[asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE

[Tong]

According to your console output it looks like there is some major latency.  What is the average ping time from your asterisk machine to the polycom phone?
  ----- Original Message ----- 
  From: Rana Dutt 
  To: Asterisk Users 
  Sent: Sunday, July 16, 2006 6:51 PM
  Subject: [asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE


  I have a customer with a Polycom 501 phone behind a NAT. His phone is connected to his Netgear router at home which in turn is connected to his cable modem. The phone is set up to register with our remote Asterisk server which is on a public, static IP address, with no NAT. 

  If we set qualify=yes, our Asterisk console shows his extension becoming UNREACHABLE for a minute, then REACHABLE for a minute, then UNREACHABLE again, in an endless cycle. If we try to call the phone while it is UNREACHABLE, the phone never rings and the call goes straight to voice mail. This is very annoying. 

  If we set qualify=no, then if we try to call the phone, the phone sometimes does not ring at all, and we hear silence. The call eventually goes to voice mail. This is equally annoying to the customer.

  What is the solution to this problem? We have other customers with Polycom phones behind NAT, and they don't have this problem. Will we have better luck if we replace the Polycom with a Linksys 942 phone? 

  Here is some console output:

  Jul 16 21:44:24 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE!  Last qualify: 174
  Jul 16 21:45:33 NOTICE[19981]: chan_sip.c:9697 handle_response_peerpoke: Peer '280' is now REACHABLE! (3181ms / 5000ms) 
  Jul 16 21:47:37 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE!  Last qualify: 175

  Here is the way the phone is set up in sip.conf:

  [280]
  type=peer
  username=280
  secret=280
  host=dynamic
  dtmfmode=rfc2833
  callerid="John" <280>
  context=company_x
  mailbox=280
  nat=yes
  canreinvite=no
  qualify=5000

  We are using Asterisk 1.2.5 with standard .conf files. We are not using realtime or databases. Any help would be highly appreciated. 

  Rana Dutt
  Softel Solutions
  rdutt at softelinc.com



------------------------------------------------------------------------------


  _______________________________________________
  --Bandwidth and Colocation provided by Easynews.com --

  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
     http://lists.digium.com/mailman/listinfo/asterisk-users



------------------------------------------------------------------------------


  No virus found in this incoming message.
  Checked by AVG Free Edition.
  Version: 7.1.394 / Virus Database: 268.10.1/389 - Release Date: 7/14/2006
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060716/2f5782d8/attachment.htm