[asterisk-users] ODBC Voicemail Storage

2006-11-15 Thread Edwin Horton
I current have a working Asterisk 1.2.12 server with ODBC voicemail storage,
realtime static maps for voicemail, sip and iax configuration files.
Realtime extensions, etc.  All works great.  I have verified that this
configuration works on my test server as well.  Now I am trying to test the
1.4B3 version on the same test server, and all works well except for ODBC
voicemail.  I am using the same table structure as before (extended ODBC),
and the ODBC system works well in that I can use it for the static maps
(extconfig.conf), or mysql native from the addons package.  With Asterisk
compiled without ODBC voicemail, it works flawless.  Anyway, Asterisk with
ODBC voicemail compile option will not start with the following console
message:

  == Parsing '/etc/asterisk/voicemail.conf': Found
[Nov 15 09:18:47] DEBUG[23599]: app_voicemail.c:7056 load_config: VM
Temperary Greeting Reminder Option disabled globally
[Nov 15 09:18:47] DEBUG[23599]: app_voicemail.c:7082 load_config: ENVELOPE
before msg enabled globally
[Nov 15 09:18:47] DEBUG[23599]: app_voicemail.c:7110 load_config: found
dialout context: fromvm
[Nov 15 09:18:47] DEBUG[23599]: app_voicemail.c:7117 load_config: found
callback context: fromvm
  == Parsing '/etc/asterisk/users.conf': Found
app_voicemail.so = (Comedian Mail (Voicemail System) with ODBC Storage)
  == Registered channel type 'Local' (Local Proxy Channel Driver)
chan_local.so = (Local Proxy Channel)
  == Parsing '/etc/asterisk/codecs.conf': Found
-- codec_gsm: using generic PLC
  == Registered translator 'gsmtolin' from format gsm to slin, cost 5
asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_voicemail.so:
undefined symbol: odbc_request_obj

I get no other information in the debug or message files.  An attempt to
backtrace, does not yield a crash dump regardless of the compile options.
Does anyone have any ideas?
Ed Horton
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] call parking and realtime_ext

2005-11-19 Thread Edwin Horton
I am using realtime_ext, asterisk-1.2.0 and am trying to understand the
correct method of adding extensions in my database to correctly handing call
parking.  I have it working fairly well by adding an extension of 700 in the
correct context and then extensions 700-7xx with the ParkedCall application.
All works well unless the call is not picked up and it returns to the
extension that parked the call.  If this extension does not answer, I get a
congested message and the following error:

WARNING[]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context
'park-dial'

I would like to add a handler for this case in thr realtime list.  Any
ideas?

Ed Horton

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Voicemail ODBC storage and realtime

2005-11-18 Thread Edwin Horton
Using Asterisk-1.2.0, I have voicemail messages stored in a MySQL database
with ODBC.  I am using a database to store certain config files, such as
sip.conf, via Realtime Static.  Since you must define the variable
odbcstorage in the voicemail.conf file to allow ODBC storage to work, what
do you do if you want to store the voicemail accounts via realtime (not
static) and have no voicemail.conf file.l.  Do you still use the
voicemail.conf file and just define all of the mailboxes via Realtime?  I
tried this, but for some reason, I could not get asterisk to query the mysql
database for the mailbox info.  If I defined the mailbox in the
voicemail.conf as usual, all was OK

Thanks..

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Realtime voicemail

2005-04-28 Thread Edwin Horton
Thank you both for the insight.  The original problem was that the voice
mail system returned a no mailbox found error since the query was looking
for a mailbox in the default context and I had defined them in other
contexts, in my case, from-sip and analog-phones.  It seems I am
confusing extension context with voicemail context.  I included the
following in my extension file:

exten = 2201,1,agi,notify.agi
exten = 2201,2,Dial(Zap/9,20)
exten = 2201,3,Answer
exten = 2201,4,Wait(1)
exten = 2201,5,Voicemail(u${EXTEN})
exten = 2201,6,Hangup
exten = 2201,105,Voicemail(b${EXTEN})
exten = 2201,106,Hangup

For the channel definition in the zapata.conf file, I have the following:

context = analog-phones
group = 3
pickupgroup = 3
signalling = fxo_ks
adsi = yes
mailbox = [EMAIL PROTECTED]
callerid = Phone 1 2201
channel = 9


I realize that I did not need to use the EXTEN variable, since I had unique
entries in this case.  I added [EMAIL PROTECTED] ( or could have used
the variable) and all works correctly.  Thank you.  I assumed that the
context entry in the voicemail_users table identified the mailbox
location.  In the past, before realtime, and with the mailboxes defined in
voicemail.conf, I did not have to append the context in the extension table.
I don't really care that it is required now, but why did it work before?

Regards,
Ed Horton


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Realtime voicemail

2005-04-25 Thread Edwin Horton
I have the latest Asterisk CVS as of 4/21/05 and a Fedora FC3 machine.  I
also set up the system to use Realtime for the voicemail mailboxes.  I am
successfully using Realtime for extensions and sip clients on this machine,
but as yet, cannot get the voicemail system to recognize the mailboxes as
defined in the MySQL database.  The other tables, Sip and Extensions are
part of the same database and they are accessed correctly.

When the voicemail system does a MySQL query, the debug output shows that
the correct mailbox is requested, but the context in the query is default,
not the context that should be active at the moment, in my case
analog-phones.  Of course, if I define the extension in the voicemail.conf
file, it works perfectly for the same context.

I must be doing something wrong, but I cannot see what.  Any help would be
greatly appreciated.

Ed Horton

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Asterisk as Media Gateway (was: ATT CallVantage Asterisk)

2005-04-16 Thread Edwin Horton
I noticed that there is some interest in MGCP slave operation for Asterisk
to enable it to work with the ATT Callvantage offering.  I have tried the
FXS/FXO connection to Asterisk and the Linksys TA with little success.
Dropped calls are the biggest problem, which does not occur when the phone
is directly connected to the TA.  Anyway, I am very interested in working on
this project if others are still interested.

Ed Horton

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] ParkAndAnnounce +${ALERT_INFO}

2005-01-26 Thread Edwin Horton
I am trying to use the _ALERT_INFO variable with ParkAndAnnounce.  My idea
was to have the phone, a Polycom IP500 auto answer so you could hear the
annoucement of the parked extension over the speaker.  This variable works
fine with the normal Dial application, but seems to be ignored by
ParkAndAnnounce.  I am not knowledgable enough to know if this is normal
operation, but a syntax error at my side.  Also, is it possiple to include
multiple SIP extensions in ParkAndAnnounce just as in the Dial application.
I tried the SIP/1001SIP/1002 context, but it was interpreted as a bad
extension by ParkAndAnnounce.  Thanks.
Ed Horton

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users