asterisk users - Oct 2009 - Calls being dropped - Cisco 7940 with SIP 8.12 image

Hi everyone,

I hope someone can help me with a problem I'm having with Cisco 7940  
phones on the SIP 8.12 image.  When I place a call from one of the  
handsets, the call proceeds as normal for 20 seconds and is then  
terminated by Asterisk (1.4.26.2):


[Oct  3 10:08:55] WARNING[1650]: chan_sip.c:1981 retrans_pkt: Maximum  
retries exceeded on transmission 00215553- 
ee04000c-0b5bc1f8-3407d9f7 at 172.16.3.245 for seqno 102 (Critical  
Response) -- See doc/sip-retransmit.txt.
[Oct  3 10:08:55] WARNING[1650]: chan_sip.c:2003 retrans_pkt: Hanging  
up call 00215553-ee04000c-0b5bc1f8-3407d9f7 at 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/ 
200-103fa658'


Turning on SIP debugging shows that it tries to send the following  
data to the 7940 six times before giving up:


<--- 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=z9hG4bK1d4425f3;received=172.16.3.245
From: "James" <sip: 
200 at pabx.spruce>;tag=00215553ee040030116ccaac-32c56370
To: <sip:917070 at pabx.spruce>;tag=as680ce289
Call-ID: 00215553-ee04000c-0b5bc1f8-3407d9f7 at 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 at 172.16.3.2>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 1622 1622 IN IP4 172.16.3.2
s=session
c=IN IP4 172.16.3.2
t=0 0
m=audio 12388 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


The following was observed on the 7940's telnet console:


SIP Phone>  Warning: Unrecognized attribute (silenceSupp)  Warning:  
Unrecognized attribute (silenceSupp) sip_sm_ccb_match_branch_cseq:  
Method index not found
SIPTaskProcessSIPMessage: Error: sip_sm_determine_ccb(): bad response.  
Dropping message.


As far as I can tell, the 'a=silenceSupp:off - - - -' header is not  
accepted by the 7940, which seems like a bug in the SIP image to me.   
However, I can't find a way to report this problem to Cisco without a  
support contract (which I do not have).  Reverting to version 7.5  
fixes the problem, but it is still present in 8.11.  The problem is  
not present if the PSTN initiates the call, nor is it present if I  
allow the handsets to reinvite each other.  Here's the sip.conf  
snippet if it helps:


[general]
port = 5060
bindaddr = 0.0.0.0
context = others
localnet=172.16.3.0/24

[200]
type=friend
host=dynamic
dtmfmode=rfc2833
nat=no
username=200
secret=*removed*
context=my-phones
canreinvite=no


Anyone else encountered this problem or have a workaround?

Regards,
James.

On Sat, Oct 3, 2009 at 6:17 AM, James Stocks <stocksy at stocksy.co.uk>
wrote:
> Hi everyone,
>
> I hope someone can help me with a problem I'm having with Cisco 7940
> phones on the SIP 8.12 image. ?When I place a call from one of the
> handsets, the call proceeds as normal for 20 seconds and is then
> terminated by Asterisk (1.4.26.2):
>
We are runing 08-12-00 on 7940/60s just fine (Asterisk 1.6.1.1), and
have been for a while.
>
> As far as I can tell, the 'a=silenceSupp:off - - - -' header is not
> accepted by the 7940, which seems like a bug in the SIP image to me.
> However, I can't find a way to report this problem to Cisco without a
> support contract (which I do not have). ?Reverting to version 7.5
> fixes the problem, but it is still present in 8.11. ?The problem is
> not present if the PSTN initiates the call, nor is it present if I
> allow the handsets to reinvite each other. ?Here's the sip.conf
> snippet if it helps:
>
That all looks fine to me.  What do your SIPDefault.cnf and
SIP<MAC>.cnf files look like?

-Jonathan