[asterisk-users] No reply to our critical packet

Mon Oct 6 19:26:35 CDT 2008

On Tue, Oct 7, 2008 at 2:22 AM, Andrew Joakimsen <joakimsen at gmail.com> wrote:
> The odd thing is on this particular phone it only happens when you
> call voicemail.
>
> It is certainly a bug in Asterisk, not the UA. Asterisk is trying to
> send to 192.168.1.x which obviously is not possible. Something in the
> NAT support is not working right.

Hi,

You should get SIP traces to see why Asterisk is trying to reply to 192.168.1.x.

To do this, enter "sip set debug on" in asterisk CLI, and post us a
log of call reaching voicemail and disconnecting.

Regards,
Atis

>
> On Mon, Oct 6, 2008 at 3:06 PM, SIP <sip at arcdiv.com> wrote:
>> This message is usually caused by Asterisk not receiving an ACK after
>> about 30 seconds of attempts. There are countless misconfigured UAs and
>> proxies out there that don't handle ACK well, so it would be nice to be
>> able to turn this 'feature' off. What's annoying is that the explanation
>> has always been "If we can't get an ACK, we can't send any RTP data."
>> This is patently false, as the RTP will often work fine even if ACK
>> handling is misconfigured (we see it all the time).
>>
>> But alas. As far as I can tell, there's no way to disable this check. I
>> suppose I could code around it, but not being the world's most
>> proficient C coder, I'm always afraid I'll break something else. ;)
>>
>> N.
>>
>>
>> Andrew Joakimsen wrote:
>>> 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
>>>
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>>
>>
>> _______________________________________________
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>>
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>> Register Now: http://www.astricon.net
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>>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835