asterisk users - Mar 2005 - Polycom IP600 Cannot answer

I googled and googled but could not find anything regarding this problem.

I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6
months, no problems with my grandstreams. I'm fairly familiar with the
ins and outs of asterisk...

IP600 with latest sip 1.4.1 and bootrom from my FTP server.
Standard config files from http://www.freedomphones.net/polycom/files/
No changes other than typical ip address of phone and server.

Grandstream (192.168.2.20) is exten 2000, Polycom (192.168.2.22) is 2006.


I can make calls out to my Grandstreams from the Polycom all day. No
problem.
When I try to call the Polycom I get this stuff:
    -- Executing Dial("SIP/2000-972f", "SIP/2006|10|r") in
new stack
    -- Called 2006
    -- SIP/2006-f8ea is ringing
    -- SIP/2006-f8ea answered SIP/2000-972f
    -- Attempting native bridge of SIP/2000-972f and SIP/2006-f8ea
    -- Got SIP response 481 "No Such Call" back from 192.168.2.20
  == Spawn extension (from-sip, 2006, 1) exited non-zero on
'SIP/2000-972f'
    -- Got SIP response 500 "Internal Server Error" back from
192.168.2.22
    -- Got SIP response 500 "Internal Server Error" back from
192.168.2.22

When I answer the polycom it just hangs up and hangs the grandstream
online. I have to manually hang up the grandstream. It doesn't get a SIP
notifcation of call failure or hangup.

When I tcpdump the asterisk box, I can see RTP streams from the
Grandstream toward the server. But nothing coming from or toward the
Polycom. When I call the Grandstream from the Polycom, the call connects
and I see both RTP streams to and from the Asterisk box for both phones
and everything is happy.

anyone have any ideas as to why inbound calls fail?

I've tried several combinations of
friend/peer/progressinband/canreinvite etc... No change at all.

Here's my sip.conf for the Polycom
[2006]
type=friend
username=2006
secret=2006
host=dynamic
dtmfmode=rfc2833
defaultip=192.168.2.22
progressinband=no
context=from-sip
mailbox=2006@local
callgroup=1
pickupgroup=1


thank you for any insight!

Mark

MDS wrote:
> I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6
> months, no problems with my grandstreams. I'm fairly familiar with the
> ins and outs of asterisk...
If you are going to use CVS HEAD, you _must_ stay up to date. There have 
been a large number of SIP-related fixes in CVS HEAD since the 19th. 
Please update your system and try again before reporting problems. Thanks!

I don't think you want both dynamic and defaultip set

But that should not cause what you describe. I hvae seen other issues 
with head. Perhaps checkout the latest?


On Mar 30, 2005, at 12:29 AM, MDS wrote:
> I googled and googled but could not find anything regarding this 
> problem.
>
> I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6
> months, no problems with my grandstreams. I'm fairly familiar with the
> ins and outs of asterisk...
>
> IP600 with latest sip 1.4.1 and bootrom from my FTP server.
> Standard config files from http://www.freedomphones.net/polycom/files/
> No changes other than typical ip address of phone and server.
>
> Grandstream (192.168.2.20) is exten 2000, Polycom (192.168.2.22) is 
> 2006.
>
>
> I can make calls out to my Grandstreams from the Polycom all day. No
> problem.
> When I try to call the Polycom I get this stuff:
>     -- Executing Dial("SIP/2000-972f", "SIP/2006|10|r")
in new stack
>     -- Called 2006
>     -- SIP/2006-f8ea is ringing
>     -- SIP/2006-f8ea answered SIP/2000-972f
>     -- Attempting native bridge of SIP/2000-972f and SIP/2006-f8ea
>     -- Got SIP response 481 "No Such Call" back from 192.168.2.20
>   == Spawn extension (from-sip, 2006, 1) exited non-zero on 
> 'SIP/2000-972f'
>     -- Got SIP response 500 "Internal Server Error" back from 
> 192.168.2.22
>     -- Got SIP response 500 "Internal Server Error" back from 
> 192.168.2.22
>
> When I answer the polycom it just hangs up and hangs the grandstream
> online. I have to manually hang up the grandstream. It doesn't get a 
> SIP
> notifcation of call failure or hangup.
>
> When I tcpdump the asterisk box, I can see RTP streams from the
> Grandstream toward the server. But nothing coming from or toward the
> Polycom. When I call the Grandstream from the Polycom, the call 
> connects
> and I see both RTP streams to and from the Asterisk box for both phones
> and everything is happy.
>
> anyone have any ideas as to why inbound calls fail?
>
> I've tried several combinations of
> friend/peer/progressinband/canreinvite etc... No change at all.
>
> Here's my sip.conf for the Polycom
> [2006]
> type=friend
> username=2006
> secret=2006
> host=dynamic
> dtmfmode=rfc2833
> defaultip=192.168.2.22
> progressinband=no
> context=from-sip
> mailbox=2006@local
> callgroup=1
> pickupgroup=1
>
>
> thank you for any insight!
>
> Mark
>
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