I installed Asterisk 1.8.12 from the Fedora 16 (64-bit) yum repo, installed FreePBX 2.10.0 from source. I have everything working except: when I get a call and there is no answer, asterisk declines the call rather than roll it over into voicemail.
[2012-07-31 18:07:41] WARNING[9594] pbx.c: No application 'VoiceMail' for extension (macro-vm, s-NOANSWER, 2)
[2012-07-31 18:07:41] VERBOSE[9594] app_macro.c: == Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'SIP/vitel-inbound-0000001f' in macro 'vm'
[2012-07-31 18:07:41] VERBOSE[9594] app_macro.c: == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/vitel-inbound-0000001f' in macro 'exten-vm'
[2012-07-31 18:07:41] VERBOSE[9594] pbx.c: == Spawn extension (from-did-direct, 2, 2) exited non-zero on 'SIP/vitel-inbound-0000001f'
[2012-07-31 18:07:41] VERBOSE[9594] pbx.c: -- Executing [h@from-did-direct:1] Macro("SIP/vitel-inbound-0000001f", "hangupcall,") in new stack
[2012-07-31 18:07:41] VERBOSE[9594] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/vitel-inbound-0000001f", "1?theend") in new stack
[2012-07-31 18:07:41] VERBOSE[9594] pbx.c: -- Goto (macro-hangupcall,s,3)
[2012-07-31 18:07:41] VERBOSE[9594] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf("SIP/vitel-inbound-0000001f", "0?Set(CDR(recordingfile)=)") in new stack
[2012-07-31 18:07:41] VERBOSE[9594] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup("SIP/vitel-inbound-0000001f", "") in new stack
[2012-07-31 18:07:41] VERBOSE[9594] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/vitel-inbound-0000001f' in macro 'hangupcall'
[2012-07-31 18:07:41] VERBOSE[9594] pbx.c: == Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/vitel-inbound-0000001f'
[2012-07-31 18:07:41] VERBOSE[9594] chan_sip.c: Scheduling destruction of SIP dialog '58ea7e792fda9a531dddcb5574645137@...' in 32000 ms (Method: INVITE)
[2012-07-31 18:07:41] VERBOSE[9594] chan_sip.c:
<--- Reliably Transmitting (NAT) to ...:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP ...:5060;branch=z9hG4bK32f97411;received=...;rport=5060
From: "WIRELESS CALLER" ;tag=as3c330886
To: ;tag=as6cc122d7
Call-ID: 58ea7e792fda9a531dddcb5574645137@...
CSeq: 102 INVITE
Server: FPBX-2.10.1(1.8.12.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
When I look in freepbx, the voicemail for the extension says "unactivated". I've tried to connect with *98 but I always get "login incorrect" for the password I see/set in freepbx.
The "No Answer" destination is "Unavail Voicemail if Enabled" and voicemail is enabled for this extension.
Previous searches for solutions have indicated there's an issue loading the voicemail so file. It do have this one:
/usr/lib64/asterisk/modules/app_voicemail_odbc.so
but it's not loaded properly:
[2012-07-31 18:16:09] WARNING[10172] loader.c: Error loading module 'app_voicemail_odbc.so': /usr/lib64/asterisk/modules/app_voicemail_odbc.so: undefined symbol: _ast_odbc_request_obj
[2012-07-31 18:16:09] WARNING[10172] loader.c: Module 'app_voicemail_odbc.so' could not be loaded.
I suspect the last error is really responsible for all my problems. Anyone have ideas on how to fix this? Where is this undefined symbol defined?
Thanks.
Unless you have manually configured voicemail to use odbc, you would know if you had, then you could add noload => voicemail_odbc.so in /etc/asterisk/modules or delete the module in var/lib64/asterisk/modules. Most FreePPii will be using the more conventional app_voicemail.so which uses /etc/asterisk/voicemail.conf where the voicemail boxes will be in the [general] section.
Perhaps cleaner downline, would be to reinstall asterisk (also libpri dahdi if needed) from source (choose your poison from http://downloads.asterisk.org/pub/telephony/asterisk/) :-
./configure&&make clean&&make menuselect&&make&&make samples&&make config&&make install
when you run the "make menuselect" bit, choose the approriate selections for your perceived deployment.
The reason for this is that that ODBC voicemail is mutually exclusive with the flatform based app_voicemail.so
I uninstalled asterisk-voicemail-odbc and installed asterisk-voicemail-plain and voicemail is now working. Thanks!
As a cautionary note, there are significant bugs in Asterisk 1.8.12, I hope you took the oppertunity to get "up-to-date" . . .
Do you mean to "yum update" from the Fedora repo or update in Modules Admin? Or something else?
Thanks!
No!!!
I have no idea who packaged "Asterisk 1.8.12 from the Fedora 16 (64-bit) yum repo"
please go there for his/her suggested solution.
FreePBX Modules Admin has absolutely nothing to do with updateing Asterisk, they are two clearly different thingies.
Please read my post again. I clearly stated. . . .
. . . . .
Perhaps cleaner downline, would be to reinstall asterisk (also libpri dahdi if needed) from source (choose your poison from http://downloads.asterisk.org/pub/telephony/asterisk/) :-
./configure&&make clean&&make menuselect&&make&&make samples&&make config&&make install
when you run the "make menuselect" bit, choose the approriate selections for your perceived deployment.
. . . . .
Am I now more clearly understood ?
It wasn't clear what "up-to-date" meant, so I asked. If you had said "install from source" then I would have understood. Anyway, I understand now, thanks for the clarification.
I've installed from source the 1.8.11 cert5 release, which I assume is preferred over 1.8.15.0.
Thanks for your help, I appreciate it.
Personally I use 1.8.15 (currently latest) and so far find no problems, but YMMV
p.s. My post included
" .. . to reinstall asterisk . . . from source . . . . ."
I guess you missed that bit :-)
Dude, look at the messages. you said:
"As a cautionary note, there are significant bugs in Asterisk 1.8.12, I hope you took the oppertunity to get "up-to-date" . . ."
I asked
"Do you mean to "yum update" from the Fedora repo or update in Modules Admin? Or something else?"
and you said:
"No!!!
I have no idea who packaged "Asterisk 1.8.12 from the Fedora 16 (64-bit) yum repo"
please go there for his/her suggested solution.
FreePBX Modules Admin has absolutely nothing to do with updateing Asterisk, they are two clearly different thingies.
Please read my post again. I clearly stated. . . .
Perhaps cleaner downline, would be to reinstall asterisk (also libpri dahdi if needed) from source"
So, how would I know what you mean before you said it? That's what the "dude, chill" was meant to convey.
Glad you got if fixed, glad I could help, sorry you got hot under the collar.
Sorry your time machine is not working yet . . . :-)
I wasn't mad, just wanted to clarify. Yes, I don't have a working time machine, unfortunately. Thanks again for the help.