 172.16.3.245:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK6db4a65a;received=172.16.3.245
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as4793fff5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:917070@172.16.3.2>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 17089 17089 IN IP4 172.16.3.2
s=session
c=IN IP4 172.16.3.2
t=0 0
m=audio 16356 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #6 (no NAT) to 172.16.3.245:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK6db4a65a;received=172.16.3.245
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as4793fff5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:917070@172.16.3.2>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 17089 17089 IN IP4 172.16.3.2
s=session
c=IN IP4 172.16.3.2
t=0 0
m=audio 16356 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- (7 headers 0 lines) ---
Really destroying SIP dialog '1a0252b6080a0ee73d03709f6a632ce1@172.16.3.2' Method: OPTIONS
[Sep 22 22:10:31] NOTICE[17111]: chan_iax2.c:1968 peercnt_remove: ip callno count decremented to 0 for 217.14.138.130
[Sep 22 22:10:31] NOTICE[17108]: chan_iax2.c:1936 peercnt_add: ip callno count incremented to 1 for 217.14.138.130
[Sep 22 22:10:31] NOTICE[17107]: chan_iax2.c:2258 sched_delay_remove: schedule decrement of callno used for 217.14.138.130 in 60 seconds
[Sep 22 22:10:32] WARNING[17117]: chan_sip.c:1981 retrans_pkt: Maximum retries exceeded on transmission 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Sep 22 22:10:32] WARNING[17117]: chan_sip.c:2003 retrans_pkt: Hanging up call 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245 - no reply to our critical packet (see doc/sip-retransmit.txt).
    -- Hungup 'Zap/1-1'
  == Spawn extension (my-phones, 917070, 1) exited non-zero on 'SIP/203-103eb538'
Really destroying SIP dialog '00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245' Method: INVITE
pabx*CLI> 
<--- SIP read from 172.16.3.245:50462 --->
BYE sip:917070@172.16.3.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK7a9cdf4b
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as4793fff5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
Max-Forwards: 70
Date: Tue, 22 Sep 2009 21:01:38 GMT
CSeq: 103 BYE
User-Agent: Cisco-CP7940G/8.0
Content-Length: 0
Proxy-Authorization: Digest username="203",realm="asterisk",uri="sip:917070@172.16.3.2",response="8e94b1b451ff11f06015a1045b359670",nonce="0afa8f28",algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
Sending to 172.16.3.245 : 5061 (no NAT)

<--- Transmitting (no NAT) to 172.16.3.245:5061 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK7a9cdf4b;received=172.16.3.245
From: "Karen" <sip:203@pabx.spruce>;tag=00215553ee04001865499a04-4c1cd490
To: <sip:917070@pabx.spruce>;tag=as4793fff5
Call-ID: 00215553-ee040004-4c5d2440-28ec31bf@172.16.3.245
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>

This is my SIPDefault.cnf

# Image Version 
image_version: "P0S3-8-12-00" 

# Proxy Server 
proxy1_address: "pabx.spruce" # IP address here alternatively 
proxy2_address: "pabx.spruce"

# Proxy Server Port (default - 5060) 
proxy1_port:"5060" 

# Emergency Proxy info 
proxy_emergency: "pabx.spruce" # IP address here alternatively 
proxy_emergency_port: "5060" 

# Backup Proxy info 
proxy_backup: "pabx.spruce" 
proxy_backup_port: "5060" 

# Outbound Proxy info 
outbound_proxy: "" 
outbound_proxy_port: "5060" 

# NAT/Firewall Traversal 
nat_enable: "0" 
nat_address: "" 
voip_control_port: "5061" 
start_media_port: "16384" 
end_media_port: "32766" 
nat_received_processing: "0" 

# Proxy Registration (0-disable (default), 1-enable) 
proxy_register: "1" 

# Phone Registration Expiration [1-3932100 sec] (Default - 3600) 
timer_register_expires: "3600" 

# Codec for media stream (g711ulaw (default), g711alaw, g729) 
preferred_codec: "none" 

# TOS bits in media stream [0-5] (Default - 5) 
tos_media: "5" 

# Enable VAD (0-disable (default), 1-enable) 
enable_vad: "0" 

# Allow for the bridge on a 3way call to join remaining parties upon hangup 
cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default) 

# Allow Transfer to be completed while target phone is still ringing 
semi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (default) 

# Telnet Level (enable or disable the ability to telnet into this phone 
telnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged 

# Inband DTMF Settings (0-disable, 1-enable (default)) 
dtmf_inband: "1" 

# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always 
- always avt ) dtmf_outofband: "avt" ~np~# DTMF dB Level Settings (1-6dB down, 2
-3db down, 3-nominal (default), 4-3db up, 5-6dB up) 
dtmf_db_level: "3" 

# SIP Timers 
timer_t1: "500" ; Default 500 msec 
timer_t2: "4000" ; Default 4 sec 
sip_retx: "10" ; Default 11 
sip_invite_retx: "6" ; Default 7 
timer_invite_expires: "180" ; Default 180 sec 

# Setting for Message speeddial to UOne box 
messages_uri: "222" 

# TFTP Phone Specific Configuration File Directory 
tftp_cfg_dir: "./" 

# Time Server 
sntp_mode: "unicast" 
sntp_server: "snakebite.spruce" # IP address here alternatively 
time_zone: "GMT" 
dst_offset: "1" 
dst_start_month: "March" 
dst_start_day: "" 
dst_start_day_of_week: "Sun" 
dst_start_week_of_month: "4" 
dst_start_time: "02" 
dst_stop_month: "Oct" 
dst_stop_day: "" 
dst_stop_day_of_week: "Sunday" 
dst_stop_week_of_month: "4" 
dst_stop_time: "2" 
dst_auto_adjust: "1" 

# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no 
user control) 
dnd_control: "0" ; Default 0 (Do Not Disturb feature is off) 

# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabl
ed no user control) 
callerid_blocking: "0" ; Default 0 (Disable sending all calls as anonymous) 

# Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-
enabled no user control) 
anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls) 

# Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enable
d with no user control) 
call_waiting: "1" ; Default 1 (Call Waiting enabled) 

# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) 
dtmf_avt_payload: "101" ; Default 100 

# XML file that specifies the dialplan desired 
dial_template: "dialplan" 

# Network Media Type (auto, full100, full10, half100, half10) 
network_media_type: "auto" 

#Autocompletion During Dial (0-off, 1-on [default]) 
autocomplete: "0" 

#Time Format (0-12hr, 1-24hr [default]) 
time_format_24hr: "1" 

# URL for external Phone Services 
services_url: "http://pabx.spruce/cisco/services.php" # IP address here alternat
ively 

# URL for external Directory location 
directory_url: "http://pabx.spruce/cisco/directory.php" # IP address here altern
atively 

# URL for branding logo 
logo_url: "http://pabx.spruce/cisco/logo.bmp" # IP address here alternatively 

# Remote Party ID 
remote_party_id: 1 ; 0-Disabled (default), 1-Enabled 

And finally the SIP00215553EE04.cnf

# SIP Configuration Generic File 

# Image Version 
image_version: "P0S3-8-12-00"
phone_label: " " 

# Line 1 appearance 
line1_displayname: "Karen" 
line1_shortname:"Karen 203" 
line1_name: 203 
line1_authname: "203" 
line1_password: "-removed-" 

# Line 2 appearance 
line2_displayname: "Skatey" 
line2_shortname: "Skatey" 
line2_name: 204 
line2_authname: "204" 
line2_password: "-removed-" 

# Line 3 appearance 
line3_displayname: "" 
line3_shortname: "" 
line3_name: UNPROVISIONED 
line3_authname: "UNPROVISIONED" 
line3_password: "UNPROVISIONED" 

# Line 4 appearance 
line4_displayname: "" 
line4_shortname: "" 
line4_name: UNPROVISIONED 
line4_authname: "UNPROVISIONED" 
line4_password: "UNPROVISIONED" 

# Line 5 appearance 
line5_displayname: "" 
line5_shortname: "" 
line5_name: UNPROVISIONED 
line5_authname: "UNPROVISIONED" 
line5_password: "UNPROVISIONED" 

# Line 6 appearance 
line6_displayname: "" 
line6_shortname: "" 
line6_name: UNPROVISIONED 
line6_authname: "UNPROVISIONED" 
line6_password: "UNPROVISIONED" 

# Phone Prompt (The prompt that will be displayed on console and telnet) 
phone_prompt: "SIP Phone" ; Limited to 15 characters (Default - SIP Phone) 

# Phone Password (Password to be used for console or telnet login) 
phone_password: "cisco" ; Limited to 31 characters (Default - cisco) 

# User classifcation used when Registering [ none(default), phone, ip ] 
user_info: none

The relevant entries in sip.conf:

[203]
type=friend
context=my-phones
secret=-removed-
host=dynamic
qualify=yes
nat=no
canreinvite=no
mailbox=298

I’ve downgraded the 7940 to the version 7.5 SIP image. This will allow me to place calls with no problem. It would be nice to get the 8.x series working, but at least I can make use of the phone for now.