[asterisk-bugs] [Asterisk 0011765]: Fake ring tone
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed Jan 16 12:08:17 CST 2008
The following issue has been RESOLVED.
======================================================================
http://bugs.digium.com/view.php?id=11765
======================================================================
Reported By: balgaa
Assigned To: russell
======================================================================
Project: Asterisk
Issue ID: 11765
Category: Addons/chan_ooh323
Reproducibility: always
Severity: major
Priority: normal
Status: resolved
Asterisk Version: 1.4.17
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
Resolution: suspended
Fixed in Version:
======================================================================
Date Submitted: 01-14-2008 11:12 CST
Last Modified: 01-16-2008 12:08 CST
======================================================================
Summary: Fake ring tone
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 at 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 at 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 at 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 at 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.
======================================================================
----------------------------------------------------------------------
russell - 01-16-08 12:08
----------------------------------------------------------------------
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.
Issue History
Date Modified Username Field Change
======================================================================
01-16-08 12:08 russell Status new => resolved
01-16-08 12:08 russell Resolution open => suspended
01-16-08 12:08 russell Assigned To => russell
01-16-08 12:08 russell Note Added: 0080778
======================================================================
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