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Summary:ASTERISK-11234: Fake ring tone
Reporter:Balgansuren Batsukh (balgaa)Labels:
Date Opened:2008-01-14 11:12:14.000-0600Date Closed:2011-06-07 14:02:59
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Addons/chan_ooh323
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) chan_h323_debug_trace.txt
Description:I reinstalled Asterisk-1.4.17 without codec negotiation patch and now from analog phone/sip phone Asterisk never crash.

But there also interesting thing happen, i can hear ringing tone on analog phone. If I call from my H323 phone to same number there voice response saying "Your called person mobile phone is off".

I saw when Asterisk get "Alerting" message from GnuGK it give sip/analog phone ringing tone. Same as before I can't hear real ringing tone from GnuGK.

Just I tried asterisk-addons ooh323 driver and after install this driver I can to hear inband ring tone from PSTN, Mobile operator switch.

If debug on ooh323:
-------------------
pbx:/home/balgaa# asterisk -r
Asterisk 1.4.17, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.17 currently running on pbx (pid = 26600)
Verbosity is at least 3
   -- Registered SIP '1100008' at 203.91.112.113 port 45900 expires 1800
   -- Executing [00197699014447@default:1] Dial("SIP/1100008-0821a560", "H323/00197699014447") in new stack
[Jan 17 01:01:58] WARNING[26674]: channel.c:3281 ast_request: No channel type registered for 'H323'
[Jan 17 01:01:58] WARNING[26674]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'H323' (cause 66 - Channel not implemented)
 == Everyone is busy/congested at this time (1:0/0/1)
 == Auto fallthrough, channel 'SIP/1100008-0821a560' status is 'CHANUNAVAIL'
   -- Executing [00597699014447@default:1] Dial("SIP/1100008-08218bc8", "OOH323/00597699014447") in new stack
---   ooh323_request - data 00597699014447 format 0x100 (g729)
---   find_peer "00597699014447"
+++   find_peer "00597699014447"
+++   ooh323_request
---   ooh323_call- 00597699014447
---   onNewCallCreated ooh323c_o_11
---   find_call
+++   find_call
+++   ooh323_call
   -- Called 00597699014447
setting callid number 1100008
Outgoing call 00597699014447(ooh323c_o_11) - Codec prefs - (g729|g723)
       Adding capabilities to call(outgoing, ooh323c_o_11)
       Adding g729A capability to call(outgoing, ooh323c_o_11)
       Adding g729 capability to call(outgoing, ooh323c_o_11)
       Adding g7231 capability to call (outgoing, ooh323c_o_11)
---   configure_local_rtp
+++   configure_local_rtp
+++   onNewCallCreated ooh323c_o_11
---   setup_rtp_connection
---   find_call
+++   find_call
+++   setup_rtp_connection
--- onAlerting ooh323c_o_11
---   find_call
+++   find_call
+++ onAlerting ooh323c_o_11
   -- OOH323/00597699014447-5079 is ringing
[Jan 17 01:02:09] NOTICE[26676]: rtp.c:787 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 202.72.242.22
---   ooh323_hangup
   hanging 00597699014447
+++   ooh323_hangup
 == Spawn extension (default, 00597699014447, 1) exited non-zero on 'SIP/1100008-08218bc8'
---   close_rtp_connection
---   find_call
+++   find_call
+++   close_rtp_connection
---   onCallCleared ooh323c_o_11
---   find_call
+++   find_call
+++   onCallCleared
---   ooh323_destroy
Destroying 00597699014447
+++   ooh323_destroy
pbx*CLI>

If I debug on chan_h323:
------------------------
pbx:/home/balgaa# asterisk -r
Asterisk 1.4.17, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.17 currently running on pbx (pid = 26709)
Verbosity is at least 3
pbx*CLI> show channeltypes
Type        Description                              Devicestate  Indications  Transfer
----------  -----------                              -----------  -----------  --------
Skinny      Skinny Client Control Protocol (Skinny)  no           yes          no
MGCP        Media Gateway Control Protocol (MGCP)    yes          yes          no
Zap         Zapata Telephony Driver w/PRI            no           yes          no
Console     OSS Console Channel Driver               no           yes          no
IAX2        Inter Asterisk eXchange Driver (Ver 2)   yes          yes          yes
H323        The NuFone Network's Open H.323 Channel  no           yes          no
Gtalk       Gtalk Channel Driver                     no           yes          no
Local       Local Proxy Channel Driver               yes          yes          no
Feature     Feature Proxy Channel Driver             no           yes          no
Agent       Call Agent Proxy Channel                 yes          yes          no
Phone       Standard Linux Telephony API Driver      no           yes          no
SIP         Session Initiation Protocol (SIP)        yes          yes          yes
----------
12 channel drivers registered.
The 'show channeltypes' command is deprecated and will be removed in a future release. Please use 'core show channeltypes' instead.
pbx*CLI> h323 set debug
H.323 debug enabled
pbx*CLI> h323 set trace 10
H.323 trace set to level trace

Please find attached debug and trace information.
Comments:By: pj (pj) 2008-01-14 11:21:00.000-0600

I have similar issue with inband messages and chan_h323, but I'm using asterisk trunk, fyi, bugreport 10497
By: Balgansuren Batsukh (balgaa) 2008-01-14 11:43:55.000-0600

PJ, are you using 1.4 trunk? Is it working fine with inband?
By: Balgansuren Batsukh (balgaa) 2008-01-14 11:52:44.000-0600

I read your report. Is latest trunk solved this issue?

Which is version trunk using now?
By: Jason Parker (jparker) 2008-01-14 12:48:55.000-0600

pj, would you agree that this should be closed as a duplicate of 10497?
By: pj (pj) 2008-01-14 14:09:19.000-0600

balgaa, I'm still waiting to my issue will be resolved, currently it's still broken in latest trunk, but if you would like to test, last inband messages working for me is trunk revision 78682, so you can checkout, compile, and try...
"svn -r 78682 checkout http://svn.digium.com/svn/asterisk/trunk asterisk-r78682"


By: pj (pj) 2008-01-14 14:17:09.000-0600

Jason, it seems, that issue is similar (not working inband messages using chan_h323), but my issue, that seems was introduced in trunk commit r78683, probably affect only asterisk trunk, it was probably not commited to 1.4branch.
also my environment is quite different:
sip-->asterisk/chan_h323-->callmanager--(h323)-->ciscogw--(isdn/pri)-->pstn
so, I think, you should keep this issue open.
By: Russell Bryant (russell) 2008-01-16 12:08:16.000-0600

This module has been unsupported for a long time.  Now, it is officially marked as unsupported.  So, only bug reports with patches will be accepted at this time.