[asterisk-users] voicemail problem
Hi:
I configured asterisk for voicemail service.My main configuration files are:
extensions.conf
[from-pstn]
exten => 9711315,1,Dial(SIP/3000,30)
exten => 9711315,2,VoiceMail([EMAIL PROTECTED])
exten => 9711315,3,PlayBack(vm-goodbye)
exten => 9711315,4,HangUp()
voicemail.conf
[ff_tutorial]
555 => 1234567,3000,[EMAIL PROTECTED]
sip.conf
[3000]
type=friend
username=3000
secret=1234567
host=dynamic
context=from-pstn
[EMAIL PROTECTED]
But when I dial 9711315, after 30s I hear goodbye and call hangups.
in console:
-- Accepting call from '3322000' to '9711315' on channel 0/2, span 1
-- Executing Dial("Zap/2-1", "SIP/3000|30") in new stack
-- Called 3000
-- SIP/3000-08f18698 is ringing
Jun 24 11:55:32 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference
Length not supported: 0
Jun 24 11:55:42 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference
Length not supported: 0
Jun 24 11:55:52 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference
Length not supported: 0
-- Nobody picked up in 3 ms
-- Executing VoiceMail("Zap/2-1", "[EMAIL PROTECTED]") in new stack
Jun 24 11:55:53 WARNING[5188]: app_voicemail.c:2461 leave_voicemail: No entry
in voicemail config file for '555'
-- Executing Playback("Zap/2-1", "vm-goodbye") in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing Hangup("Zap/2-1", "") in new stack
== Spawn extension (from-pstn, 9711315, 4) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
Jun 24 11:56:02 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference
Length not supported: 0
what's problem?
should I do something in sip phone for voicemail?
I'd appreciate any help.
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[asterisk-users] Voicemail Problem
I have a problem with the current trunk code for 1.6.0 as it relates to voicemail. I had the same problem in a previous trunk version as well so I just updated myself to current code - Asterisk SVN-branch-1.6.0-r184281M I have voicemail using ODBC storage. When a new voicemail message is left and the system is, I am guessing, trying to generate the email notification it core dumps. Here is what was on the console: [2009-03-26 10:20:04.814] -- Saving message as is [2009-03-26 10:20:04.815] -- Playing 'vm-msgsaved.gsm' (language 'en') [2009-03-26 10:20:06.757] == Parsing '/var/spool/asterisk/voicemail/ourvm/108/INBOX/msg0001.txt': [2009-03-26 10:20:06.757] == Found The message is saved in the database as I can retrieve the voicemail from the phone. It just seems there is some problem with email notification. Has anyone seen this problem as well? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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
[asterisk-users] voicemail problem
i had configured voicemail, here is the config files voicemail.conf, sip.conf,
extensions.conf, zaptel.conf, zapata.conf
voice mail is working when ever call is received, extension 2000 receives it
and if not answered in 20 secs, message is stored in
voicemail no problem in that. after creating voice mail if some one again call
at that no this time even bell dosent ring, busy
tone is heard, but when i restart machine call can be rceived in extension
2000, but as soon voicemail is created after 20 secs,
same problem than no one can call at that no,again gives busy tone.
###
[r...@localhost ~]# vi /etc/asterisk/voicemail.conf
[general]
format = wav
attach = yes
[default]
; Syntax for new entries looks like this:
; MailboxNumber => password,name,e-mail,pager,options
; (usually, the MailboxNumber is the same as the Extension)
2000 => 1234,abc,a...@abc.com
2001 => 1234,def,d...@def.com
#
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=voicemail
secret=1234
host=dynamic
[2001]
type=friend
context=voicemail
secret=1234
host=dynamic
#
vi /extensions.conf
[r...@localhost ~]# vi /etc/asterisk/extensions.conf
[from-zaptel]; plz check zapata.conf
exten => s,1,wait(2)
exten => s,n,Goto(voicemail,2000,1)
[voicemail]
exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,VoiceMail(2000,u)
exten => 2999,1,VoiceMailMain(${CALLERID(num)},s)
#
vi zaptel.conf
#autogenerated by /usr/sbin/wancfg_zaptel do not hand edit
#autogenrated on 2010-02-09
#Zaptel Channels Configurations
#For detailed Zaptel options, view /etc/zaptel.conf.bak
loadzone=us
defaultzone=us
#Sangoma USB U100 [bus:2-3 span:1]
fxsks=1
fxsks=2
###
vi zapata.conf
;autogenerated by /usr/sbin/wancfg_zaptel do not hand edit
;autogenrated on 2010-02-09
;Zaptel Channels Configurations
;For detailed Zaptel options, view /etc/asterisk/zapata.conf.bak
[trunkgroups]
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma AU100 [slot:0 bus: span:1]
context=from-zaptel
group=0
signalling = fxs_ks
channel => 1
context=from-zaptel
group=0
signalling = fxs_ks
channel => 2
#
Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!
http://downloads.yahoo.com/in/internetexplorer/i had configured voicemail, here is the config files voicemail.conf, sip.conf,
extensions.conf, zaptel.conf, zapata.conf
voice mail is working when ever call is received, extension 2000 receives it
and if not answered in 20 secs, message is stored in
voicemail no problem in that. after creating voice mail if some one again call
at that no this time even bell dosent ring, busy
tone is heard, but when i restart machine call can be rceived in extension
2000, but as soon voicemail is created after 20 secs,
same problem than no one can call at that no,again gives busy tone.
###
[r...@localhost ~]# vi /etc/asterisk/voicemail.conf
[general]
format = wav
attach = yes
[default]
; Syntax for new entries looks like this:
; MailboxNumber => password,name,e-mail,pager,options
; (usually, the MailboxNumber is the same as the Extension)
2000 => 1234,abc,a...@abc.com
2001 => 1234,def,d...@def.com
#
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=voicemail
secret=1234
host=dynamic
[2001]
type=friend
context=voicemail
secret=1234
host=dynamic
#
vi /extensions.conf
[r...@localhost ~]# vi /etc/asterisk/extensions.conf
[from-zaptel]; plz check zapata.conf
exten => s,1,wait(2)
exten => s,n,Goto(voicemail,2000,1)
[voicemail]
exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,VoiceMail(2000,u)
exten => 2999,1,VoiceMailMain(${CALLERID(num)},s)
#
vi zaptel.conf
#autogenerated by /usr/sbin/wancfg_zaptel do not hand edit
#autogenrated on 2010-02-09
#Zaptel Channels Configurations
#For detailed Zaptel options, view /etc/zaptel.conf.bak
loadzone=us
def
[asterisk-users] voicemail problem
Hi people! I am running Asterisk 1.6.1.12 and I set up the voicemail. That far I have set upt the voicemailbox with my personal greeting message. If somebody calls me and is forwarded to my mailbox, my personal recorded greeting is played back + the default message "please record your message after the tone and hang up or press the pound key". Is there a way to delete the second part from the voicemail, that only my personal recorded message is played back and a signal tone comes to signal the caller to start talking?! Tamer Higazi -- _ -- 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] Voicemail problem
I setted up my *
mailbox. Howerver, when I access my mailbox by extension 8. I cannot hear the
prompt to input mailbox number and PIN. * console tells me: "RFC3389 support
imcomplete. Turn off on client if possible". Here is the complete log. Can
anybody tell me how to let it work.
Thank
you,
Wei
Jan 6 10:43:03 WARNING[6150]:
chan_sip.c:2771 process_sdp: No compatible codecs !Jan 6 10:43:04
WARNING[6150]: chan_sip.c:2771 process_sdp: No compatible codecs
! -- Executing Ringing("SIP/2201-76bf", "") in new
stack -- Executing Wait("SIP/2201-76bf", "2") in new
stack -- Executing VoiceMailMain("SIP/2201-76bf", "") in
new stack -- Playing 'vm-login' (language
'en') -- Got SIP response 481 "Call Leg/Transaction Does
Not Exist" back from 192.168.1.102RFC3389: 1 bytes, level
4...Jan 6 10:43:08 NOTICE[23567]: rtp.c:289 process_rfc3389: RFC3389
support incomp lete. Turn off on client if possibleRFC3389: 1 bytes,
level 4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level
4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level 4...RFC3389:
1 bytes, level 4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level
4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level 4...RFC3389:
1 bytes, level 4... -- Username not enteredRFC3389: 1
bytes, level 4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level
4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level 4...RFC3389:
1 bytes, level 4... -- Timeout on SIP/2201-76bf
== CDR updated on SIP/2201-76bf -- Executing
Goto("SIP/2201-76bf", "#|1") in new stack -- Goto
(default,#,1) -- Executing Playback("SIP/2201-76bf",
"demo-thanks") in new stack -- Playing 'demo-thanks'
(language 'en')RFC3389: 1 bytes, level 4...Jan 6 10:43:34
WARNING[23567]: file.c:548 ast_readaudio_callback: Failed to write
frame
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[Asterisk-Users] Voicemail problem
I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,[EMAIL PROTECTED] 101 => 1234,Jason Madden,[EMAIL PROTECTED] 102 => 1234,Melinda Garland,[EMAIL PROTECTED] Sean Garland, MCP+I, A+ Siskiyou Technology Consultants ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 problem
I'm trying the voicemail but I can't receive nothing in my mail account, the message records well but it does not seem to deliver anything... what I'm doing wrong? Thanxs! ___ 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 Problem
I have just setup my
OPENSER to work with the asterisk 1.2.2.
I've set extension
400 in extension.conf to point to the VoicemailMain()
application
The entire program
works fine, but there seems to be some problem whenever the call is hangup,
either by pushing # to exit the VoicemailMain() apps or by hanging the phone. If
the # button is push, should Asterisk send something back to tell OPENSER to
hang up the party ?
Here's the log of
verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave' (language 'en') -- Playing 'vm-no'
(language 'en') -- Playing 'vm-messages' (language
'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-goodbye' (language
'en') -- Executing Playback("SIP/210.23.1.139-081ee3d8",
"Goodbye") in new stackFeb 9 15:05:06
WARNING[23242]: file.c:509 ast_openstream_full: File Goodbye does not exist in
any formatFeb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile:
Unable to open Goodbye (format alaw): No such file or directoryFeb
9 15:05:06 WARNING[23242]: app_playback.c:132 playback_exec: ast_streamfile
failed on SIP/203.125.68.66-081ee3d8for Goodbye --
Executing Hangup("SIP/203.125.68.66-081ee3d8", "") in new stack ==
Spawn extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is
this all about ?
Regards,Sam
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[Asterisk-Users] Voicemail Problem
(sent this earlier with my gmail account, think there is a problem there so I'm sending it here. If anyone replied to this, please resend it to this email address. thanks alot) I am new to asterisk and I'm setting up a test box to flesh out a switchover we're going to do at work. Right now I'm working on voicemail. I can leave a message fine, but when I attempt to listen to messages, I am having trouble. I can dial an extension for voice mail main, login, and it'll tell me how many messages I have. I press one and it will give me the date/time for the message but when playback would normally start, VMMain hangs up. The message in the * console is: (removed sound playback lines) Parsing '/var/spool/asterisk/voicemail/default/105/Old/ msg.txt': Found (playback some sound files) == Spawn extension (internal, 500, 1) exited non-zero on 'SIP/ joseph-0e7f' At this point it hangs up on me. I'm using X-Lite softphone for testing (can't buy any phones till we're sure we have it all working right :) ) Thanks in advance for any help you guys can give. I'll include the appropriate config stuff below as well: sip.conf: (nothing in [general]) [joseph] ;exten105 type=friend secret=welcome qualify=yes nat=no host=dynamic canreinvite=no context=internal [EMAIL PROTECTED] voicemail.conf: [default] 105 => 1234,Joseph Blake,[EMAIL PROTECTED] extensions.conf: [internal] exten => 1,1,Answer() exten => 1,2,Playback(all-your-base) exten => 1,3,Hangup() exten => 105,1,Answer() exten => 105,2,Dial(SIP/joseph,30) exten => 105,3,VoiceMail([EMAIL PROTECTED]) exten => 105,4,Hangup() exten => 500,1,VoiceMailMain() ___ --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 problem
Hello ! I am using asterisk at home 1.5, and i have a really big program. I setup multiple extensions, all with voicemail feature, but the voicemail does not kick in at all. Any idea what might be wrong ? I am allowing all possible codecs, but i cannot see what is possible wrong. Thank you in advance ! ___ --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 problem
Hi, when I call the voicemail app, it starts and die suddenly. Has anyone already had this problem? Log: app.c:644 ast_play_and_record: No audio available on SIP/-6fca?? -- User hung up Tks, D.K. ___ --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
Re: [asterisk-users] voicemail problem
On Tuesday 17 June 2008 04:05:58 fateme fatah wrote:
> I configured asterisk for voicemail service.My main configuration files
> are:
>
>
> voicemail.conf
>
> [ff_tutorial]
> 555 => 1234567,3000,[EMAIL PROTECTED]
>
> But when I dial 9711315, after 30s I hear goodbye and call hangups.
>
> in console:
>
> -- Accepting call from '3322000' to '9711315' on channel 0/2, span 1
> -- Executing Dial("Zap/2-1", "SIP/3000|30") in new stack
> -- Called 3000
> -- SIP/3000-08f18698 is ringing
> -- Nobody picked up in 3 ms
> -- Executing VoiceMail("Zap/2-1", "[EMAIL PROTECTED]") in new stack
> Jun 24 11:55:53 WARNING[5188]: app_voicemail.c:2461 leave_voicemail: No
> entry in voicemail config file for '555'
Did you reload after changing voicemail.conf? What is the output of
'voicemail show users'?
--
Tilghman
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Re: [asterisk-users] voicemail problem
Since the application just does a playback of the "canned" sounds in /var/lib/sounds/asterisk, you can use SOX, Audacity, etc. to mix and chop these sounds in whatever way you see fit. Do a core set verbose 10 on the CLI and watch the output as you leave a voicemail to see which files to tweak. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi Sent: Monday, March 22, 2010 6:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voicemail problem Hi people! I am running Asterisk 1.6.1.12 and I set up the voicemail. That far I have set upt the voicemailbox with my personal greeting message. If somebody calls me and is forwarded to my mailbox, my personal recorded greeting is played back + the default message "please record your message after the tone and hang up or press the pound key". Is there a way to delete the second part from the voicemail, that only my personal recorded message is played back and a signal tone comes to signal the caller to start talking?! Tamer Higazi -- _ -- 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 -- _ -- 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 problem
Hi Use the us option and not b _/List with the possible options/_ /*s* - _without_ this option a message will be played. The message by default is: "Please leave your message after the tone. When done, hang up, or press the pound key." If you _set_ this option, the message won’t be played. *u* - If you set this option, an unavailable message will be played. The message by default is: "The person at extension is unavailable". Also you will hear and the instructions: "Please leave your message after the tone. When done, hang up, or press the pound key." *b* - If you set this option, a busy message will be played. The message by default is: "The person at extension is on the phone." Also you will hear and the instructions: "Please leave your message after the tone. When done, hang up, or press the pound key." *su* - You will hear the unavailable message: "The person at extension is unavailable". The instruction message will be skipped. *sb* - You will hear the busy message: "The person at extension is on the phone". The instruction message will be skipped./ Ish Tamer Higazi wrote: > Hi people! > I am running Asterisk 1.6.1.12 and I set up the voicemail. That far I > have set upt the voicemailbox with my personal greeting message. If > somebody calls me and is forwarded to my mailbox, my personal recorded > greeting is played back + > > the default message "please record your message after the tone and hang > up or press the pound key". > > Is there a way to delete the second part from the voicemail, that only > my personal recorded message is played back and a signal tone comes to > signal the caller to start talking?! > > > Tamer Higazi > > -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 problem
turning off VAD and silence suppression at the client can
solve this problem.
Yusuf
Alakavuk
Teknik Danışman - Technical
Consultant
Grid Bilişim
Teknolojileri A.Ş.
Kuştepe Mahallesi Leylak
Sokak
Murat İş Merkezi A Blok Kat:2
Daire:9
34387 Şişli İstanbul
Türkiye
Tel :
+90 (212) 336 92 55
Fax : +90
(212) 266 25 50
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wei
SuSent: 06 Ocak 2005 Perşembe 20:56To:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Voicemail
problem
I setted up my *
mailbox. Howerver, when I access my mailbox by extension 8. I cannot hear the
prompt to input mailbox number and PIN. * console tells me: "RFC3389 support
imcomplete. Turn off on client if possible". Here is the complete log. Can
anybody tell me how to let it work.
Thank you,
Wei
Jan 6 10:43:03 WARNING[6150]:
chan_sip.c:2771 process_sdp: No compatible codecs !Jan 6 10:43:04
WARNING[6150]: chan_sip.c:2771 process_sdp: No compatible codecs
! -- Executing Ringing("SIP/2201-76bf", "") in new
stack -- Executing Wait("SIP/2201-76bf", "2") in new
stack -- Executing VoiceMailMain("SIP/2201-76bf", "") in
new stack -- Playing 'vm-login' (language
'en') -- Got SIP response 481 "Call Leg/Transaction Does
Not Exist" back from 192.168.1.102RFC3389: 1 bytes, level
4...Jan 6 10:43:08 NOTICE[23567]: rtp.c:289 process_rfc3389: RFC3389
support incomp lete. Turn off on client if possibleRFC3389: 1 bytes,
level 4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level
4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level 4...RFC3389:
1 bytes, level 4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level
4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level 4...RFC3389:
1 bytes, level 4... -- Username not enteredRFC3389: 1
bytes, level 4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level
4...RFC3389: 1 bytes, level 4...RFC3389: 1 bytes, level 4...RFC3389:
1 bytes, level 4... -- Timeout on SIP/2201-76bf
== CDR updated on SIP/2201-76bf -- Executing
Goto("SIP/2201-76bf", "#|1") in new stack -- Goto
(default,#,1) -- Executing Playback("SIP/2201-76bf",
"demo-thanks") in new stack -- Playing 'demo-thanks'
(language 'en')RFC3389: 1 bytes, level 4...Jan 6 10:43:34
WARNING[23567]: file.c:548 ast_readaudio_callback: Failed to write
frame
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RE: [Asterisk-Users] Voicemail problem
Sean, I use the sendmail app on the pbx itself (redhat 9.1) with the serveremail=localhost Not a lot of overhead on this process, of course sendmail needs to be able to route to the internet to send out mail, so this can't be a private subnet only pbx. -Bryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Garland Sent: Friday, June 11, 2004 3:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicemail problem I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,[EMAIL PROTECTED] 101 => 1234,Jason Madden,[EMAIL PROTECTED] 102 => 1234,Melinda Garland,[EMAIL PROTECTED] Sean Garland, MCP+I, A+ Siskiyou Technology Consultants ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.701 / Virus Database: 458 - Release Date: 6/7/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.701 / Virus Database: 458 - Release Date: 6/7/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail problem
How do you specify sendmail, or any mail program? I changed the servermail= to equal my in-house exchange server, and allowed relaying by it's the pbx's IP address, but I still don't understand how it know where to send or what program it uses.. Thanks Sean -Original Message- From: public [mailto:[EMAIL PROTECTED] Sent: Friday, June 11, 2004 2:50 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voicemail problem Sean, I use the sendmail app on the pbx itself (redhat 9.1) with the serveremail=localhost Not a lot of overhead on this process, of course sendmail needs to be able to route to the internet to send out mail, so this can't be a private subnet only pbx. -Bryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Garland Sent: Friday, June 11, 2004 3:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicemail problem I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,[EMAIL PROTECTED] 101 => 1234,Jason Madden,[EMAIL PROTECTED] 102 => 1234,Melinda Garland,[EMAIL PROTECTED] Sean Garland, MCP+I, A+ Siskiyou Technology Consultants ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.701 / Virus Database: 458 - Release Date: 6/7/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.701 / Virus Database: 458 - Release Date: 6/7/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail problem
- Original Message - > From: "Sean Garland" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, June 15, 2004 9:53 PM > Subject: RE: [Asterisk-Users] Voicemail problem > How do you specify sendmail, or any mail program? I changed the > servermail= to equal my in-house exchange server, and allowed relaying > by it's the pbx's IP address, but I still don't understand how it know > where to send or what program it uses.. In voicemail.conf.. fromstring=The Asterisk PBX ; Change the From: string serveremail=asterisk(atsign)domain.com ; Who the e-mail notification should appear to come from mailcmd=/usr/sbin/sendmail -t ; You can override the default program to send e-mail Note: "fromstring" & "serveremail" are translated into "The Asterisk PBX" in your email "From:" address. section of email header Date: Wed, 09 Jun 2004 21:04:28 +0200 From: The Asterisk PBX To: Soren soren(atsign)domain.com Subject: New VM (1) - 2:04 long in mailbox 100 from "Joe User" <12345678> Message-ID: Asterisk-1-100-2792(atsign)asterisk.domain.com On your server you install/setup/configure sendmail and have it point to your normal mailserver as relaymailer, this way you can control who gets what from your normal mailserver. As fas as I remember, the only thing I changed in /etc/var/sendmail.cf was: # "Smart" relay host (may be null) DSmail.domain.com to enable relaying via mail.domain.com. It may not be the most secure way to do stuff, but I have everything behind NAT and Firewall with SMTP traffic only allowed to/from my regular mailserver. -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail Problem
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday, February 09, 2006 3:30 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Voicemail
Problem
I have just setup my
OPENSER to work with the asterisk 1.2.2.
I've set extension
400 in extension.conf to point to the VoicemailMain()
application
The entire program
works fine, but there seems to be some problem whenever the call is hangup,
either by pushing # to exit the VoicemailMain() apps or by hanging the phone. If
the # button is push, should Asterisk send something back to tell OPENSER to
hang up the party ?
Here's the log of
verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave' (language 'en') -- Playing 'vm-no'
(language 'en') -- Playing 'vm-messages' (language
'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-goodbye' (language
'en') -- Executing Playback("SIP/210.23.1.139-081ee3d8",
"Goodbye") in new stackFeb 9 15:05:06
WARNING[23242]: file.c:509 ast_openstream_full: File Goodbye does not exist in
any formatFeb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile:
Unable to open Goodbye (format alaw): No such file or directoryFeb
9 15:05:06 WARNING[23242]: app_playback.c:132 playback_exec: ast_streamfile
failed on SIP/203.125.68.66-081ee3d8for Goodbye --
Executing Hangup("SIP/203.125.68.66-081ee3d8", "") in new stack ==
Spawn extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is
this all about ?
Regards,Sam
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Re: [Asterisk-Users] Voicemail Problem
You don't have 'vm-goodbye' voice file. Check under
/var/lib/asterisk/sounds
Wojtek
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 8:38
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday, February 09, 2006 3:30 PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Voicemail Problem
I have just setup
my OPENSER to work with the asterisk 1.2.2.
I've set extension
400 in extension.conf to point to the VoicemailMain()
application
The entire program
works fine, but there seems to be some problem whenever the call is hangup,
either by pushing # to exit the VoicemailMain() apps or by hanging the phone.
If the # button is push, should Asterisk send something back to tell OPENSER
to hang up the party ?
Here's the log of
verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave' (language 'en') -- Playing
'vm-no' (language 'en') -- Playing 'vm-messages'
(language 'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-goodbye' (language
'en') -- Executing Playback("SIP/210.23.1.139-081ee3d8",
"Goodbye") in new stackFeb 9 15:05:06
WARNING[23242]: file.c:509 ast_openstream_full: File Goodbye does not exist in
any formatFeb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile:
Unable to open Goodbye (format alaw): No such file or
directoryFeb 9 15:05:06 WARNING[23242]: app_playback.c:132
playback_exec: ast_streamfile failed on SIP/203.125.68.66-081ee3d8for
Goodbye -- Executing
Hangup("SIP/203.125.68.66-081ee3d8", "") in new stack == Spawn
extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is
this all about ?
Regards,Sam
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RE: [Asterisk-Users] Voicemail Problem
Strange thing that , its there !
[EMAIL PROTECTED]:/home/sam# ls
/var/lib/asterisk/sounds/goodbye.gsm/var/lib/asterisk/sounds/goodbye.gsm
[EMAIL PROTECTED]:/home/sam#
That's why i found it very strange. Thanks for replying.
Are there any other ideas ?
Regards,Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wojciech
TrycSent: Friday, February 10, 2006 9:59 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Voicemail Problem
You don't have 'vm-goodbye' voice file. Check under
/var/lib/asterisk/sounds
Wojtek
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 8:38
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday, February 09, 2006 3:30 PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Voicemail Problem
I have just setup
my OPENSER to work with the asterisk 1.2.2.
I've set extension
400 in extension.conf to point to the VoicemailMain()
application
The entire program
works fine, but there seems to be some problem whenever the call is hangup,
either by pushing # to exit the VoicemailMain() apps or by hanging the phone.
If the # button is push, should Asterisk send something back to tell OPENSER
to hang up the party ?
Here's the log of
verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave' (language 'en') -- Playing
'vm-no' (language 'en') -- Playing 'vm-messages'
(language 'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-goodbye' (language
'en') -- Executing Playback("SIP/210.23.1.139-081ee3d8",
"Goodbye") in new stackFeb 9 15:05:06
WARNING[23242]: file.c:509 ast_openstream_full: File Goodbye does not exist in
any formatFeb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile:
Unable to open Goodbye (format alaw): No such file or
directoryFeb 9 15:05:06 WARNING[23242]: app_playback.c:132
playback_exec: ast_streamfile failed on SIP/203.125.68.66-081ee3d8for
Goodbye -- Executing
Hangup("SIP/203.125.68.66-081ee3d8", "") in new stack == Spawn
extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is
this all about ?
Regards,Sam
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Re: [Asterisk-Users] Voicemail Problem
You are looking for vn-goodbye, most likely under
sounds/vm
W
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 9:21
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Strange thing that , its there !
[EMAIL PROTECTED]:/home/sam# ls
/var/lib/asterisk/sounds/goodbye.gsm/var/lib/asterisk/sounds/goodbye.gsm
[EMAIL PROTECTED]:/home/sam#
That's why i found it very strange. Thanks for replying.
Are there any other ideas ?
Regards,Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wojciech
TrycSent: Friday, February 10, 2006 9:59 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Voicemail Problem
You don't have 'vm-goodbye' voice file. Check
under /var/lib/asterisk/sounds
Wojtek
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 8:38
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday, February 09, 2006 3:30 PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Voicemail Problem
I have just
setup my OPENSER to work with the asterisk 1.2.2.
I've set
extension 400 in extension.conf to point to the VoicemailMain()
application
The entire
program works fine, but there seems to be some problem whenever the call is
hangup, either by pushing # to exit the VoicemailMain() apps or by hanging
the phone. If the # button is push, should Asterisk send something back to
tell OPENSER to hang up the party ?
Here's the log
of verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave'
(language 'en') -- Playing 'vm-no' (language
'en') -- Playing 'vm-messages' (language
'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-goodbye' (language
'en') -- Executing
Playback("SIP/210.23.1.139-081ee3d8", "Goodbye") in new stackFeb 9 15:05:06 WARNING[23242]: file.c:509
ast_openstream_full: File Goodbye does not exist in any formatFeb
9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile: Unable to open Goodbye
(format alaw): No such file or directoryFeb 9 15:05:06
WARNING[23242]: app_playback.c:132 playback_exec: ast_streamfile failed on
SIP/203.125.68.66-081ee3d8for Goodbye -- Executing
Hangup("SIP/203.125.68.66-081ee3d8", "") in new stack == Spawn
extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is
this all about ?
Regards,Sam
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RE: [Asterisk-Users] Voicemail Problem
It is also there ..
[EMAIL PROTECTED]:/home/sam# ls
/var/lib/asterisk/sounds/vm-goodbye.gsm/var/lib/asterisk/sounds/vm-goodbye.gsm[EMAIL PROTECTED]:/home/sam#
Regards,Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wojciech
TrycSent: Friday, February 10, 2006 10:59 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Voicemail Problem
You are looking for vn-goodbye, most likely under
sounds/vm
W
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 9:21
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Strange thing that , its there !
[EMAIL PROTECTED]:/home/sam# ls
/var/lib/asterisk/sounds/goodbye.gsm/var/lib/asterisk/sounds/goodbye.gsm
[EMAIL PROTECTED]:/home/sam#
That's why i found it very strange. Thanks for replying.
Are there any other ideas ?
Regards,Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wojciech
TrycSent: Friday, February 10, 2006 9:59 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Voicemail Problem
You don't have 'vm-goodbye' voice file. Check
under /var/lib/asterisk/sounds
Wojtek
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 8:38
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday, February 09, 2006 3:30 PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Voicemail Problem
I have just
setup my OPENSER to work with the asterisk 1.2.2.
I've set
extension 400 in extension.conf to point to the VoicemailMain()
application
The entire
program works fine, but there seems to be some problem whenever the call is
hangup, either by pushing # to exit the VoicemailMain() apps or by hanging
the phone. If the # button is push, should Asterisk send something back to
tell OPENSER to hang up the party ?
Here's the log
of verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave'
(language 'en') -- Playing 'vm-no' (language
'en') -- Playing 'vm-messages' (language
'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-goodbye' (language
'en') -- Executing
Playback("SIP/210.23.1.139-081ee3d8", "Goodbye") in new stackFeb 9 15:05:06 WARNING[23242]: file.c:509
ast_openstream_full: File Goodbye does not exist in any formatFeb
9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile: Unable to open Goodbye
(format alaw): No such file or directoryFeb 9 15:05:06
WARNING[23242]: app_playback.c:132 playback_exec: ast_streamfile failed on
SIP/203.125.68.66-081ee3d8for Goodbye -- Executing
Hangup("SIP/203.125.68.66-081ee3d8", "") in new stack == Spawn
extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is
this all about ?
Regards,Sam
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RE: [Asterisk-Users] Voicemail Problem
Hack hack hack 8-) Now - comments inline...
> Here's the log of verbose level 3
>
> Asterisk*CLI>
> -- Playing 'vm-youhave' (language 'en')
> -- Playing 'vm-no' (language 'en')
> -- Playing 'vm-messages' (language 'en')
> -- Playing 'vm-opts' (language 'en')
> -- Playing 'vm-goodbye' (language 'en')
Here Asterisk says 'Goodbye'
> -- Executing Playback("SIP/210.23.1.139-081ee3d8",
>"Goodbye") in new stack
Oh! Looky Not Playing but Playback!!!
And it's looking for 'Goodbye' - not vm-goodbye not goodbye
> Feb 9 15:05:06 WARNING[23242]: file.c:509
>ast_openstream_full: File Goodbye does not exist in any format
> Feb 9 15:05:06 WARNING[23242]: file.c:821
>ast_streamfile: Unable to open Goodbye (format alaw): No such file or
>dire
> ctory
> Feb 9 15:05:06 WARNING[23242]: app_playback.c:132
>playback_exec: ast_streamfile failed on SIP/203.125.68.66-081ee3d8
> for Goodbye
> -- Executing Hangup("SIP/203.125.68.66-081ee3d8",
>"") in new stack
> == Spawn extension (default, 400, 3) exited non-zero
>on 'SIP/203.125.68.66-081ee3d8'
> Asterisk*CLI>
So apparently you have a 'h' extension and call 'Goodbye'
like:
exten => 'h',1,Playback(Goodbye);
which it ain't gonna find
Brett
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RE: [Asterisk-Users] Voicemail Problem
Case
sensitivity? The CLI references Goodbye but your filename is goodbye.gsm.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam Lee
Sent: Friday, 10 February 2006
1:22 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Voicemail Problem
Strange thing that , its there !
[EMAIL PROTECTED]:/home/sam#
ls /var/lib/asterisk/sounds/goodbye.gsm
/var/lib/asterisk/sounds/goodbye.gsm
[EMAIL PROTECTED]:/home/sam#
That's why i found it very strange. Thanks
for replying. Are there any other ideas ?
Regards,
Sam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wojciech Tryc
Sent: Friday, February 10, 2006
9:59 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Voicemail Problem
You don't have 'vm-goodbye' voice file. Check under
/var/lib/asterisk/sounds
Wojtek
- Original Message -
From: Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February
09, 2006 8:38 PM
Subject: RE:
[Asterisk-Users] Voicemail Problem
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam Lee
Sent: Thursday, February 09, 2006
3:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Voicemail Problem
I have just setup my OPENSER to work with the asterisk 1.2.2.
I've set extension 400 in extension.conf to point to the
VoicemailMain() application
The entire program works fine, but there seems to be some
problem whenever the call is hangup, either by pushing # to exit the
VoicemailMain() apps or by hanging the phone. If the # button is push, should
Asterisk send something back to tell OPENSER to hang up the party ?
Here's the log of verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-goodbye' (language 'en')
-- Executing Playback("SIP/210.23.1.139-081ee3d8",
"Goodbye") in new stack
Feb 9 15:05:06 WARNING[23242]:
file.c:509 ast_openstream_full: File Goodbye does not exist in any format
Feb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile: Unable to open
Goodbye (format alaw): No such file or dire
ctory
Feb 9 15:05:06 WARNING[23242]: app_playback.c:132 playback_exec:
ast_streamfile failed on SIP/203.125.68.66-081ee3d8
for Goodbye
-- Executing Hangup("SIP/203.125.68.66-081ee3d8",
"") in new stack
== Spawn extension (default, 400, 3) exited non-zero on 'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is this all about ?
Regards,
Sam
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RE: [Asterisk-Users] Voicemail problem
Title: Message
Do you mean
voicemail isn't kicking in after a SIP phone has been called, or it isn't
kicking in when your trying to check voicemail?
Here's my working
voicemail related configs.
sip.conf:
[1000]type=friendusername=1000secret=xxxcallerid=1000[EMAIL PROTECTED]host=dynamiccontext=internalcanreinvite=yesnat=nodtmfmode=rfc2833qualify=yes
extensions.conf:
[globals];
Define global variables
herePHONE1=SIP/2001PHONE1VM=2001
[internal]
exten =>
*5,1,VoicemailMain(${CALLERIDNUM})
vociemail.conf:
[internal]1000
=> 3434223288,1000,[EMAIL PROTECTED]
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Narcis
GRATIANUSent: Sunday, 11 September 2005 12:03 AMTo:
Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] Voicemail
problem
Hello !I am using asterisk at
home 1.5, and i have a really big program. I setup multiple extensions, all
with voicemail feature, but the voicemail does not kick in at all. Any idea
what might be wrong ? I am allowing all possible codecs, but i cannot see what
is possible wrong.Thank you in advance !
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Re: [Asterisk-Users] Voicemail problem
have you added a line for your mailbox in voicemail.conf as well? reloaded app_voicemail.so(in *CLI)? if that isn't the problem, please post your extensions.conf regards christian On Sat, 10 Sep 2005 17:03:01 +0300 Narcis GRATIANU <[EMAIL PROTECTED]> wrote: > Hello ! > > I am using asterisk at home 1.5, and i have a really big program. I setup > multiple extensions, all with voicemail feature, but the voicemail does not > kick in at all. Any idea what might be wrong ? I am allowing all possible > codecs, but i cannot see what is possible wrong. > > Thank you in advance ! > ___ --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
Re: [Asterisk-Users] Voicemail problem
Hi Daniel can you give us more information so that it would be easy to debug.like voice mail configuration etc Thanks,GIridhar Bandi.On 4/18/06, Daniel Korndorfer <[EMAIL PROTECTED]> wrote: Hi,when I call the voicemail app, it starts and die suddenly. Has anyonealready had this problem?Log:app.c:644 ast_play_and_record: No audio available on SIP/-6fca??-- User hung upTks, D.K.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 Problem or bug?
Ok I have a question about the voicemail program with Asterisk. This is with the current head CVS as of 7/28/04 and every other one before it. When apending to a message that you forward, to stop recording you press any key. But it take however long you record for it to save the message then return to a menu. If your add 2 minutes of recording it takes that long to return to you. If you press the # key it will delete the message and your out of there. This is a major problem with people how use cell phones. You get dead air while your waiting. Is there a fix for this. Does someone have a better voicemail program than this one. - \ \\_ Ariel Batista // / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 problem, not playing back audio
Hey all, am running into a problem with * 1.2.1 recently. When we leave a voicemail for someone, occasionally when they check the vm, * doesn't play back the message that was recorded. I can see the vm audio file on the server, but when it's checked from a phone, it just skips to the end of the session asking if you'd like to delete the message, even though it never played. Any ideas what might be causing this? I've deleted & recreated the VM boxes, but the issue continues across the company, any pointers on where to look to track this down would be greatly appreciated. Thx in advance Dan ___ --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
Re: [Asterisk-Users] Voicemail problem, not playing back audio
Dan Elder wrote: Hey all, am running into a problem with * 1.2.1 recently. When we leave a voicemail for someone, occasionally when they check the vm, * doesn't play back the message that was recorded. I can see the vm audio file on the server, but when it's checked from a phone, it just skips to the end of the Maybe the audio file is damaged? Doug ___ --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
