Posted: Thu Jan 31, 2008 5:34 am Post subject: [asterisk-users] pulling my hair out over voicemail
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
addition being by SIP phone, and my incoming voip. Last week I got
everything setup, calls were working, etc.,.
I cam across a tutorial for voicemail, followed it, and it worked. When
I call my phone and dont answer, it goes to voicemail, and message is
stored on server.
And that worked. Granted, everything is still defaults, so when I dial
1000, I get the "Comedian Mail" greeting, then it prompts for mailbox
and password, then I get the menu.
Now, here is how it gets weird. Today I go and setup a new second SIP
phone, and proceed to set it up for voicemail. Inbound calls go to
voicemail properly when nobody answers, but I cant retrieve the
messages.
When I dial extension 1000, its rings for 2 seconds, then just goes
silent. No greeting, no mailbox prompts, nothing.
Any ideas what could be going on? I tried tweaking the extension 1000
so it looks like:
exten => 1000,3,VoicemailMain,s6000
Where 6000 is my mailbox. But still nothing, when I dial 1000, it just
goes silent.
Please help. This is driving me nuts. I even tried re-installing
asterisk from scratch - no change.
-john
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Posted: Thu Jan 31, 2008 5:48 am Post subject: [asterisk-users] pulling my hair out over voicemail
You might need Voicemailmain(${EXTEN}@default)
PaulH
On Thu, 2008-01-31 at 00:30 -0500, John Von Essen wrote:
Quote:
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
addition being by SIP phone, and my incoming voip. Last week I got
everything setup, calls were working, etc.,.
I cam across a tutorial for voicemail, followed it, and it worked. When
I call my phone and dont answer, it goes to voicemail, and message is
stored on server.
And that worked. Granted, everything is still defaults, so when I dial
1000, I get the "Comedian Mail" greeting, then it prompts for mailbox
and password, then I get the menu.
Now, here is how it gets weird. Today I go and setup a new second SIP
phone, and proceed to set it up for voicemail. Inbound calls go to
voicemail properly when nobody answers, but I cant retrieve the
messages.
When I dial extension 1000, its rings for 2 seconds, then just goes
silent. No greeting, no mailbox prompts, nothing.
Any ideas what could be going on? I tried tweaking the extension 1000
so it looks like:
exten => 1000,3,VoicemailMain,s6000
Where 6000 is my mailbox. But still nothing, when I dial 1000, it just
goes silent.
Please help. This is driving me nuts. I even tried re-installing
asterisk from scratch - no change.
-john
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Posted: Thu Jan 31, 2008 5:49 am Post subject: [asterisk-users] pulling my hair out over voicemail
Can you get some verbose output from your console/logs? It may be more
obvious once you see what Asterisk is attempting to do when this
extension is dialed.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John Von
Essen
Sent: Wednesday, January 30, 2008 21:30
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] pulling my hair out over voicemail
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
addition being by SIP phone, and my incoming voip. Last week I got
everything setup, calls were working, etc.,.
I cam across a tutorial for voicemail, followed it, and it worked. When
I call my phone and dont answer, it goes to voicemail, and message is
stored on server.
And that worked. Granted, everything is still defaults, so when I dial
1000, I get the "Comedian Mail" greeting, then it prompts for mailbox
and password, then I get the menu.
Now, here is how it gets weird. Today I go and setup a new second SIP
phone, and proceed to set it up for voicemail. Inbound calls go to
voicemail properly when nobody answers, but I cant retrieve the
messages.
When I dial extension 1000, its rings for 2 seconds, then just goes
silent. No greeting, no mailbox prompts, nothing.
Any ideas what could be going on? I tried tweaking the extension 1000 so
it looks like:
exten => 1000,3,VoicemailMain,s6000
Where 6000 is my mailbox. But still nothing, when I dial 1000, it just
goes silent.
Please help. This is driving me nuts. I even tried re-installing
asterisk from scratch - no change.
-john
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Posted: Thu Jan 31, 2008 6:28 am Post subject: [asterisk-users] pulling my hair out over voicemail
John Von Essen wrote:
Quote:
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
addition being by SIP phone, and my incoming voip. Last week I got
everything setup, calls were working, etc.,.
I cam across a tutorial for voicemail, followed it, and it worked. When
I call my phone and dont answer, it goes to voicemail, and message is
stored on server.
And that worked. Granted, everything is still defaults, so when I dial
1000, I get the "Comedian Mail" greeting, then it prompts for mailbox
and password, then I get the menu.
Now, here is how it gets weird. Today I go and setup a new second SIP
phone, and proceed to set it up for voicemail. Inbound calls go to
voicemail properly when nobody answers, but I cant retrieve the
messages.
When I dial extension 1000, its rings for 2 seconds, then just goes
silent. No greeting, no mailbox prompts, nothing.
Any ideas what could be going on? I tried tweaking the extension 1000
so it looks like:
exten => 1000,3,VoicemailMain,s6000
Where 6000 is my mailbox. But still nothing, when I dial 1000, it just
goes silent.
Please help. This is driving me nuts. I even tried re-installing
asterisk from scratch - no change.
-john
_______________________________________________
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I usually don't see this (I'm more production and haven't done heavy
debug for a long time):
[Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8
to write format gsm
[Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8
to write format ulaw
Since it's within the same second, I'm not sure which is actually being
set.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John Von
Essen
Sent: Wednesday, January 30, 2008 22:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] pulling my hair out over voicemail
Tried it, but no change.
A few updates. Even though I dont hear anything, if I hit a keys on the
phone and then hang up, message log says:
[Jan 30 21:26:57] WARNING[7917] app_voicemail.c: Unable to read password
I enabled logging of everything, and the below is the snippet for when
my SIP/6001 phone dial extension 1000 for Voicemail:
[Jan 30 21:26:35] DEBUG[7917] pbx.c: Launching 'Ringing'
[Jan 30 21:26:35] VERBOSE[7917] logger.c: -- Executing
[1000@default:1] Ringing("SIP/6001-081de7a8", "") in new stack
[Jan 30 21:26:35] DEBUG[7917] pbx.c: Launching 'Wait'
[Jan 30 21:26:35] VERBOSE[7917] logger.c: -- Executing
[1000@default:2] Wait("SIP/6001-081de7a8", "2") in new stack
[Jan 30 21:26:37] DEBUG[7917] pbx.c: Launching 'VoiceMailMain'
[Jan 30 21:26:37] VERBOSE[7917] logger.c: -- Executing
[1000@default:3] VoiceMailMain("SIP/6001-081de7a8", "1000@default") in
new stack
[Jan 30 21:26:37] DEBUG[7917] app_voicemail.c: Before ast_answer
[Jan 30 21:26:37] DEBUG[7917] devicestate.c: Notification of state
change to be queued on device/channel SIP/6001-081de7a8
[Jan 30 21:26:37] DEBUG[7890] chan_sip.c: Checking device state for
peer 6001
[Jan 30 21:26:37] DEBUG[7890] devicestate.c: Changing state for
SIP/6001 - state 5 (Unavailable)
[Jan 30 21:26:37] DEBUG[7890] chan_sip.c: Checking device state for
peer 6001
[Jan 30 21:26:37] DEBUG[7917] chan_sip.c: SIP answering channel:
SIP/6001-081de7a8
[Jan 30 21:26:37] DEBUG[7917] chan_sip.c: Setting framing from config
on incoming call
[Jan 30 21:26:37] DEBUG[7917] chan_sip.c: ** Our capability: 0x4 (ulaw)
Video flag: True
[Jan 30 21:26:37] DEBUG[7917] chan_sip.c: ** Our prefcodec: 0x0
(nothing)
[Jan 30 21:26:37] DEBUG[7917] chan_sip.c: -- Done with adding codecs to
SDP
[Jan 30 21:26:37] DEBUG[7917] chan_sip.c: Done building SDP. Settling
with this capability: 0x4 (ulaw)
[Jan 30 21:26:37] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8
to write format gsm
[Jan 30 21:26:37] DEBUG[7917] rtp.c: Ooh, format changed from unknown
to ulaw
[Jan 30 21:26:37] DEBUG[7917] rtp.c: Created smoother: format: 4 ms: 20
len: 160
[Jan 30 21:26:37] VERBOSE[7917] logger.c: -- <SIP/6001-081de7a8>
Playing 'vm-login' (language 'en')
[Jan 30 21:26:37] DEBUG[7910] app_queue.c: Device 'SIP/6001' changed to
state '5' (Unavailable) but we don't care because they're not a member
of any queue.
[Jan 30 21:26:37] DEBUG[7893] chan_sip.c: Stopping retransmission on
'3735eef706fa0b2a@192.168.1.112' of Response 12349: Match Not Found
[Jan 30 21:26:39] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8
to write format ulaw
[Jan 30 21:26:50] DEBUG[7917] app_voicemail.c: Before find user for
mailbox 8563682102
[Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8
to write format gsm
[Jan 30 21:26:50] DEBUG[7917] rtp.c: Difference is 82416, ms is 10322
[Jan 30 21:26:50] VERBOSE[7917] logger.c: -- <SIP/6001-081de7a8>
Playing 'vm-password' (language 'en')
[Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8
to write format ulaw
[Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Auto destroying SIP dialog
'76c69e4258ca84fe7837717768a08e8c@208.82.128.10'
[Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Destroying SIP dialog
76c69e4258ca84fe7837717768a08e8c@208.82.128.10
[Jan 30 21:26:53] VERBOSE[7893] logger.c: Really destroying SIP dialog
'76c69e4258ca84fe7837717768a08e8c@208.82.128.10' Method: REGISTER
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 00000005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: Sending dtmf: 53 (5), at
76.161.192.192
[Jan 30 21:26:55] DTMF[7917] channel.c: DTMF begin '5' received on
SIP/6001-081de7a8
[Jan 30 21:26:55] DTMF[7917] channel.c: DTMF begin ignored '5' on
SIP/6001-081de7a8
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 00000005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 00000005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 00000005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 00000005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 00000005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 00000005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 00000005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 00000005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: Sending dtmf: 53 (5), at
76.161.192.192
[Jan 30 21:26:55] DTMF[7917] channel.c: DTMF end '5' received on
SIP/6001-081de7a8, duration 120 ms
[Jan 30 21:26:55] DTMF[7917] channel.c: DTMF end passthrough '5' on
SIP/6001-081de7a8
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 00000005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 00000005 (len =
4)
[Jan 30 21:26:57] DEBUG[7893] chan_sip.c: Setting SIP_ALREADYGONE on
dialog 3735eef706fa0b2a@192.168.1.112
[Jan 30 21:26:57] DEBUG[7893] chan_sip.c: Received bye, issuing owner
hangup
[Jan 30 21:26:57] WARNING[7917] app_voicemail.c: Unable to read password
[Jan 30 21:26:57] DEBUG[7917] app_voicemail.c: After vm_authenticate
[Jan 30 21:26:57] DEBUG[7917] pbx.c: Extension 1000, priority 3
returned normally even though call was hung up
[Jan 30 21:26:57] DEBUG[7917] channel.c: Soft-Hanging up channel
'SIP/6001-081de7a8'
[Jan 30 21:26:57] DEBUG[7917] channel.c: Hanging up channel
'SIP/6001-081de7a8'
[Jan 30 21:26:57] DEBUG[7917] chan_sip.c: Hangup call
SIP/6001-081de7a8, SIP callid 3735eef706fa0b2a@192.168.1.112)
[Jan 30 21:26:57] DEBUG[7917] devicestate.c: Notification of state
change to be queued on device/channel SIP/6001-081de7a8
On Jan 31, 2008, at 12:41 AM, Paul Hales wrote:
Quote:
(${EXTEN}@default)
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You do not specify the mailbox number in the call to the application.
You only specify the number to VoiceMail()
HTH,
Shane
On 1/31/08, Drew Gibson <drew@oanda.com> wrote:
Quote:
John Von Essen wrote:
> Any ideas what could be going on? I tried tweaking the extension 1000
> so it looks like:
>
> exten => 1000,3,VoicemailMain,s6000
>
>
It may be your syntax, try :-
exten => 1000,3,VoicemailMain(6000|s)
regards,
Drew
--
Drew Gibson
Systems Administrator
OANDA Corporation
www.oanda.com
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You do not specify the mailbox number in the call to the application.
You only specify the number to VoiceMail()
HTH,
Shane
On 1/31/08, Drew Gibson <drew@oanda.com> wrote:
> John Von Essen wrote:
>> Any ideas what could be going on? I tried tweaking the extension 1000
>> so it looks like:
>>
>> exten => 1000,3,VoicemailMain,s6000
>>
>>
> It may be your syntax, try :-
>
> exten => 1000,3,VoicemailMain(6000|s)
>
>
> regards,
>
> Drew
>
>
> --
> Drew Gibson
>
> Systems Administrator
> OANDA Corporation
> www.oanda.com
What do you mean you do not use the mailbox in Voicemailmain see below:
*CLI>
-= Info about application 'VoiceMailMain' =-
[Synopsis]
Check Voicemail messages
[Description]
VoiceMailMain([mailbox][@context][|options]): This application allows the
calling party to check voicemail messages. A specific mailbox, and optional
corresponding context, may be specified. If a mailbox is not provided, the
calling party will be prompted to enter one. If a context is not specified,
the 'default' context will be used.
Options:
p - Consider the mailbox parameter as a prefix to the mailbox that
is entered by the caller.
g(#) - Use the specified amount of gain when recording a voicemail
message. The units are whole-number decibels (dB).
s - Skip checking the passcode for the mailbox.
a(#) - Skip folder prompt and go directly to folder specified.
Defaults to INBOX
JohnM
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Calling from phone to phone is fine, and inbound and outbound calling
is fine. But when I call voicemail, I dont hear anything.
When I view console in CLI I see this when attempting to dial the
voicemail extension:
-- Executing [1000@default:1] Ringing("SIP/6001-081d65c8", "") in
new stack
-- Executing [1000@default:2] Wait("SIP/6001-081d65c8", "2") in new
stack
-- Executing [1000@default:3] VoiceMailMain("SIP/6001-081d65c8",
"1000@default") in new stack
-- <SIP/6001-081d65c8> Playing 'vm-login' (language 'en')
[Jan 31 06:42:49] WARNING[8513]: app_voicemail.c:6281 vm_authenticate:
Couldn't read username
Really destroying SIP dialog 'b4c0564313527d89@192.168.1.112' Method:
BYE
So it plays the greetings, and is working, I just cant hear it.
-john
On Jan 31, 2008, at 3:00 AM, Andrew Joakimsen wrote:
Quote:
On Jan 31, 2008 12:30 AM, John Von Essen <john@quonix.net> wrote:
>
> Any ideas what could be going on? I tried tweaking the extension 1000
> so it looks like:
Maybe the SIP config is wrong?
>
> Where 6000 is my mailbox. But still nothing, when I dial 1000, it just
> goes silent.
Can you places other calls from that new phone?
> Please help. This is driving me nuts. I even tried re-installing
> asterisk from scratch - no change.
What version?
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Posted: Thu Jan 31, 2008 4:08 pm Post subject: [asterisk-users] pulling my hair out over voicemail
Okay, What I ment was you don't have to.
On 1/31/08, John Millican <jmillican@sentinelcommunications.com> wrote:
Quote:
Shane D wrote:
> Try this:
> exten => 1000,1,Answer()
> exten => 1000,2,Wait(2)
> exten => 1000,3,VoiceMailMain()
>
> You do not specify the mailbox number in the call to the application.
> You only specify the number to VoiceMail()
>
> HTH,
> Shane
>
> On 1/31/08, Drew Gibson <drew@oanda.com> wrote:
>> John Von Essen wrote:
>>> Any ideas what could be going on? I tried tweaking the extension 1000
>>> so it looks like:
>>>
>>> exten => 1000,3,VoicemailMain,s6000
>>>
>>>
>> It may be your syntax, try :-
>>
>> exten => 1000,3,VoicemailMain(6000|s)
>>
>>
>> regards,
>>
>> Drew
>>
>>
>> --
>> Drew Gibson
>>
>> Systems Administrator
>> OANDA Corporation
>> www.oanda.com
What do you mean you do not use the mailbox in Voicemailmain see below:
*CLI>
-= Info about application 'VoiceMailMain' =-
[Synopsis]
Check Voicemail messages
[Description]
VoiceMailMain([mailbox][@context][|options]): This application allows the
calling party to check voicemail messages. A specific mailbox, and optional
corresponding context, may be specified. If a mailbox is not provided, the
calling party will be prompted to enter one. If a context is not specified,
the 'default' context will be used.
Options:
p - Consider the mailbox parameter as a prefix to the mailbox that
is entered by the caller.
g(#) - Use the specified amount of gain when recording a voicemail
message. The units are whole-number decibels (dB).
s - Skip checking the passcode for the mailbox.
a(#) - Skip folder prompt and go directly to folder specified.
Defaults to INBOX
JohnM
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Calling from phone to phone is fine, and inbound and outbound calling
is fine. But when I call voicemail, I dont hear anything.
When I view console in CLI I see this when attempting to dial the
voicemail extension:
-- Executing [1000@default:1] Ringing("SIP/6001-081d65c8", "") in
new stack
-- Executing [1000@default:2] Wait("SIP/6001-081d65c8", "2") in new
stack
-- Executing [1000@default:3] VoiceMailMain("SIP/6001-081d65c8",
"1000@default") in new stack
-- <SIP/6001-081d65c8> Playing 'vm-login' (language 'en')
[Jan 31 06:42:49] WARNING[8513]: app_voicemail.c:6281 vm_authenticate:
Couldn't read username
Really destroying SIP dialog 'b4c0564313527d89@192.168.1.112' Method:
BYE
So it plays the greetings, and is working, I just cant hear it.
-john
On Jan 31, 2008, at 3:00 AM, Andrew Joakimsen wrote:
> On Jan 31, 2008 12:30 AM, John Von Essen <john@quonix.net> wrote:
>>
>> Any ideas what could be going on? I tried tweaking the extension 1000
>> so it looks like:
>
> Maybe the SIP config is wrong?
>
>>
>> Where 6000 is my mailbox. But still nothing, when I dial 1000, it just
>> goes silent.
>
> Can you places other calls from that new phone?
>
>> Please help. This is driving me nuts. I even tried re-installing
>> asterisk from scratch - no change.
>
> What version?
>
> _______________________________________________
> -- 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
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