[asterisk-users] No reply to our critical packet

[Andrew Joakimsen]

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
public with no NAT... everything works on the Asterisk end just fine
EXCEPT that I can never check voice mail

After about 30 seconds the call drops with these messagess:

[Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2 (Critical
Response)
[Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging
up call 320893f1-50c13ba3-78c26164 at 192.168.1.54 - no reply to our
critical packet.

It seems to me that the problem is the way Asterisk is handling this
"critical packet" -- of course it can not be sent to 192.168.1.54, the
phone is at that IP behind a NAT and the Asterisk server is not. I can
make any other phone call from this same phone as long as it is not
voicemail and I can be on the line for hours with no problem.

I am really at a loss here. I have searched a bit and come up with
nothing other than blaming the UA. I know the Polycoms dont have the
best NAT support but besides this it works problem-free. It's odd I
can make a call anywhere else even for hours and not have any issues
at all but 30 seconds into a voicemail call it just drops....


app5*CLI> sip show peer 17865221569
app5*CLI>

  * Name       : 17865221569
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : blended-lcr
  Subscr.Cont. : sla_stations
  Language     : en
  AMA flags    : Unknown
  Transfer mode: closed
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      : 17865221569
  VM Extension : 14193016245
  LastMsgsSent : 0/0
  Call limit   : 2
  Dynamic      : Yes
  Callerid     : "" <CENSORED>
  MaxCallBR    : 256 kbps
  Expire       : 63
  Insecure     : no
  Nat          : Always
  ACL          : No
  T38 pt UDPTL : Yes
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : Yes
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 74.CENSORED.213 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Reg. exten   :
  Def. Username: 17865221569
  SIP Options  : (none)
  Codecs       : 0x104 (ulaw|g729)
  Codec Order  : (g729:20,ulaw:20)
  Auto-Framing:  No
  Status       : OK (130 ms)
  Useragent    : PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
  Reg. Contact : sip:17865221569 at 192.168.1.54


app5*CLI> core show version
Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on
2008-07-09 01:41:43 UTC