[asterisk-bugs] [Asterisk 0010824]: Poor audio quality and sometimes disconnection after hangup

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10824 
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Reported By:                swatchy
Assigned To:                mnicholson
====================================================================== 
Project:                    Asterisk
Issue ID:                   10824
Category:                   Addons/chan_mobile
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.6.1-beta3 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 83834 
Request Review:              
====================================================================== 
Date Submitted:             2007-09-26 03:18 CDT
Last Modified:              2009-03-30 05:06 CDT
====================================================================== 
Summary:                    Poor audio quality and sometimes disconnection after
hangup
Description: 
Some information about my system:
OS Ubuntu 7.04
Bluez: 3.9-0ubuntu1
Bluetooth: Internal Bluetooth from Acer C110 Notebook
Asterisk: SVN-trunk-r83834
chan_mobile: Revision 451
Cellphone: Nokia 6230i

These things are working:
1. Notebook and Nokia are paired
2. Outgoing call from iax phone -> asterisk -> pstn works
3. Incoming call not tested
4. Incoming/outgoing SMS are working

The problem:
At the time the pstn phone picks up the call from the cellphone, you can
hear the person sitting behind the iax softphone very quiet and with a
really poor audio quality. It sounds like a robot is speaking to you. 

My tries to solve the problem:
In mobile.conf:
forcemaster set to no and yes
alignmentdetection set to no and yes
Set the iax softphone to gsm and ulaw

Another problem:
Sometimes after hangup the call the connection between asterisk and the
cellphone breaks up. I don't know why? See this for more information:






====================================================================== 

---------------------------------------------------------------------- 
 (0102369) nikkk (reporter) - 2009-03-30 05:06
 http://bugs.digium.com/view.php?id=10824#c102369 
---------------------------------------------------------------------- 
Test System:

Fedora 10 Kernel 2.6.27.19-170.2.35
Bluez 4.30
Asterisk SVN-branch-1.6.1-r184345

Conceptronic usb 2.0 CBTU2A, Samsung GT-C6620

Logs:

--- DIAL PHASE ---


[Mar 30 10:50:22] DEBUG[26641]: pbx.c:3174 pbx_extension_helper: Launching
'Dial'
    -- Executing [903498727189 at default:1] Dial("SIP/204-09beb688",
"Mobile/GT-C6620/3498727189,50,Ttr") in new stack
[Mar 30 10:50:22] DEBUG[26641]: rtp.c:2129 ast_rtp_make_compatible:
Channel 'Mobile/GT-C6620-3925' has no RTP, not doing anything
[Mar 30 10:50:22] DEBUG[26641]: channel.c:3975
ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Mar 30 10:50:22] DEBUG[26641]: channel.c:3975
ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Mar 30 10:50:22] DEBUG[26641]: channel.c:3975
ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Mar 30 10:50:22] DEBUG[26641]: channel.c:3975
ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Mar 30 10:50:22] DEBUG[26641]: channel.c:3975
ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Mar 30 10:50:22] DEBUG[26641]: channel.c:3975
ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Mar 30 10:50:22] DEBUG[26641]: channel.c:3975
ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
localhost*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk ending (0).

Dial from cellphone proceed as normal ... asterisk dies

dmesg:

asterisk[26641]: segfault at c ip 001e963d sp b752a360 error 6 in
chan_mobile.so[1df000+11000]




--- HANGUP PHASE ---


[Mar 30 10:54:20] DEBUG[26686]: chan_mobile.c:1647 sco_write: sco_write()
[Mar 30 10:54:20] DEBUG[26686]: chan_mobile.c:1647 sco_write: sco_write()
[Mar 30 10:54:20] DEBUG[26676]: chan_sip.c:19557 handle_incoming: ****
Received BYE (8) - Command in SIP BYE
[Mar 30 10:54:20] DEBUG[26676]: chan_sip.c:2643 sip_alreadygone: Setting
SIP_ALREADYGONE on dialog d32ca005-4e0a2098 at localhost
[Mar 30 10:54:20] DEBUG[26676]: chan_sip.c:19058 handle_request_bye:
Received bye, issuing owner hangup
[Mar 30 10:54:20] DEBUG[26676]: chan_sip.c:2868 __sip_xmit: Trying to put
'SIP/2.0 20' onto UDP socket destined for 192.168.3.104:5060
[Mar 30 10:54:20] DEBUG[26686]: channel.c:4482 ast_generic_bridge: Didn't
get a frame from channel: SIP/204-0943dc88
[Mar 30 10:54:20] DEBUG[26686]: channel.c:4903 ast_channel_bridge: Bridge
stops bridging channels SIP/204-0943dc88 and Mobile/GT-C6620-f54d
[Mar 30 10:54:20] DEBUG[26686]: channel.c:1641 ast_hangup: Hanging up
channel 'Mobile/GT-C6620-f54d'
[Mar 30 10:54:20] DEBUG[26686]: chan_mobile.c:952 mbl_hangup: [GT-C6620]
hanging up device
[Mar 30 10:54:20] DEBUG[26686]: chan_mobile.c:960 mbl_hangup: Closing SCO
socket
[Mar 30 10:54:20] DEBUG[26686]: chan_mobile.c:966 mbl_hangup: Sending
HANGUP
[Mar 30 10:54:20] DEBUG[26686]: chan_mobile.c:1338 rfcomm_write_full:
rfcomm_write() (21) [AT+CHUP
localhost*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk ending (0).


Hangup on cellphone is done correctly ... asterisk dies

dmesg:

asterisk[26686]: segfault at c ip 00239031 sp b75ee3a0 error 6 in
chan_mobile.so[22f000+11000] 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-30 05:06 nikkk          Note Added: 0102369                          
======================================================================