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Dahdi 2.6.1 with OSLEC support

In order solve my incoming caller ID problem, I upgrade the dahdi to
version 2.6.1 from version 2.4.x. After upgrade, I found the echo
cancellation doesn’t working (I’m using Digium AEX800B PCI Express card). I
can hear my self talking on the phone. How to solve this? I think I need to
recompile dahdi 2.6.1 with OSLEC support? how?

[root@callcenter ~]# dahdi_cfg -vvv
DAHDI Tools Version – 2.6.1

DAHDI Version: 2.6.1
Echo Canceller(s): HWEC
Configuration
======================

Channel map:

Channel 01: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 04)
Channel 05: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 05)
Channel 06: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 06)
Channel 07: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 07)
Channel 08: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 08)
Channel 09: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 09)
Channel 10: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 10)
Channel 11: FXS Kewlstart (Default) (Echo Canceler: oslec)(Slaves: 11)
Channel 12: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 12)
Channel 13: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 13)
Channel 14: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 14)
Channel 15: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 15)
Channel 16: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 16)

16 channels to configure.

Setting echocan for channel 1 to oslec
DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22)

I try to change to /etc/dahdi/system.conf to
fxsks=1
echocanceller=hwec,1

Still doesn’t work and this error still occured: DAHDI_ATTACH_ECHOCAN
failed on channel 1: Invalid argument (22)

[root@callcenter dahdi]# asterisk -rvvv
Asterisk 1.8.7.0, Copyright (C) 1999 – 2011 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for
detail
s.
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.
=========================================================================
== Parsing ‘/etc/asterisk/asterisk.conf’: == Found
== Parsing ‘/etc/asterisk/extconfig.conf’: == Found
Connected to Asterisk 1.8.7.0 currently running on callcenter (pid = 6908)
Verbosity is at least 3
callcenter*CLI> dahdi show channel 1
Channel: 1
File Descriptor: 12
Span: 1
Extension:
Dialing: no
Context: from-pstn
Caller ID:
Calling TON: 0
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner:
Real:

Callwait:

Threeway:

Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: no
TDD: no
Relax DTMF: yes
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
*Echo Cancellation:
128 taps
(unless TDM bridged) currently OFF*
Wait for dialtone: 0ms
Actual Confinfo: Num/0, Mode/0×0000
Actual Confmute: No
Hookstate (FXS only): Offhook
callcenter*CLI>

Disable All Asterisk Features (blind xfer, disconnect, etc)

Hi Guys,

is there any way to disable all Asterisk Features? We are having false dtmf
detections and randon calls being put on-hold and suspect that dtmf
features is the cause.

Changing features.conf aparently keeps the default options. Since we dont
use it, is there any way to disable it?

Thanks,

Eduardo

Asterisk 1.6.2 > 1.8.12

Jonas Kellens wrote:
> Will ODBC become the default then ?

As far as I’m aware, yes.

You need to make sure that unix-odbc development libraries are installed.

Under Mandriva, Mageia:

urpmi unixodbc-devel

Ubuntu/Debian:

apt-get install unixodbc-dev

And here is a good link to get you started:

http://nerdvittles.com/?p=604

Doug

Problem with blank/empty voicemails

Hi,
I hope for a hint on this issue.

I had a voicemail running on ast release 1.6.2 latest which i upgraded
to 1.8.11 latest release.
during this process I did add a couple of fields like minsecs and maxsecs.

I do now get empty emails where the attached voicefile only contains the
voice header,
the message length written in the email is ok.
If I go to the voicemailbox during the recording then I can se the files
grow to the filesize i would expect, looks like everything is ok until then.

When I press the ‘#’ or hangup then the email is generated with the
empty attachment and the voicefiles in the INBOX is now
44 bytes for .wav and
60 byes for .WAV

I have voicemail in mysql and messages stored i filesystem
here is some of the config (which worked ok on release 1.6.2)
attachfmt=wav
deletevoicemail=no
volgain=0.0
minsecs=2
maxsecs=600

[Apr 23 21:59:46] DEBUG[31841] app.c: Locked path
‘/var/spool/asterisk/voicemail/default/88855/INBOX’
[Apr 23 21:59:46] DEBUG[31841] app.c: Unlocked path
‘/var/spool/asterisk/voicemail/default/88855/INBOX’
[Apr 23 21:59:46] DEBUG[31841] channel.c: Scheduling timer at (50
requested / 50 actual) timer ticks per second
[Apr 23 21:59:46] VERBOSE[31841] file.c: –
Playing ‘beep.alaw’ (language ‘dk’)
[Apr 23 21:59:46] DEBUG[31841] channel.c: Scheduling timer at (182
requested / 182 actual) timer ticks per second
[Apr 23 21:59:46] DEBUG[31841] channel.c: Scheduling timer at (0
requested / 0 actual) timer ticks per second
[Apr 23 21:59:46] DEBUG[31841] channel.c: Scheduling timer at (0
requested / 0 actual) timer ticks per second
[Apr 23 21:59:46] DEBUG[31841] channel.c: Scheduling timer at (0
requested / 0 actual) timer ticks per second
[Apr 23 21:59:46] VERBOSE[31841] app_voicemail.c: — Recording the
message
[Apr 23 21:59:46] DEBUG[31841] app.c: play_and_record: ,
/var/spool/asterisk/voicemail/default/88855/tmp/N0NNlv, ‘wav49|gsm|wav’
[Apr 23 21:59:46] DEBUG[31841] app.c: Recording Formats: sfmts=wav49
[Apr 23 21:59:46] VERBOSE[31841] app.c: — x=0, open writing:
/var/spool/asterisk/voicemail/default/88855/tmp/N0NNlv format: wav49,
0xb3501378
[Apr 23 21:59:46] VERBOSE[31841] app.c: — x=1, open writing:
/var/spool/asterisk/voicemail/default/88855/tmp/N0NNlv format: gsm,
0xb266b1b0
[Apr 23 21:59:46] VERBOSE[31841] app.c: — x=2, open writing:
/var/spool/asterisk/voicemail/default/88855/tmp/N0NNlv format: wav,
0xb3519948
[Apr 23 21:59:46] DEBUG[31841] dsp.c: Setup tone 1100 Hz, 500 ms,
block_size=160, hits_required=21
[Apr 23 21:59:46] DEBUG[31841] dsp.c: Setup tone 2100 Hz, 2600 ms,
block_size=160, hits_required=116
[Apr 23 21:59:46] DEBUG[31841] channel.c: Set channel
SIP/_Mw_cHFm6vZAyV-00012a5c to read format slin
[Apr 23 22:00:03] VERBOSE[31841] app.c: — User hung up
[Apr 23 22:00:03] DEBUG[31841] channel.c: Set channel
SIP/_Mw_cHFm6vZAyV-00012a5c to read format alaw
[Apr 23 22:00:03] DEBUG[31841] app.c: Locked path
‘/var/spool/asterisk/voicemail/default/88855/INBOX’
[Apr 23 22:00:03] DEBUG[31841] app.c: Unlocked path
‘/var/spool/asterisk/voicemail/default/88855/INBOX’
[Apr 23 22:00:03] DEBUG[31841] app_voicemail.c: Attaching file
‘/var/spool/asterisk/voicemail/default/88855/INBOX/msg0000′, format
‘wav’, uservm is ’2048′, global is 2048
[Apr 23 22:00:03] VERBOSE[31841] config.c: == Parsing
‘/var/spool/asterisk/voicemail/default/88855/INBOX/msg0000.txt’: [Apr 23
22:00:03] DEBUG[31841] config.c: Parsing
/var/spool/asterisk/voicemail/default/88855/INBOX/msg0000.txt
[Apr 23 22:00:03] VERBOSE[31841] config.c: == Found
[Apr 23 22:00:03] VERBOSE[31841] config.c: == Parsing
‘/var/spool/asterisk/voicemail/default/88855/INBOX/msg0000.txt’: [Apr 23
22:00:03] DEBUG[31841] config.c: Parsing
/var/spool/asterisk/voicemail/default/88855/INBOX/msg0000.txt
[Apr 23 22:00:03] VERBOSE[31841] config.c: == Found
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Evaluating ‘VM_NAME’ (from ‘VM_NAME}:
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Result of ‘VM_NAME’ is ‘Freddi Hansen’
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Evaluating ‘VM_DUR’ (from ‘VM_DUR}
lang besked (number ${VM_MSGNUM})
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Result of ‘VM_DUR’ is ’0:16′
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Evaluating ‘VM_MSGNUM’ (from
‘VM_MSGNUM})
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Result of ‘VM_MSGNUM’ is ’1′
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Evaluating ‘VM_MAILBOX’ (from
‘VM_MAILBOX} fra ${VM_CALLERID}, den ${VM_DATE}. Tak!
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Result of ‘VM_MAILBOX’ is ’88855′
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Evaluating ‘VM_CALLERID’ (from
‘VM_CALLERID}, den ${VM_DATE}. Tak!
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Result of ‘VM_CALLERID’ is
‘”voicemailtest” <_mw_chfm6vzayv>‘
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Evaluating ‘VM_DATE’ (from
‘VM_DATE}. Tak!
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Result of ‘VM_DATE’ is ‘Monday, 23
April 2012 at 22:00:03′
[Apr 23 22:00:03] DEBUG[31841] app_voicemail.c: creating attachment
filename msg0000.wav, no second attachment.
[Apr 23 22:00:03] DEBUG[31841] app_voicemail.c: Sent mail to
fh@danovation.dk with command ‘/usr/sbin/sendmail -t’
[Apr 23 22:00:03] DEBUG[31841] pbx.c: Spawn extension
(voicemailtest,222,2) exited non-zero on ‘SIP/_Mw_cHFm6vZAyV-00012a5c’

HELP!! Caller ID “unknown” for all inbound call

This is a very strange problem (at least for me). I just realized that
started from April 20th 2012 every inbound call is from “unknown”.
Prior that, asterisk succesfully displayed the caller caller’s ID for SOME
of the calls (30-50% success rate). I am using PBX | monitoring menu to see
this report.

As far as I remember, I dont modify any settings that related to caller ID,
but few days ago (I dont remember the exact date), I modify the rxgain and
txgain value in chan_dahdi.conf.
The inbound caller ID doesn’t display on the log and on the phone.

I’m running asterisk 1.8.7.0
FreePBX 2.8.1
FXO card using Digium AEX800B

The caller ID is display well on the phone If I connect the phone directly
without connecting to asterisk (just for testing purpose)

This is my chan_dahdi.conf

[trunkgroups]

[channels]
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=8.0
txgain=8.0
callgroup=1
pickupgroup=1

;Uncomment these lines if you have problems with the disconection of your
analog lines
;busydetect=yes
;busycount=3

immediate=no

#include dahdi-channels.conf
#include chan_dahdi_additional.conf
chan_dahdi.conf (END)

This is my dahdi-channels.conf
; Autogenerated by /usr/sbin/dahdi_genconf on Fri Mar 30 22:32:16 2012
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is
intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global
settings
;

; Span 1: WCTDM/0 “Wildcard AEX800 Board 1″ (MASTER)
;;; line=”1 WCTDM/0/0 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default

;;; line=”2 WCTDM/0/1 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 2
callerid=
group=
context=default

;;; line=”3 WCTDM/0/2 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 3
callerid=
group=
context=default

;;; line=”4 WCTDM/0/3 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 4
callerid=
group=
context=default

;;; line=”5 WCTDM/0/4 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 5
callerid=
group=
context=default

;;; line=”6 WCTDM/0/5 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 6
callerid=
group=
context=default

Any help would be very appeciated.

Thanks and best regards,
Anam.

Asterisk Insecure

This setting specifies how to handle connections with peers. By default Asterisk will authenticate all connections (this is the same as insecure=no). The parameters invite and port were added in v1.2.x  (and are to be used when you trust the IP of the caller), yes and very were removed in v1.6.x.

Note: yes/very are deprecated in 1.8