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john at quonix.net
Guest
PostPosted: Thu Jan 31, 2008 12:30 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with 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.

I created an extension to retrieve the messages:

exten => 1000,1,Ringing
exten => 1000,2,Wait(2)
exten => 1000,3,VoicemailMain

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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pdhales at optusnet.co...
Guest
PostPosted: Thu Jan 31, 2008 12:41 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
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.

I created an extension to retrieve the messages:

exten => 1000,1,Ringing
exten => 1000,2,Wait(2)
exten => 1000,3,VoicemailMain

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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To UNSUBSCRIBE or update options visit:
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ddunkin at netos.net
Guest
PostPosted: Thu Jan 31, 2008 12:42 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
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 at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John Von
Essen
Sent: Wednesday, January 30, 2008 21:30
To: asterisk-users at 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.

I created an extension to retrieve the messages:

exten => 1000,1,Ringing
exten => 1000,2,Wait(2)
exten => 1000,3,VoicemailMain

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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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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anthonyf at rockynet.com
Guest
PostPosted: Thu Jan 31, 2008 1:22 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
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.

I created an extension to retrieve the messages:

exten => 1000,1,Ringing
exten => 1000,2,Wait(2)
exten => 1000,3,VoicemailMain

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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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

I would suggest showing us the extensions configs for both phones Smile.
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john at quonix.net
Guest
PostPosted: Thu Jan 31, 2008 1:35 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
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 at 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 at 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 at default:3] VoiceMailMain("SIP/6001-081de7a8", "1000 at 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 at 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 at 208.82.128.10'
[Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Destroying SIP dialog
76c69e4258ca84fe7837717768a08e8c at 208.82.128.10
[Jan 30 21:26:53] VERBOSE[7893] logger.c: Really destroying SIP dialog
'76c69e4258ca84fe7837717768a08e8c at 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 at 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 at 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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ddunkin at netos.net
Guest
PostPosted: Thu Jan 31, 2008 2:31 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
How about your sip.conf for your extensions?

Example:
[6001]
host=dynamic
type=friend
disallow=all
allow=ulaw

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 at lists.digium.com
[mailto:asterisk-users-bounces at 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 at 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 at 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 at default:3] VoiceMailMain("SIP/6001-081de7a8", "1000 at 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 at 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 at 208.82.128.10'
[Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Destroying SIP dialog
76c69e4258ca84fe7837717768a08e8c at 208.82.128.10
[Jan 30 21:26:53] VERBOSE[7893] logger.c: Really destroying SIP dialog
'76c69e4258ca84fe7837717768a08e8c at 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 at 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 at 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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joakimsen at gmail.com
Guest
PostPosted: Thu Jan 31, 2008 3:00 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
On Jan 31, 2008 12:30 AM, John Von Essen <john at quonix.net> wrote:
Quote:

Any ideas what could be going on? I tried tweaking the extension 1000
so it looks like:

Maybe the SIP config is wrong?

Quote:

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?

Quote:
Please help. This is driving me nuts. I even tried re-installing
asterisk from scratch - no change.

What version?
Back to top
drew at oanda.com
Guest
PostPosted: Thu Jan 31, 2008 9:47 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
John Von Essen wrote:
Quote:
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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chatter8712 at gmail.com
Guest
PostPosted: Thu Jan 31, 2008 10:19 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
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 at oanda.com> wrote:
Quote:
John Von Essen wrote:
Quote:
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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Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
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jmillican at sentinelc...
Guest
PostPosted: Thu Jan 31, 2008 10:51 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
Shane D wrote:
Quote:
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 at oanda.com> wrote:
Quote:
John Von Essen wrote:
Quote:
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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john at quonix.net
Guest
PostPosted: Thu Jan 31, 2008 10:59 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
Here are my configs:
sip.conf:

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw

[6000]
type=friend
secret=letmein
host=dynamic
dtmfmode=rfc2833
mailbox=6000
context=default

extensions.conf:

[default]
exten => 1000,1,Ringing
exten => 1000,2,Wait(2)
exten => 1000,3,VoicemailMain

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 at default:1] Ringing("SIP/6001-081d65c8", "") in
new stack
-- Executing [1000 at default:2] Wait("SIP/6001-081d65c8", "2") in new
stack
-- Executing [1000 at default:3] VoiceMailMain("SIP/6001-081d65c8",
"1000 at 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 at 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 at quonix.net> wrote:
Quote:

Any ideas what could be going on? I tried tweaking the extension 1000
so it looks like:

Maybe the SIP config is wrong?

Quote:

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?

Quote:
Please help. This is driving me nuts. I even tried re-installing
asterisk from scratch - no change.

What version?

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chatter8712 at gmail.com
Guest
PostPosted: Thu Jan 31, 2008 11:01 am    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
Okay, What I ment was you don't have to.

On 1/31/08, John Millican <jmillican at sentinelcommunications.com> wrote:
Quote:
Shane D wrote:
Quote:
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 at oanda.com> wrote:
Quote:
John Von Essen wrote:
Quote:
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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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
Back to top
chatter8712 at gmail.com
Guest
PostPosted: Thu Jan 31, 2008 1:00 pm    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
Very odd. Could you try taking the mailbox line out of sip.conf and
see what happens?

On 1/31/08, John Von Essen <john at quonix.net> wrote:
Quote:
Here are my configs:


sip.conf:

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw

[6000]
type=friend
secret=letmein
host=dynamic
dtmfmode=rfc2833
mailbox=6000
context=default

extensions.conf:

[default]
exten => 1000,1,Ringing
exten => 1000,2,Wait(2)
exten => 1000,3,VoicemailMain

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 at default:1] Ringing("SIP/6001-081d65c8", "") in
new stack
-- Executing [1000 at default:2] Wait("SIP/6001-081d65c8", "2") in new
stack
-- Executing [1000 at default:3] VoiceMailMain("SIP/6001-081d65c8",
"1000 at 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 at 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 at quonix.net> wrote:
Quote:

Any ideas what could be going on? I tried tweaking the extension 1000
so it looks like:

Maybe the SIP config is wrong?

Quote:

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?

Quote:
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



_______________________________________________
-- 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

--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
Back to top
support at drdos.info
Guest
PostPosted: Thu Jan 31, 2008 1:16 pm    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
John Von Essen wrote:
Quote:
Here are my configs:



[6000]
type=friend
secret=letmein
host=dynamic
dtmfmode=rfc2833
mailbox=6000


I believe you need to include a context on your mailbox line, such as
mailbox=4121 at sip

Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
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elam at officegeneral.com
Guest
PostPosted: Thu Jan 31, 2008 2:12 pm    Post subject: [asterisk-users] pulling my hair out over voicemail Reply with quote
John Von Essen wrote:
Quote:
Here are my configs:


sip.conf:

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw

[6000]
type=friend
secret=letmein
host=dynamic
dtmfmode=rfc2833
mailbox=6000
context=default

extensions.conf:

[default]
exten => 1000,1,Ringing
exten => 1000,2,Wait(2)
exten => 1000,3,VoicemailMain

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 at default:1] Ringing("SIP/6001-081d65c8", "") in
new stack
-- Executing [1000 at default:2] Wait("SIP/6001-081d65c8", "2") in new
stack
-- Executing [1000 at default:3] VoiceMailMain("SIP/6001-081d65c8",
"1000 at 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 at 192.168.1.112' Method:
BYE

So it plays the greetings, and is working, I just cant hear it.

what's your voicemail.conf looks like?
also check the file permission and make sure asterisk can read it.
--
Edwin Lam <edwin.lam at officegeneral.com>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
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