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Summary:ASTERISK-23683: #includes - wildcard character in a path more than one directory deep - results in no config parsing on module reload
Reporter:tootai (tootai)Labels:
Date Opened:2014-04-29 10:37:44Date Closed:2014-06-05 12:38:27
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:Core/Configuration
Versions:1.8.27.0 11.9.0 Frequency of
Occurrence
Related
Issues:
is caused byASTERISK-23383 Wrong sense test on stat return code causes unchanged config check to break with include files.
is a clone ofASTERISK-23689 CLONE - SIP reload does nothing - Explanation why
duplicatesASTERISK-23819 #include in queue.conf doesn't work
Environment:Debian Wheezy amd64 kernel 3.2.0.4Attachments:
Description:After upgrade from previous versions, on 2 different asterisk servers, sip reload does nothing

[Edit by Rusty- adding below description and notes]

The below include lines work fine in 11.9.0
{noformat}
;#include local/additional_sip-general.conf  ;this works fine
;#include local/additional_sip-register.conf   ;this works fine
;#include local/sipd/sip_included.conf  ;this works fine
;#include local/*.conf    ;this works fine
{noformat}

The below include lines result in module reloads not parsing the defined configuration.
{noformat}
;#include local/sip.d/*.conf  ;this is the reporters original triggering line, does not work
;#include local/sipd/*.conf  ;this does not work either
{noformat}

I tested with SVN rev r409834, in which all of the above include lines work. After moving to r409917 (ASTERISK-23383) the issue occurs.
Comments:By: tootai (tootai) 2014-04-29 10:52:58.634-0500

Additional: does nothing means on the CLI with core set verbose 3

pabx*CLI> sip reload
Reloading SIP
pabx*CLI>

On previous version all readed files where displayed, so with this versin we don't know if the command is executed or not.

Added: on our test server we tested by mofifying sip conf then doing a sip reload and a reload module chan_sip.so => modification not taken in account. After a "core stop now" and "service asterisk start", modifications are taken in account.

By: Matt Jordan (mjordan) 2014-04-29 15:52:50.422-0500

Works on my machine:

{noformat}
*CLI> !touch /etc/asterisk/sip.conf
*CLI> sip reload
Reloading SIP
 == Parsing '/etc/asterisk/sip.conf':   == Found
*CLI>   == Parsing '/etc/asterisk/users.conf':   == Found
 == Using SIP CoS mark 4
 == Parsing '/etc/asterisk/sip_notify.conf':   == Found
{noformat}

Touch and reload reloads {{sip.conf}} and all associated {{.conf}} files.

{noformat}

*CLI> sip show settings


Global Settings:
----------------
 UDP Bindaddress:        0.0.0.0:5060
 TCP SIP Bindaddress:    Disabled
 TLS SIP Bindaddress:    Disabled
 Videosupport:           No
 Textsupport:            No
 Ignore SDP sess. ver.:  No
 AutoCreate Peer:        No
 Match Auth Username:    No
 Allow unknown access:   Yes
 Allow subscriptions:    Yes
 Allow overlap dialing:  No
 Allow promisc. redir:   No
 Enable call counters:   No
 SIP domain support:     No
 Realm. auth:            No
 Our auth realm          asterisk
 Use domains as realms:  No
 Call to non-local dom.: Yes
 URI user is phone no:   No
 Always auth rejects:    Yes
 Direct RTP setup:       No
 User Agent:             Asterisk PBX SVN-branch-1.8-r412922
 SDP Session Name:       Asterisk PBX SVN-branch-1.8-r412922
 SDP Owner Name:         root
 Reg. context:           (not set)
 Regexten on Qualify:    No
 Legacy userfield parse: No
 Caller ID:              asterisk
 From: Domain:          
 Record SIP history:     Off
 Call Events:            Off
 Auth. Failure Events:   Off
 T.38 support:           No
 T.38 EC mode:           Unknown
 T.38 MaxDtgrm:          -1
 SIP realtime:           Disabled
 Qualify Freq :          60000 ms
 Q.850 Reason header:    No
 Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
 IP ToS SIP:             CS0
 IP ToS RTP audio:       CS0
 IP ToS RTP video:       CS0
 IP ToS RTP text:        CS0
 802.1p CoS SIP:         4
 802.1p CoS RTP audio:   5
 802.1p CoS RTP video:   6
 802.1p CoS RTP text:    5
 Jitterbuffer enabled:   No

Network Settings:
---------------------------
 SIP address remapping:  Disabled, no localnet list
 Externhost:             <none>
 Externaddr:             (null)
 Externrefresh:          10

Global Signalling Settings:
---------------------------
 Codecs:                 0x80000008000e (gsm|ulaw|alaw|h263|testlaw)
 Codec Order:            none
 Relax DTMF:             No
 RFC2833 Compensation:   No
 Symmetric RTP:          No
 Compact SIP headers:    No
 RTP Keepalive:          0 (Disabled)
 RTP Timeout:            0 (Disabled)
 RTP Hold Timeout:       0 (Disabled)
 MWI NOTIFY mime type:   application/simple-message-summary
 DNS SRV lookup:         Yes
 Pedantic SIP support:   Yes
 Reg. min duration       60 secs
 Reg. max duration:      3600 secs
 Reg. default duration:  120 secs
 Outbound reg. timeout:  20 secs
 Outbound reg. attempts: 0
 Outbound reg. retry 403:0
 Notify ringing state:   Yes
   Include CID:          No
 Notify hold state:      No
 SIP Transfer mode:      open
 Max Call Bitrate:       384 kbps
 Auto-Framing:           No
 Outb. proxy:            <not set>
 Session Timers:         Accept
 Session Refresher:      uas
 Session Expires:        1800 secs
 Session Min-SE:         90 secs
 Timer T1:               500
 Timer T1 minimum:       100
 Timer B:                32000
 No premature media:     Yes
 Max forwards:           70

Default Settings:
-----------------
 Allowed transports:     UDP
 Outbound transport:  UDP
 Context:                public
 Force rport:            Yes
 DTMF:                   rfc2833
 Qualify:                0
 Use ClientCode:         No
 Progress inband:        Never
 Language:              
 MOH Interpret:          default
 MOH Suggest:            
 Voice Mail Extension:   asterisk

----
{noformat}

Change the {{context}} in {{[global]}} to {{public_1}} and reload:

{noformat}

*CLI> !sudo vim /etc/asterisk/sip.conf
*CLI> sip reload
*CLI>  Reloading SIP
 == Parsing '/etc/asterisk/sip.conf':   == Found
 == Parsing '/etc/asterisk/users.conf':   == Found
 == Using SIP CoS mark 4
 == Parsing '/etc/asterisk/sip_notify.conf':   == Found

*CLI> sip show settings


Global Settings:
----------------
 UDP Bindaddress:        0.0.0.0:5060
 TCP SIP Bindaddress:    Disabled
 TLS SIP Bindaddress:    Disabled
 Videosupport:           No
 Textsupport:            No
 Ignore SDP sess. ver.:  No
 AutoCreate Peer:        No
 Match Auth Username:    No
 Allow unknown access:   Yes
 Allow subscriptions:    Yes
 Allow overlap dialing:  No
 Allow promisc. redir:   No
 Enable call counters:   No
 SIP domain support:     No
 Realm. auth:            No
 Our auth realm          asterisk
 Use domains as realms:  No
 Call to non-local dom.: Yes
 URI user is phone no:   No
 Always auth rejects:    Yes
 Direct RTP setup:       No
 User Agent:             Asterisk PBX SVN-branch-1.8-r412922
 SDP Session Name:       Asterisk PBX SVN-branch-1.8-r412922
 SDP Owner Name:         root
 Reg. context:           (not set)
 Regexten on Qualify:    No
 Legacy userfield parse: No
 Caller ID:              asterisk
 From: Domain:          
 Record SIP history:     Off
 Call Events:            Off
 Auth. Failure Events:   Off
 T.38 support:           No
 T.38 EC mode:           Unknown
 T.38 MaxDtgrm:          -1
 SIP realtime:           Disabled
 Qualify Freq :          60000 ms
 Q.850 Reason header:    No
 Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
 IP ToS SIP:             CS0
 IP ToS RTP audio:       CS0
 IP ToS RTP video:       CS0
 IP ToS RTP text:        CS0
 802.1p CoS SIP:         4
 802.1p CoS RTP audio:   5
 802.1p CoS RTP video:   6
 802.1p CoS RTP text:    5
 Jitterbuffer enabled:   No

Network Settings:
---------------------------
 SIP address remapping:  Disabled, no localnet list
 Externhost:             <none>
 Externaddr:             (null)
 Externrefresh:          10

Global Signalling Settings:
---------------------------
 Codecs:                 0x80000008000e (gsm|ulaw|alaw|h263|testlaw)
 Codec Order:            none
 Relax DTMF:             No
 RFC2833 Compensation:   No
 Symmetric RTP:          No
 Compact SIP headers:    No
 RTP Keepalive:          0 (Disabled)
 RTP Timeout:            0 (Disabled)
 RTP Hold Timeout:       0 (Disabled)
 MWI NOTIFY mime type:   application/simple-message-summary
 DNS SRV lookup:         Yes
 Pedantic SIP support:   Yes
 Reg. min duration       60 secs
 Reg. max duration:      3600 secs
 Reg. default duration:  120 secs
 Outbound reg. timeout:  20 secs
 Outbound reg. attempts: 0
 Outbound reg. retry 403:0
 Notify ringing state:   Yes
   Include CID:          No
 Notify hold state:      No
 SIP Transfer mode:      open
 Max Call Bitrate:       384 kbps
 Auto-Framing:           No
 Outb. proxy:            <not set>
 Session Timers:         Accept
 Session Refresher:      uas
 Session Expires:        1800 secs
 Session Min-SE:         90 secs
 Timer T1:               500
 Timer T1 minimum:       100
 Timer B:                32000
 No premature media:     Yes
 Max forwards:           70

Default Settings:
-----------------
 Allowed transports:     UDP
 Outbound transport:  UDP
 Context:                public_1
 Force rport:            Yes
 DTMF:                   rfc2833
 Qualify:                0
 Use ClientCode:         No
 Progress inband:        Never
 Language:              
 MOH Interpret:          default
 MOH Suggest:            
 Voice Mail Extension:   asterisk

----
*CLI>
{noformat}

By: tootai (tootai) 2014-04-30 03:22:27.182-0500

On the test server running asterisk 11.9.0 I reinstall asterisk from tar.gz file (original version is a 11.8.0 patched) downloaded this morning from asterisk.org => same result

I switch back to 11.8,.1 => everything is OK.

The problem is not only with sip reload, same happend to iax2 reload or voicemail reload. Dialplan reload is OK.

Also, help in CLI doesn't show iax help for example.

If you would like, I can give you ssh access to our test server.

By: tootai (tootai) 2014-04-30 04:58:57.490-0500

I start a new install from 11.8.0: make clean && ./configure && make && make install => everything is OK

The same with 11.9.0 => problem appears

Definitely, something is wrong.

Daniel
By: tootai (tootai) 2014-04-30 05:35:45.122-0500

I installed 1.8.27 version on a customer site one week ago and checked this problem: guess what, it's the same, no reload!

Debian Squeeze  2.6.32-5-amd64, real machine.


By: tootai (tootai) 2014-04-30 05:57:54.058-0500

OK, I got it. The way you are doing is working because you are modifying sip.conf. The reload is taken in account *ONLY* if modification is done in sip.conf! It this file contain includes -our case- and you modify one of this includes, no reload

By: Richard Mudgett (rmudgett) 2014-05-29 18:11:47.834-0500

A patch for v1.8 is up on reviewboard here:
https://reviewboard.asterisk.org/r/3575/