[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