Re: [asterisk-users] VoiceMail greetings
On Sat, Nov 28, 2009 at 9:34 PM, matthieu Nicaise <
techni...@thinkrosystem.com> wrote:
>
> -rw-r--r-- 1 root root 4.1K 2009-11-28 23:47 unavail.WAV
> -rw-r--r-- 1 root root 4.1K 2009-11-28 23:47 unavail.gsm
> -rw-r--r-- 1 root root 40K 2009-11-28 23:47 unavail.wav
>
> I made an error in my first mail, i'm calling voicemail in extensions.conf
> this way :
>
> exten => _*.,1,Dial(SIP/${EXTEN:0},60)
> exten => _*.,n,VoiceMail(${EXTEN:0},u)
> exten => _*.,n,Playback(ss-noservice)
>
It would appear as though you haven't reloaded your dialplan since you've
added the 'u' option to your Voicemail() command, since it's not appearing
your cli output. Make sure your extensions.conf file has been saved and
then try "dialplan reload" in the cli and then try calling extension *11
again.
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] VoiceMail greetings
Thank you Jonathan and Warren, I now have the answer i needed ! Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ Le 29 nov. 09 à 04:41, Jonathan Thurman a écrit : On Sat, Nov 28, 2009 at 5:22 PM, matthieu Nicaise wrote: Hello everybody, I'm using Asterisk ( 1.6.1.9 ) Voicemail. For example, if i call extension *11 which is not registered, from extension *12, i have no greetings at all, i only have the "please leave a message after the beep". I tried to record the busy, unavailable and temporary greetings for extension *11 using VoiveMailMain and the file are well created on the file system. I cannot understand why those files are not played. If i use VoiceMail(*11) in the extension.conf i have exactly the same behaviour. If i user VoiceMail(*11,b) the busy message is read. Is that a normal behaviour ? I can't understand why Asterisk is not using the Dial status automaticaly. Thank you for your help ! The default option for voicemail is to play only the instructions. Take a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail for more details on the options. You will have to parse the Dial status in the dialplan, and pass 'u' for unavailable message to be played. You can see one way to parse the dial status in the sample extensions.conf file under the stdexten subroutine. There are lots of reasons to let the admin decide which greeting to play. For example, my canned 'receptionist' context plays the busy greeting as the after-hours greeting, otherwise playing the unavailable greeting. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail greetings
On Sat, Nov 28, 2009 at 7:34 PM, matthieu Nicaise
wrote:
> I made an error in my first mail, i'm calling voicemail in extensions.conf
> this way :
> exten => _*.,1,Dial(SIP/${EXTEN:0},60)
> exten => _*.,n,VoiceMail(${EXTEN:0},u)
> exten => _*.,n,Playback(ss-noservice)
You don't need the ":0", but that shouldn't cause any issues.
>> [Nov 29 03:38:13] WARNING[24635]: app_dial.c:1528 dial_exec_full: Unable
>> to create channel of type 'SIP' (cause 20 - Unknown)
>> == Everyone is busy/congested at this time (1:0/0/1)
>> -- Executing [...@local:2] VoiceMail("SIP/*15-0849a370", "*11") in new
>> stack
That last line should look like (from my 1.6.1.1 system):
-- Executing [...@local:2] VoiceMail("SIP/*15-0849a370", "*11","u")
in new stack
Did you reload the dialplan after the change?
-Jonathan
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Re: [asterisk-users] VoiceMail greetings
On Sat, Nov 28, 2009 at 5:22 PM, matthieu Nicaise wrote: > Hello everybody, > I'm using Asterisk ( 1.6.1.9 ) Voicemail. > For example, if i call extension *11 which is not registered, from extension > *12, i have no greetings at all, i only have the "please leave a message > after the beep". > I tried to record the busy, unavailable and temporary greetings for > extension *11 using VoiveMailMain and the file are well created on the file > system. > I cannot understand why those files are not played. > If i use VoiceMail(*11) in the extension.conf i have exactly the same > behaviour. > If i user VoiceMail(*11,b) the busy message is read. > Is that a normal behaviour ? > I can't understand why Asterisk is not using the Dial status automaticaly. > Thank you for your help ! The default option for voicemail is to play only the instructions. Take a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail for more details on the options. You will have to parse the Dial status in the dialplan, and pass 'u' for unavailable message to be played. You can see one way to parse the dial status in the sample extensions.conf file under the stdexten subroutine. There are lots of reasons to let the admin decide which greeting to play. For example, my canned 'receptionist' context plays the busy greeting as the after-hours greeting, otherwise playing the unavailable greeting. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail greetings
The content of the voicemail directory is :
ls -lh /var/spool/asterisk/voicemail/default/*11/
total 324K
drwxr-xr-x 2 root root 4.0K 2009-11-28 23:49 INBOX/
drwxr-xr-x 2 root root 4.0K 2009-11-28 23:46 Old/
drwxr-xr-x 2 root root 4.0K 2009-11-28 23:46 Urgent/
-rw-r--r-- 1 root root 3.5K 2009-11-28 23:47 busy.WAV
-rw-r--r-- 1 root root 3.5K 2009-11-28 23:47 busy.gsm
-rw-r--r-- 1 root root 34K 2009-11-28 23:47 busy.wav
-rw-r--r-- 1 root root 17K 2009-11-28 23:44 greet.WAV
-rw-r--r-- 1 root root 17K 2009-11-28 23:44 greet.gsm
-rw-r--r-- 1 root root 163K 2009-11-28 23:44 greet.wav
drwxr-xr-x 2 root root 4.0K 2009-11-28 23:49 tmp/
-rw-r--r-- 1 root root 4.1K 2009-11-28 23:47 unavail.WAV
-rw-r--r-- 1 root root 4.1K 2009-11-28 23:47 unavail.gsm
-rw-r--r-- 1 root root 40K 2009-11-28 23:47 unavail.wav
I made an error in my first mail, i'm calling voicemail in
extensions.conf this way :
exten => _*.,1,Dial(SIP/${EXTEN:0},60)
exten => _*.,n,VoiceMail(${EXTEN:0},u)
exten => _*.,n,Playback(ss-noservice)
Matthieu NICAISE
Responsable technique
GSM : 06 72 19 09 55
techni...@thinkrosystem.com
Thinkro System
http://www.thinkrosystem.com/
Le 29 nov. 09 à 04:26, Warren Selby a écrit :
On Sat, Nov 28, 2009 at 8:39 PM, matthieu Nicaise > wrote:
Here is the output of the CLI with verbose and debug set to 3 :
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Executing [...@local:1] Dial("SIP/*15-0849a370", "SIP/
*11,60") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
[Nov 29 03:38:13] WARNING[24635]: app_dial.c:1528 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [...@local:2] VoiceMail("SIP/*15-0849a370", "*11")
in new stack
-- Playing 'vm-intro.alaw' (language 'fr')
-- Playing 'beep.alaw' (language 'fr')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: wav49, 0x849b338
-- x=1, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: gsm, 0x849c7c0
-- x=2, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: wav, 0x849cb08
-- User hung up
== Parsing '/var/spool/asterisk/voicemail/default/*11/INBOX/
msg.txt': == Found
== Spawn extension (local, *11, 2) exited non-zero on 'SIP/
*15-0849a370'
-- Executing [...@local:1] Hangup("SIP/*15-0849a370", "") in new
stack
== Spawn extension (local, h, 1) exited non-zero on 'SIP/
*15-0849a370'
Th Warren
Matthieu NICAISE
Responsable technique
GSM : 06 72 19 09 55
techni...@thinkrosystem.com
Thinkro System
http://www.thinkrosystem.com/
What is the output of 'ls -lh /var/spool/asterisk/voicemail/default/
*11/' ?
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] VoiceMail greetings
On Sat, Nov 28, 2009 at 8:39 PM, matthieu Nicaise <
techni...@thinkrosystem.com> wrote:
> Here is the output of the CLI with verbose and debug set to 3 :
>
> == Using SIP RTP CoS mark 5
> == Using SIP VRTP CoS mark 6
> -- Executing [...@local:1] Dial("SIP/*15-0849a370", "SIP/*11,60") in
> new stack
> == Using SIP RTP CoS mark 5
> == Using SIP VRTP CoS mark 6
> [Nov 29 03:38:13] WARNING[24635]: app_dial.c:1528 dial_exec_full: Unable to
> create channel of type 'SIP' (cause 20 - Unknown)
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Executing [...@local:2] VoiceMail("SIP/*15-0849a370", "*11") in new
> stack
> -- Playing 'vm-intro.alaw' (language 'fr')
> -- Playing 'beep.alaw' (language 'fr')
> -- Recording the message
> -- x=0, open writing:
> /var/spool/asterisk/voicemail/default/*11/tmp/40taTt format: wav49,
> 0x849b338
> -- x=1, open writing:
> /var/spool/asterisk/voicemail/default/*11/tmp/40taTt format: gsm, 0x849c7c0
> -- x=2, open writing:
> /var/spool/asterisk/voicemail/default/*11/tmp/40taTt format: wav, 0x849cb08
> -- User hung up
> == Parsing '/var/spool/asterisk/voicemail/default/*11/INBOX/msg.txt':
> == Found
> == Spawn extension (local, *11, 2) exited non-zero on 'SIP/*15-0849a370'
> -- Executing [...@local:1] Hangup("SIP/*15-0849a370", "") in new stack
> == Spawn extension (local, h, 1) exited non-zero on 'SIP/*15-0849a370'
>
> Th Warren
>
> Matthieu NICAISE
> Responsable technique
>
> GSM : 06 72 19 09 55
> techni...@thinkrosystem.com
>
> Thinkro System
> http://www.thinkrosystem.com/
>
What is the output of 'ls -lh /var/spool/asterisk/voicemail/default/*11/' ?
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] VoiceMail greetings
Here is the output of the CLI with verbose and debug set to 3 :
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Executing [...@local:1] Dial("SIP/*15-0849a370", "SIP/*11,60")
in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
[Nov 29 03:38:13] WARNING[24635]: app_dial.c:1528 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [...@local:2] VoiceMail("SIP/*15-0849a370", "*11")
in new stack
-- Playing 'vm-intro.alaw' (language 'fr')
-- Playing 'beep.alaw' (language 'fr')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: wav49, 0x849b338
-- x=1, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: gsm, 0x849c7c0
-- x=2, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: wav, 0x849cb08
-- User hung up
== Parsing '/var/spool/asterisk/voicemail/default/*11/INBOX/
msg.txt': == Found
== Spawn extension (local, *11, 2) exited non-zero on 'SIP/
*15-0849a370'
-- Executing [...@local:1] Hangup("SIP/*15-0849a370", "") in new
stack
== Spawn extension (local, h, 1) exited non-zero on 'SIP/
*15-0849a370'
Th Warren
Matthieu NICAISE
Responsable technique
GSM : 06 72 19 09 55
techni...@thinkrosystem.com
Thinkro System
http://www.thinkrosystem.com/
Le 29 nov. 09 à 03:19, Warren Selby a écrit :
Do you have *11 registered in your voicemail.conf file? What does
the cli output look like when you try to leave a voicemail?
Thanks,
--Warren Selby
On Nov 28, 2009, at 7:22 PM, matthieu Nicaise > wrote:
Hello everybody,
I'm using Asterisk ( 1.6.1.9 ) Voicemail.
For example, if i call extension *11 which is not registered, from
extension *12, i have no greetings at all, i only have the "please
leave a message after the beep".
I tried to record the busy, unavailable and temporary greetings for
extension *11 using VoiveMailMain and the file are well created on
the file system.
I cannot understand why those files are not played.
If i use VoiceMail(*11) in the extension.conf i have exactly the
same behaviour.
If i user VoiceMail(*11,b) the busy message is read.
Is that a normal behaviour ?
I can't understand why Asterisk is not using the Dial status
automaticaly.
Thank you for your help !
Matthieu NICAISE
Responsable technique
GSM : 06 72 19 09 55
techni...@thinkrosystem.com
Thinkro System
http://www.thinkrosystem.com/
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Re: [asterisk-users] VoiceMail greetings
Do you have *11 registered in your voicemail.conf file? What does the cli output look like when you try to leave a voicemail? Thanks, --Warren Selby On Nov 28, 2009, at 7:22 PM, matthieu Nicaise > wrote: Hello everybody, I'm using Asterisk ( 1.6.1.9 ) Voicemail. For example, if i call extension *11 which is not registered, from extension *12, i have no greetings at all, i only have the "please leave a message after the beep". I tried to record the busy, unavailable and temporary greetings for extension *11 using VoiveMailMain and the file are well created on the file system. I cannot understand why those files are not played. If i use VoiceMail(*11) in the extension.conf i have exactly the same behaviour. If i user VoiceMail(*11,b) the busy message is read. Is that a normal behaviour ? I can't understand why Asterisk is not using the Dial status automaticaly. Thank you for your help ! Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com --- - Thinkro System http://www.thinkrosystem.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail greetings
Hello everybody, I'm using Asterisk ( 1.6.1.9 ) Voicemail. For example, if i call extension *11 which is not registered, from extension *12, i have no greetings at all, i only have the "please leave a message after the beep". I tried to record the busy, unavailable and temporary greetings for extension *11 using VoiveMailMain and the file are well created on the file system. I cannot understand why those files are not played. If i use VoiceMail(*11) in the extension.conf i have exactly the same behaviour. If i user VoiceMail(*11,b) the busy message is read. Is that a normal behaviour ? I can't understand why Asterisk is not using the Dial status automaticaly. Thank you for your help ! Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Greetings Will Not Save
Hi All, Just wanted to post a follow up in case anyone else has the same issue in the future. I recompiled Asterisk and in the makemenu system there is a Voicemail Build Options, in there there is []ODBC Storage and []IMAP Storage. I had ODBC Storage checked on my last compile, I unchecked it, finished building and it all works now. Apparenlty this does not install the option of using ODBC storage, it commits you to ODBC storage without any additional configuration. Tilghman, thanks, your question is what ultimately led me to my solution. Respectfully, Dr. Kenneth Noisewater, Phd On Apr 13, 2009, at 6:14 PM, Tilghman Lesher wrote: > On Monday 13 April 2009 05:54:38 pm Dr. Kenneth Noisewater wrote: >> Hi All, >> >> -My asterisk will not save voicemail greetings when you call in and >> record them. >> -It also will not save voicemail messages after emailing them,even >> though delete=no. >> -Folder permissions are fine, no errors in asterisk cli. >> -If i go into /var/spool/asterisk/voicemail/default/200 and touch >> unavail.wav, and then call in and record new unavail message, >> unavail.wav disappears? >> >> Can anyone help point me towards any possible info to fix this, i'm >> stumped and losing hair! > > You wouldn't happen to have built voicemail with ODBC and/or IMAP > support, would you? That would make the most sense, as both of > those engines remove recordings from the directory after having > sucked them into the relevant backend storage device. > > -- > Tilghman > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Greetings Will Not Save
I very probably did build them with ODBC or MySQL support. IMAP I don't think so, but where would I look for configs that tell asterisk to use such support? I'm almost positive I compiled it to support database, but I definitely never configured it for use. Or is this something it does automatically and I need to recompile? Thank you very much for your help. On Apr 13, 2009, at 6:14 PM, Tilghman Lesher wrote: > On Monday 13 April 2009 05:54:38 pm Dr. Kenneth Noisewater wrote: >> Hi All, >> >> -My asterisk will not save voicemail greetings when you call in and >> record them. >> -It also will not save voicemail messages after emailing them,even >> though delete=no. >> -Folder permissions are fine, no errors in asterisk cli. >> -If i go into /var/spool/asterisk/voicemail/default/200 and touch >> unavail.wav, and then call in and record new unavail message, >> unavail.wav disappears? >> >> Can anyone help point me towards any possible info to fix this, i'm >> stumped and losing hair! > > You wouldn't happen to have built voicemail with ODBC and/or IMAP > support, would you? That would make the most sense, as both of > those engines remove recordings from the directory after having > sucked them into the relevant backend storage device. > > -- > Tilghman > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Greetings Will Not Save
On Monday 13 April 2009 05:54:38 pm Dr. Kenneth Noisewater wrote: > Hi All, > > -My asterisk will not save voicemail greetings when you call in and > record them. > -It also will not save voicemail messages after emailing them,even > though delete=no. > -Folder permissions are fine, no errors in asterisk cli. > -If i go into /var/spool/asterisk/voicemail/default/200 and touch > unavail.wav, and then call in and record new unavail message, > unavail.wav disappears? > > Can anyone help point me towards any possible info to fix this, i'm > stumped and losing hair! You wouldn't happen to have built voicemail with ODBC and/or IMAP support, would you? That would make the most sense, as both of those engines remove recordings from the directory after having sucked them into the relevant backend storage device. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Greetings Will Not Save
Hi All,
-My asterisk will not save voicemail greetings when you call in and
record them.
-It also will not save voicemail messages after emailing them,even
though delete=no.
-Folder permissions are fine, no errors in asterisk cli.
-If i go into /var/spool/asterisk/voicemail/default/200 and touch
unavail.wav, and then call in and record new unavail message,
unavail.wav disappears?
Can anyone help point me towards any possible info to fix this, i'm
stumped and losing hair!
I am running
FreePBX 2.5.1
Asterisk 1.4.23.1
CentOS 5.3
CLI Output during my attempt to call in and record a greeting:
-- Executing [...@from-internal:4] Macro("SIP/200-00fd3150", "get-
vmcontext|200") in new stack
-- Executing [...@macro-get-vmcontext:1] Set40m("SIP/200-00fd3150",
"VMCONTEXT=default") in new stack
-- Executing [...@macro-get-vmcontext:2] GotoIf("SIP/200-00fd3150",
"0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [...@macro-get-vmcontext:300] NoOp("SIP/200-00fd3150",
"") in new stack
-- Executing [...@from-internal:5] MailboxExists37;40m("SIP/
200-00fd3150", "2...@default") in new stack
-- Executing [...@from-internal:6] GotoIf("SIP/200-00fd3150", "1?
mbexist") in new stack
-- Goto (from-internal,*97,106)
-- Executing [...@from-internal:106] VoiceMailMain("SIP/
200-00fd3150", "2...@default") in new stack
-- Playing 'vm-password' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-options' (language 'en')
-- Recording the message
-- Playing 'vm-rec-unv' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
-- Playing 'beep' (language 'en')
-- x=0, open writing: /var/spool/asterisk/voicemail/default/200/
unavail.tmp format: wav49, 0x97d788
-- x=1, open writing: /var/spool/asterisk/voicemail/default/200/
unavail.tmp format: gsm, 0x9eecd8
-- x=2, open writing: /var/spool/asterisk/voicemail/default/200/
unavail.tmp format: wav, 0xa4f778
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
-- User ended message by pressing #
-- Playing 'auth-thankyou' (language 'en')
-- Playing 'vm-review' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
-- Saving message as is
-- Playing 'vm-msgsaved' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
-- Playing 'vm-options' (language 'en')
== Manager 'admin' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
== Spawn extension (from-internal, *97, 106) exited non-zero on
'SIP/200-00fd3150'
-- Executing [...@from-internal:1] Macro("SIP/200-00fd3150",
"hangupcall") in new stack
-- Executing [...@macro-hangupcall:1] ResetCDR("SIP/200-00fd3150",
"w") in new stack
-- Executing [...@macro-hangupcall:2] NoCDR("SIP/200-00fd3150", "")
in new stack
-- Executing [...@macro-hangupcall:3] GotoIf("SIP/200-00fd3150", "1?
skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [...@macro-hangupcall:6] GotoIf("SIP/200-00fd3150", "1?
skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [...@macro-hangupcall:9] GotoIf("SIP/200-00fd3150", "1?
theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [...@macro-hangupcall:11] Hangup("SIP/200-00fd3150",
"") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/200-00fd3150' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/
200-00fd3150'
Here is voicemail.conf:
[general]
format = wav49|gsm|wav
s
Re: [asterisk-users] Voicemail Greetings Will Not Save
Dr. Kenneth Noisewater wrote: > > -Asterisk and FreePBX source installs on CentOS 5.4 > Without version numbers and console output and samples of your dialplan, it'g going to make it very difficult to help. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Greetings Will Not Save
Hi All, -My asterisk will not save voicemail greetings when you call in and record them. -It also will not save voicemail messages after emailing them,even though delete=no. -Folder permissions are fine, no errors in asterisk cli. -If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears? -Asterisk and FreePBX source installs on CentOS 5.4 Can anyone help point me towards any possible info to fix this, i'm stumped and losing hair! Respectfully, Dr. Kenneth Noisewater, Phd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail greetings
On Jan 31, 2006, at 10:03 AM, Michaël Gaudette wrote: Hi, I`ve been trying to figure out voicemail, but there is something that is obviously escaping me. Using * 1.2.3, standard built with asterisk-addons. I have two voicemails, one is 702 and one is 705. Both in different contexts, but that doesn`t matter (I think). The point is in the /voicemail/context/702 directory I have the files unavail.gsm, temp.gsm and greet.gsm. While in the other directory, I have greet.gsm, unavail.gsm and busy.gsm. So in one directory I have temp.gsm and in the other busy.gsm. How did that happen and what does it mean? What i found out is that in the one voicemail that doesn`t have temp.gsm, when somebody tries to leave me a message that person gets an asterisk greeting (as opposed to one with my wonderful voice). Also, WHEN are the file used? I have the option of recording my busy message and my unavailable message, but really, how does Asterisk choose which one I am? (unavailable vs busy)? This isn`t clear to me, hopefully somebody has a quick and simple answer. simplified answer If the phone rings then goes to vm it is unavailable. If the phone does not ring it is busy. More detailed If you phone is set to allow n calls to ring in and n+1 tries, then the phone will return a busy If your phone is in do not disturb it will return a busy If you phone is unreachable it will return a busy Mostly depends on your phones exactly how/when things happen, but definately controllable from the dialplan also ymmv___ --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] Voicemail greetings
Hi, I`ve been trying to figure out voicemail, but there is something that is obviously escaping me. Using * 1.2.3, standard built with asterisk-addons. I have two voicemails, one is 702 and one is 705. Both in different contexts, but that doesn`t matter (I think). The point is in the /voicemail/context/702 directory I have the files unavail.gsm, temp.gsm and greet.gsm. While in the other directory, I have greet.gsm, unavail.gsm and busy.gsm. So in one directory I have temp.gsm and in the other busy.gsm. How did that happen and what does it mean? What i found out is that in the one voicemail that doesn`t have temp.gsm, when somebody tries to leave me a message that person gets an asterisk greeting (as opposed to one with my wonderful voice). Also, WHEN are the file used? I have the option of recording my busy message and my unavailable message, but really, how does Asterisk choose which one I am? (unavailable vs busy)? This isn`t clear to me, hopefully somebody has a quick and simple answer. Mike ___ --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
