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Summary:ASTERISK-16759: Several issues with app_meetme (realtime) after upgrading from 1.6.1.20 to 1.6.2.13
Reporter:Maciej Krajewski (jamicque)Labels:
Date Opened:2010-10-01 05:57:08Date Closed:2011-05-30 07:58:45
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Applications/app_meetme
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:After upgrading form 16.1.20 to 1.6.2.13 moh stopped working,
conference room asks for pin, even if no is defined (related to 0017908, however this patch didn't help)
next conference room asks to record name, it never happended before with meetme options scM
after entering the conference room moh is not played.

The same thing happens in 1.6.2.14-rc1.

Meetme is configured via relatime.
<pre>
confno | name  | description | username | domain | pin | adminpin | members | max_users | config | announce | moh  | silence
--------+-------+-------------+----------+--------+-----+----------+---------+-----------+--------+----------+------+---------
1      | ddddd |             | admin    |        |     | 1367     |       0 |         0 | true   | true     | true | flase
</pre>


****** ADDITIONAL INFORMATION ******

 == Using SIP RTP TOS bits 136
 == Using SIP RTP CoS mark 4
 == Using SIP VRTP TOS bits 136
 == Using SIP VRTP CoS mark 4
 == Using UDPTL TOS bits 136
 == Using UDPTL CoS mark 4
   -- Executing [25@CALLEX:1] GotoIf("SIP/test003-00000001", "0?3") in new stack
   -- Executing [25@CALLEX:2] Set("SIP/test003-00000001", "__ORGDEST=25") in new stack
   -- Executing [25@CALLEX:3] AGI("SIP/test003-00000001", "agi://127.0.0.1/script.agi") in new stack
   -- AGI Script Executing Application: (Set) Options: (CHANNEL(language)=pl)
   -- AGI Script Executing Application: (Set) Options: (CALLERID(all)="test003"<13>)
   -- AGI Script Executing Application: (Set) Options: (CDR(is_inbound)=true)
   -- AGI Script Executing Application: (Set) Options: (CDR(is_outbound)=true)
   -- AGI Script Executing Application: (Set) Options: (CDR(number_a)=Unknown)
   -- AGI Script Executing Application: (Set) Options: (CDR(number_b)=-25)
   -- AGI Script Executing Application: (Set) Options: (CDR(src_interface)=CALLEX/test003)
   -- AGI Script Executing Application: (Set) Options: (CDR(dst_interface)=CALLEX/CONFERENCE)
   -- AGI Script Executing Application: (Set) Options: (CDR(src_account)=test003)
   -- AGI Script Executing Application: (Set) Options: (CDR(dst_account)=ddddd)
   -- AGI Script Executing Application: (Meetme) Options: (1,scM)
 == Parsing '/etc/asterisk/meetme.conf':   == Found
   -- Created MeetMe conference 1023 for conference '1'
   -- <SIP/test003-00000001> Playing 'conf-getpin.alaw' (language 'pl')
      > Starting recording of MeetMe Conference 1 into file ..
   -- <SIP/test003-00000001> Playing 'vm-rec-name.gsm' (language 'pl')
   -- <SIP/test003-00000001> Playing 'beep.gsm' (language 'pl')
   -- x=0, open writing:  /var/spool/asterisk/meetme/meetme-username-1-1 format: sln, 0x840efb0
   -- User ended message by pressing #
   -- <SIP/test003-00000001> Playing 'auth-thankyou.gsm' (language 'pl')
   -- <SIP/test003-00000001> Playing 'conf-onlyperson.alaw' (language 'pl')
Comments:By: Maciej Krajewski (jamicque) 2010-10-15 09:01:18

I've noticed that from time to time. I'm getting such logs when calling to meetme

 == Using SIP RTP TOS bits 136
 == Using SIP RTP CoS mark 4
 == Using SIP VRTP TOS bits 136
 == Using SIP VRTP CoS mark 4
 == Using UDPTL TOS bits 136
 == Using UDPTL CoS mark 4
   -- Executing [25@CALLEX:1] GotoIf("SIP/test001-0000003d", "0?3") in new stack
   -- Executing [25@CALLEX:2] Set("SIP/test001-0000003d", "__ORGDEST=25") in new stack
   -- Executing [25@CALLEX:3] AGI("SIP/test001-0000003d", "agi://127.0.0.1/script.agi") in new stack
   -- AGI Script Executing Application: (Set) Options: (CHANNEL(language)=pl)
   -- AGI Script Executing Application: (Set) Options: (CALLERID(all)="test001"<11>)
   -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=ALLOWED_PASSED_SCREEN)
   -- AGI Script Executing Application: (Set) Options: (CDR(is_inbound)=true)
   -- AGI Script Executing Application: (Set) Options: (CDR(is_outbound)=true)
   -- AGI Script Executing Application: (Set) Options: (CDR(number_a)=Unknown)
   -- AGI Script Executing Application: (Set) Options: (CDR(number_b)=-25)
   -- AGI Script Executing Application: (Set) Options: (CDR(src_interface)=CALLEX/test001)
   -- AGI Script Executing Application: (Set) Options: (CDR(dst_interface)=CALLEX/CONFERENCE)
   -- AGI Script Executing Application: (Set) Options: (CDR(src_account)=test001)
   -- AGI Script Executing Application: (Set) Options: (CDR(dst_account)=test)
   -- AGI Script Executing Application: (Meetme) Options: (1,rscM)
[2010-10-15 15:59:31] WARNING[6657]: app.c:1843 parse_options: Missing closing parenthesis for argument '7' in string '¨?ýÉ?&XH?`?V?(¨ ¨ô?V?`?V?¸Ž?ýÉ?&XH?`?V?¸Ž°Ž8?2??2?'
   -- AGI Script Executing Application: (Hangup) Options: ()
By: Andrew Thomas (raffles) 2010-10-18 03:41:10

Isn't this similar to https://issues.asterisk.org/view.php?id=18006 ?
By: Maciej Krajewski (jamicque) 2010-10-18 04:12:15

probably yes.
app_meetme simply does not work properly. It's a major bug.
By: Maciej Krajewski (jamicque) 2010-10-20 16:58:42

in the newest svn 1.6.2 branch it works fine
By: Maciej Krajewski (jamicque) 2010-10-21 02:47:41

Sorry, wrong info it was a coincidence that it worked, problem still occurs.
By: Maciej Krajewski (jamicque) 2010-10-21 02:51:37

Problem seems to occur only if "r" option is enabled
By: Maciej Krajewski (jamicque) 2010-10-22 09:47:37

patch https://issues.asterisk.org/view.php?id=18182 fixes the issue with strange signes
[2010-10-15 15:59:31] WARNING[6657]: app.c:1843 parse_options: Missing closing parenthesis for argument '7' in string '¨?ýÉ?&XH?`?V?(¨ ¨ô?V?`?V?¸Ž?ýÉ?&XH?`?V?¸Ž°Ž8?2??2?'

However recording does not work.
Here is the log after that patch:

 == Using SIP RTP TOS bits 136
 == Using SIP RTP CoS mark 4
 == Using SIP VRTP TOS bits 136
 == Using SIP VRTP CoS mark 4
 == Using UDPTL TOS bits 136
 == Using UDPTL CoS mark 4
   -- Executing [25@CALLEX:1] GotoIf("SIP/test003-00000005", "0?3") in new stack
   -- Executing [25@CALLEX:2] Set("SIP/test003-00000005", "__ORGDEST=25") in new stack
   -- Executing [25@CALLEX:3] AGI("SIP/test003-00000005", "agi://127.0.0.1/script.agi") in new stack
   -- AGI Script Executing Application: (Set) Options: (CHANNEL(language)=pl)
   -- AGI Script Executing Application: (Set) Options: (CALLERID(all)="test003"<+48257405100>)
   -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=ALLOWED_PASSED_SCREEN)
   -- AGI Script Executing Application: (Set) Options: (CDR(is_inbound)=true)
   -- AGI Script Executing Application: (Set) Options: (CDR(is_outbound)=true)
   -- AGI Script Executing Application: (Set) Options: (CDR(number_a)=+48257405100)
   -- AGI Script Executing Application: (Set) Options: (CDR(number_b)=-25)
   -- AGI Script Executing Application: (Set) Options: (CDR(src_interface)=CALLEX/test003)
   -- AGI Script Executing Application: (Set) Options: (CDR(dst_interface)=CALLEX/CONFERENCE)
   -- AGI Script Executing Application: (Set) Options: (CDR(src_account)=test003)
   -- AGI Script Executing Application: (Set) Options: (CDR(dst_account)=sadas)
   -- AGI Script Executing Application: (Meetme) Options: (1,rscM)
 == Parsing '/etc/asterisk/meetme.conf':   == Found
   -- Created MeetMe conference 1023 for conference '1'
   -- <SIP/test003-00000005> Playing 'conf-onlyperson.alaw' (language 'pl')
[2010-10-22 16:45:43] WARNING[10250]: file.c:1165 ast_writefile: No such format ''
   -- Started music on hold, class 'default', on SIP/test003-00000005
   -- Stopped music on hold on SIP/test003-00000005
By: Maciej Krajewski (jamicque) 2011-01-24 19:02:17.000-0600

in 1.6.2.16.1 problem still exists
By: Dmitry Andrianov (dimas) 2011-01-25 02:28:57.000-0600

check the patch from https://issues.asterisk.org/view.php?id=18182
By: Maciej Krajewski (jamicque) 2011-01-25 05:41:28.000-0600

didn't help
By: Leif Madsen (lmadsen) 2011-05-11 13:54:34

Can you reproduce this on 1.8?

~~~~~

Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch.

For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
By: Leif Madsen (lmadsen) 2011-05-30 07:58:44

Please use Asterisk 1.8 for further support.