[Home]

Summary:ASTERISK-06975: No reply to PUBLISH requests
Reporter:alexb (alexb)Labels:
Date Opened:2006-05-15 03:52:10Date Closed:2006-07-26 12:01:12
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) SNOM-active_calls_2.log
( 1) SNOM-active_calls_3.log
( 2) SNOM-active_calls.log
Description:Dialing on SNOM phones keypad when "Number Guessing" is set to "On" (default setting) causes memory consumption because

1 digit => 1 active SIP call which will never expire!

I have a server with hundreds and hundreds of active SIP calls at the end of the day.

Please note that on high-load servers, this is a critical issue.
Comments:By: Olle Johansson (oej) 2006-05-16 03:42:12

THat's interesting, will take a look at that. How did you configure this on your SNOM?
By: alexb (alexb) 2006-05-16 04:46:55

Do you mean where to configure number guessing on Snom phones? It's under Advanced Settings/Phone behavior.

FYI, firmware rev. is 5.5, and in particular:

Application-Version: snom320-SIP 5.5
Rootfs-Version:      snom320 jffs2 v3.36
By: Kevin P. Fleming (kpfleming) 2006-05-18 16:25:51

This makes no sense. Your phone is sending PUBLISH to the Asterisk server, which we don't support. Asterisk has explicit code to reply with '501 Method Not Implemented' and generate a log message, but that is not appearing in your SIP trace and log.

Are you absolutely sure you are running Asterisk 1.2.7.1?
By: alexb (alexb) 2006-05-20 05:19:34

Of course!

efw-voiceone*CLI> show version
Asterisk 1.2.7.1 built by root @ powerplant on a i686 running Linux on 2006-04-14 06:33:55 UTC
By: Kevin P. Fleming (kpfleming) 2006-05-24 15:49:12

OK, I don't understand how Asterisk is not generating a response to those packets.

Can you generate a new console trace, this time with 'sip debug' _and_ 'set debug 10' in effect? Also ensure that the 'debug' channel is enabled for the console in logger.conf.
By: alexb (alexb) 2006-05-25 06:17:08

Here you are!
By: Olle Johansson (oej) 2006-06-06 03:31:27

Agree, this is very strange.
By: Serge Vecher (serge-v) 2006-06-12 19:38:13

AlexB: since the latest version of stable is now 1.2.9.1, can you please upgrade and produce a new sip debug trace?
By: alexb (alexb) 2006-06-19 04:32:17

I've just uploaded the new trace.

FYI:

efw-voiceone*CLI> show version
Asterisk 1.2.9.1 built by root @ powerplant on a i686 running Linux on 2006-06-07 05:30:21 UTC

SNOM:
System Information:
Phone Type: snom320-SIP
MAC-Address:
IP-Address: 192.168.2.201
Kernel Version: Linux 192.168.0.101 2.4.20 ASTERISK-39 Tue Aug 30 12:21:18 CEST 2005 mips
Application-Version: snom320-SIP 6.2
Rootfs-Version: snom320 jffs2 v3.36
By: Olle Johansson (oej) 2006-06-26 16:00:14

Wonder if this is a locking issue, since the retransmit comes rather quickly from the phone... I can't figure this one out. Will have to test with a SNOM phone.
By: Tilghman Lesher (tilghman) 2006-06-27 01:17:55

On your upload log, could you include not just debug, but also error, warning, and notice logging levels?  Also, are you running with pedantic=yes in sip.conf?
By: alexb (alexb) 2006-06-27 02:52:30

Hi,

OK, I'll upload a new trace as soon as possible.

As regards pedantic, I didn't set it explicitly, so I think it's running with the default pedantic=no.
By: Olle Johansson (oej) 2006-06-27 11:26:47

Please include a "sip show settings" as well as all the logging channels. Thanks.
By: Olle Johansson (oej) 2006-06-27 11:50:12

I have tried sending PUBLISH requests both to 1.2 and svn trunk. In both cases, I immediately get a response "501 Method Not Implemented" and the SIP pvt is destroyed...
By: alexb (alexb) 2006-06-27 12:21:22

Starting with sip show settings...


Global Settings:
----------------
 SIP Port:               5060
 Bindaddress:            0.0.0.0
 Videosupport:           No
 AutoCreatePeer:         No
 Allow unknown access:   Yes
 Promsic. redir:         No
 SIP domain support:     No
 Call to non-local dom.: Yes
 URI user is phone no:   No
 Our auth realm          mydomain.tld
 Realm. auth:            No
 Always auth rejects:    No
 User Agent:             Asterisk PBX
 MWI checking interval:  10 secs
 Reg. context:           (not set)
 Caller ID:              asterisk
 From: Domain:
 Record SIP history:     On
 Call Events:            Off
 IP ToS:                 0x10
 OSP Support:            No
 SIP realtime:           Enabled

Global Signalling Settings:
---------------------------
 Codecs:                 alaw,ulaw,gsm,ilbc,g729
 Relax DTMF:             No
 Compact SIP headers:    No
 RTP Timeout:            0 (Disabled)
 RTP Hold Timeout:       0 (Disabled)
 MWI NOTIFY mime type:   application/simple-message-summary
 DNS SRV lookup:         Yes
 Pedantic SIP support:   No
 Reg. max duration:      3600 secs
 Reg. default duration:  120 secs
 Outbound reg. timeout:  20 secs
 Outbound reg. attempts: 0
 Notify ringing state:   Yes

Default Settings:
-----------------
 Context:                DefaultIncomingRule
 Nat:                    Always
 DTMF:                   auto
 Qualify:                2000
 Use ClientCode:         No
 Progress inband:        Never
 Language:               (Defaults to English)
 Musicclass:             default
 Voice Mail Extension:   test

Realtime SIP Settings:
----------------------
 Realtime Peers:         Yes
 Realtime Users:         Yes
 Cache Friends:          Yes
 Update:                 Yes
 Ignore Reg. Expire:     No
 Auto Clear:             120

----
By: Olle Johansson (oej) 2006-06-28 10:52:25

Which realtime driver?
By: alexb (alexb) 2006-06-29 02:38:48

mysql
By: Olle Johansson (oej) 2006-06-30 03:04:35

Can you please try without the MYSQL realtime driver and see if we get proper replies then? There has been several very strange bugs reported that was caused by this driver. Maybe you can try using ODBC instead to connect to MySQL or just try without it once.
By: Serge Vecher (serge-v) 2006-07-12 13:56:52

AlexB: no response for two weeks!
By: alexb (alexb) 2006-07-13 02:20:51

I've been out of office so far, and I'm still away.
By: alexb (alexb) 2006-07-19 10:46:03

Hi,

seems it was related to compiler or hardware.

I changed hardware and recompiled Asterisk 1.2.10 w/o optimizations and now it correctly reply with 501 to PUBLISH.

I think you can close this issue.
Thanks.
By: alexb (alexb) 2006-07-24 04:54:58

I reopen this issue because I understood the reason of such behaviour.

I completely forgot that, when compiling Asterisk on the old hardware, I automatically apply the patch for call pickup found at http://bugs.digium.com/view.php?id=5014 (pickup-mgernoth-2006-02-28.patch.txt).
However, sip channels remain open only with recordhistory=yes (see my previous post which showed my sip settings: "Record SIP history: On").
Last but not least, the NOTICE reply to the PUBLISH command is removed by this patch, too.

I think that oej should be informed.

Thanks,
Alex
By: Jason Parker (jparker) 2006-07-26 11:47:32

Per kpfleming, this bug should be closed, since it involves a patch from ASTERISK-4887 that has not been merged to trunk.  If there are issues with that patch, please post details on that bug.
By: Tilghman Lesher (tilghman) 2006-07-26 11:48:04

We cannot be expected to fix problems with unmerged patches.