[Asterisk-Users] Polycom IP600 Cannot answer

Jerry jjones at quiddesign.com
Wed Mar 30 07:22:43 MST 2005


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 at local
> callgroup=1
> pickupgroup=1
>
>
> thank you for any insight!
>
> Mark
>
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