[asterisk-users] ODBC Voicemail Storage
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
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
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
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
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)
I noticed that there is some interest in MGCP slave operation for Asterisk to enable it to work with the AT&T 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}
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/1001&SIP/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
