[asterisk-users] voicemail with odbc
Hi,
I have some issue with voice mail with ODBC on asterisk 11.7 box. I may not
understand database functionality on asterisk fully. The most suspected
area is func_odbc. I already googled but not luck. Your guide is warmly
welcomed
*Error messages when I make call and leave message.*
-- Playing 'auth-thankyou.g722' (language 'en')
[2014-05-28 14:55:13] DEBUG[12260][C-0006]: app_voicemail.c:3824
last_message_index: Directory
'/var/spool/asterisk/voicemail/default/701/INBOX' has no messages and
therefore no index was retrieved.
== Parsing '/var/spool/asterisk/voicemail/default/701/INBOX/msg.txt':
Found
[2014-05-28 14:55:13] WARNING[12260][C-0006]: app_voicemail.c:4086
insert_data_cb: SQL Direct Execute failed!
[2014-05-28 14:55:13] WARNING[12260][C-0006]: res_odbc.c:608
ast_odbc_direct_execute: SQL Execute error! Verifying connection to
asterisk [asterisk-connector]...
[2014-05-28 14:55:13] WARNING[12260][C-0006]: app_voicemail.c:4086
insert_data_cb: SQL Direct Execute failed!
[2014-05-28 14:55:13] WARNING[12260][C-0006]: app_voicemail.c:4202
store_file: SQL Execute error!
[INSERT INTO voicemessages
(dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext,flag,msg_id)
VALUES (?,?,?,?,?,?,?,?,?,?,?,?)]
== Parsing '/var/spool/asterisk/voicemail/default/701/INBOX/msg.txt':
Found
== Parsing '/var/spool/asterisk/voicemail/default/701/INBOX/msg.txt':
Found
-- Auto fallthrough, channel 'SIP/1ffa9-0007' status is 'NOANSWER'
*Dialplan Configuration*
[internal]
exten => 701,1,Dial(SIP/ffbb,17,tT)
same => n,VoiceMail(${EXTEN}@default,u)
exten => 702,1,Dial(SIP/xlite-1,17,tT)
same => n,VoiceMail(${EXTEN}@default,u)
exten => 703,1,Dial(SIP/ffa9,17,tT)
same => n,VoiceMail(${EXTEN}@default,u)
*Voicemail Configuration*
[general]
format=wav49|wav
attach=yes
maxmsg=999
maxsecs=600!
minsecs= 3
skipms=3000
maxlogins=3
odbcstorage=asterisk
odbctable=voicemessages
emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just
${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?left:forwarded)} a ${VM_DUR}
long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from
${VM_CALLERID}, on ${VM_DATE},\n${IF($["${VM_CIDNUM}" =
"${ORIG_VM_CIDNUM}"]?so:(originally sent by ${ORIG_VM_CALLERID} on
${ORIG_VM_DATE})\nso)} you might want to check it when you get a chance.
Thanks!\n\n\t\t\t\t--Asterisk\n
emaildateformat=%A, %B %d, %Y at %r
pagerdateformat=%A, %B %d, %Y at %r
tz=me ; Timezone from zonemessages below. Irrelevant if
envelope=no.
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HMi
me=Asia/Dubai|'vm-received' Q 'digit/at' H N 'hours'
[default]
;1234 => 4242,Example Mailbox,root@localhost
701 => -7012,User One,pronek...@gmail.com
702 => -7023,Soft Phone,sp@localhost
703 => -7034,Pro Nek,pro...@gmail.com
*res_odbc Configuration*
[asterisk]
enabled => yes
dsn => asterisk-connector
username => thet
password => MyPassword
pooling => 1
limit => 5
pre-connect => yes
*func_odbc Configuration*
[SQL]
dsn=mysql1,asterisk
readsql=${ARG1}
; ODBC_ANTIGF - A blacklist.
[ANTIGF]
dsn=mysql1,mysql2 ; Use mysql1 as the primary handle, but fall back to
mysql2
; if mysql1 is down. Supports up to 5 comma-separated
; DSNs. "dsn" may also be specified as "readhandle" and
; "writehandle", if it is important to separate reads
and
; writes to different databases.
readsql=SELECT COUNT(*) FROM exgirlfriends WHERE
callerid='${SQL_ESC(${ARG1})}'
syntax=
synopsis=Check if a specified callerid is contained in the ex-gf database
; ODBC_PRESENCE - Retrieve and update presence
[PRESENCE]
dsn=mysql1
readsql=SELECT location FROM presence WHERE id='${SQL_ESC(${ARG1})}'
writesql=UPDATE presence SET location='${SQL_ESC(${VAL1})}' WHERE
id='${SQL_ESC(${ARG1})}'
*voicemail show command*
abox*CLI> voicemail show users
You must specify a specific context to show users from realtime!
Usage: voicemail show users [for ]
Lists all mailboxes currently set up
abox*CLI>
*extconfig Configuration file*
voicemail =>mysql,asterisk,voicemessages
I create table voicemessages in mysql exactly as description in Definitive
Guide 4th edition book.
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Re: [asterisk-users] 'restart when convenient'
Hi Matthew, Em 28/05/2014 15:09, "Matthew Jordan" escreveu: > * now - tell all CDRs to go submit themselves. Tell all channels to Where CDR is quoted is it also valid for CEL? Tks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail with odbc
On Thu, May 29, 2014 at 3:33 AM, ProNek wrote: > Hi, > > I have some issue with voice mail with ODBC on asterisk 11.7 box. I may not > understand database functionality on asterisk fully. The most suspected area > is func_odbc. I already googled but not luck. Your guide is warmly welcomed > You already started another mailing list thread on this topic a few hours before this. Please don't do that in the future. If you are going to post again, just post to the thread you already started instead of starting a new one. Did you double-check your database table carefully against the required schema? https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] maxsecs not working
On Thu, May 22, 2014 at 6:22 PM, David Cunningham wrote: > Hello, > > We have servers running Asterisk 1.8.20.1 and 11.7.0, and on both setting > maxsecs in voicemail.conf doesn't seem to have any effect. A voicemail keeps > recording after the specified time, and when the caller hangs up the > voicemail is saved in the mailbox. > > Are we doing something really silly? Nope that configuration looks fine and it works on my systems as expected in the latest of those branches. Using your configuration I tried changing the maxsecs value and it appears to be respected. If you can reproduce the issue and provide debug to demonstrate, then you might file a bug report. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.28.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.28.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.28.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: --- * ASTERISK-23547 - [patch] app_queue removing callers from queue when reloading (Reported by Italo Rossi) * ASTERISK-22846 - testsuite: masquerade super test fails on all branches (still) (Reported by Matt Jordan) * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think (Reported by Walter Doekes) * ASTERISK-23620 - Code path in app_stack fails to unlock list (Reported by Bradley Watkins) * ASTERISK-18331 - app_sms failure (Reported by David Woodhouse) * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by Krzysztof Chmielewski) * ASTERISK-23707 - Realtime Contacts: Apparent mismatch between PGSQL database state and Asterisk state (Reported by Mark Michelson) * ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+) (Reported by Guillaume Maudoux) * ASTERISK-22977 - chan_sip+CEL: missing ANSWER and PICKUP event for INVITE/w/replaces pickup (Reported by Walter Doekes) * ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone (Reported by Steve Davies) * ASTERISK-23650 - Intermittent segfault in string functions (Reported by Roel van Meer) Improvements made in this release: --- * ASTERISK-23754 - [patch] Use var/lib directory for log file configured in asterisk.conf (Reported by Igor Goncharovsky) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.28.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.10.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.10.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.10.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: --- * ASTERISK-23547 - [patch] app_queue removing callers from queue when reloading (Reported by Italo Rossi) * ASTERISK-23559 - app_voicemail fails to load after fix to dialplan functions (Reported by Corey Farrell) * ASTERISK-22846 - testsuite: masquerade super test fails on all branches (still) (Reported by Matt Jordan) * ASTERISK-23545 - Confbridge talker detection settings configuration load bug (Reported by John Knott) * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think (Reported by Walter Doekes) * ASTERISK-23620 - Code path in app_stack fails to unlock list (Reported by Bradley Watkins) * ASTERISK-23616 - Big memory leak in logger.c (Reported by ibercom) * ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS (Reported by Sebastian Wiedenroth) * ASTERISK-23550 - Newer sound sets don't show up in menuselect (Reported by Rusty Newton) * ASTERISK-18331 - app_sms failure (Reported by David Woodhouse) * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by Krzysztof Chmielewski) * ASTERISK-23605 - res_http_websocket: Race condition in shutting down websocket causes crash (Reported by Matt Jordan) * ASTERISK-23707 - Realtime Contacts: Apparent mismatch between PGSQL database state and Asterisk state (Reported by Mark Michelson) * ASTERISK-23381 - [patch]ChanSpy- Barge only works on the initial 'spy', if the spied-on channel makes a new call, unable to barge. (Reported by Robert Moss) * ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+) (Reported by Guillaume Maudoux) * ASTERISK-23664 - Incorrect H264 specification in SDP. (Reported by Guillaume Maudoux) * ASTERISK-22977 - chan_sip+CEL: missing ANSWER and PICKUP event for INVITE/w/replaces pickup (Reported by Walter Doekes) * ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone (Reported by Steve Davies) Improvements made in this release: --- * ASTERISK-23649 - [patch]Support for DTLS retransmission (Reported by NITESH BANSAL) * ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently available in a CLI command (Reported by Patrick Laimbock) * ASTERISK-23754 - [patch] Use var/lib directory for log file configured in asterisk.conf (Reported by Igor Goncharovsky) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.10.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 12.3.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 12.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Improvements made in this release: --- * ASTERISK-23553 - Add ast_spinlock capability to lock.h (Reported by George Joseph) * ASTERISK-23649 - [patch]Support for DTLS retransmission (Reported by NITESH BANSAL) * ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently available in a CLI command (Reported by Patrick Laimbock) * ASTERISK-23754 - [patch] Use var/lib directory for log file configured in asterisk.conf (Reported by Igor Goncharovsky) Bugs fixed in this release: --- * ASTERISK-23547 - [patch] app_queue removing callers from queue when reloading (Reported by Italo Rossi) * ASTERISK-22846 - testsuite: masquerade super test fails on all branches (still) (Reported by Matt Jordan) * ASTERISK-23390 - NewExten Event with application AGI shows up before and after AGI runs (Reported by Benjamin Keith Ford) * ASTERISK-23584 - PJSIP 'Unable to create channel' when attempting to call from endpoint with UDP transport to one using WebSockets (Reported by Rusty Newton) * ASTERISK-23545 - Confbridge talker detection settings configuration load bug (Reported by John Knott) * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think (Reported by Walter Doekes) * ASTERISK-22904 - bridges: lock the bridge when creating bridge snapshots (Reported by Matt Jordan) * ASTERISK-23620 - Code path in app_stack fails to unlock list (Reported by Bradley Watkins) * ASTERISK-23616 - Big memory leak in logger.c (Reported by ibercom) * ASTERISK-23588 - ARI: Crash when unsubscribing from bridge (Reported by Matt Jordan) * ASTERISK-23502 - Channel variable SIPREFERTOHDR not being set during blind transfer (Reported by John Bigelow) * ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS (Reported by Sebastian Wiedenroth) * ASTERISK-23514 - The pjsip.conf aor qualify contact parameters are not updated on reload. (Reported by Richard Mudgett) * ASTERISK-23550 - Newer sound sets don't show up in menuselect (Reported by Rusty Newton) * ASTERISK-22677 - Playbacks on bridge via ARI are not queued (Reported by John Bigelow) * ASTERISK-18331 - app_sms failure (Reported by David Woodhouse) * ASTERISK-23487 - features.conf cant load from realtime because features_config.c starts before loader.c (Reported by Denis) * ASTERISK-23282 - Documentation - Tab completion and CLI usage documentation do not indicate that 'all' is accepted for 'confbridge kick all' (Reported by Dorian Logan) * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by Krzysztof Chmielewski) * ASTERISK-23573 - Crash when transferring unbridged call - in bridge_app_subscribed at stasis/app.c (Reported by Mark Michelson) * ASTERISK-23639 - PJSIP Realtime: Alembic migration needed in order to widen some string columns (Reported by Mark Michelson) * ASTERISK-23560 - [ARI] MOH doesn't indicate progress (Reported by Jan Svoboda) * ASTERISK-23605 - res_http_websocket: Race condition in shutting down websocket causes crash (Reported by Matt Jordan) * ASTERISK-23498 - Asterisk PJSIP transport configuration fails on parsing of 'cipher' option, any valid option is reported as unsupported (Reported by Anthony Messina) * ASTERISK-23672 - PJSIP Digium presence notifications are not sent if only the subtype or message changes (Reported by Mark Michelson) * ASTERISK-23501 - Copy 'Referred-By' header to outgoing INVITE (Reported by John Bigelow) * ASTERISK-23707 - Realtime Contacts: Apparent mismatch between PGSQL database state and Asterisk state (Reported by Mark Michelson) * ASTERISK-23675 - [patch] Segmentation Fault on first SIP registration using res_config_odbc (Reported by Leandro Dardini) * ASTERISK-23381 - [patch]ChanSpy- Barge only works on the initial 'spy', if the spied-on channel makes a new call, unable to barge. (Reported by Robert Moss) * ASTERISK-23497 - chan_sip SIP protocol attended transfer, with directmedia=yes results in a simple bridge, typically with no audio (Reported by Etienne Lessard) * ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+) (Reported by Guillaume Maudoux) * ASTERISK-23664 - Incorrect H264 specification in SDP. (Reported by Guillaume Maudoux) * ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone (Reported by Steve Davies) * ASTERISK-23758 - 500 internal server error when a
Re: [asterisk-users] Asterisk 11.10.0 Now Available
Asterisk Development Team wrote: > The Asterisk Development Team has announced the release of Asterisk 11.10.0. > This release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk > > The release of Asterisk 11.10.0 resolves several issues reported by the > community and would have not been possible without your participation. ... > * ASTERISK-23754 - [patch] Use var/lib directory for log file > configured in asterisk.conf (Reported by Igor Goncharovsky) Is this mandatory -- what is wrong with /var/log/asterisk for those files? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.10.0 Now Available
- Original Message - > From: cov...@ccs.covici.com > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Thursday, May 29, 2014 6:42:05 PM > Subject: Re: [asterisk-users] Asterisk 11.10.0 Now Available > > > * ASTERISK-23754 - [patch] Use var/lib directory for log file > > configured in asterisk.conf (Reported by Igor Goncharovsky) > Is this mandatory -- what is wrong with /var/log/asterisk for those > files? > The title on that issue is very misleading. The patch that went in was just for chan_ooh323. The change was to have chan_ooh323 use the log directory configured in asterisk.conf instead of using a hard coded value. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.10.0 Now Available
Michael L. Young wrote: > - Original Message - > > From: cov...@ccs.covici.com > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > Sent: Thursday, May 29, 2014 6:42:05 PM > > Subject: Re: [asterisk-users] Asterisk 11.10.0 Now Available > > > > > * ASTERISK-23754 - [patch] Use var/lib directory for log file > > > configured in asterisk.conf (Reported by Igor Goncharovsky) > > Is this mandatory -- what is wrong with /var/log/asterisk for those > > files? > > > > The title on that issue is very misleading. The patch that went in was just > for chan_ooh323. The change was to have chan_ooh323 use the log directory > configured in asterisk.conf instead of using a hard coded value. OK, thanks, boy that title is sure misleading! -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk mixmonitor with 16khz
Hi I have a transcription software which requires media files with wide band frequencies. Is it possible asterisk can record calls with 16khz not 8khz ? Best regards Muhammad Yousuf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
