Re: [asterisk-users] pulling my hair out over voicemail
Don't forget to 1000,1,Answer the call Moj John Von Essen wrote: > 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 > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pulling my hair out over voicemail
Ok, I have made some progress debugging this. I dont believe it has
anything to do with asterisk or my phone. Rather I think it is an
issues with STUN and/or my Linksys router at home.
The phones I am testing all sit behind a NAT'd firewall, your basic
Linksys router for the Home DSL user.
The phones all of STUN setup, and the STUN server IP is the IP of the
asterisk server - which is purely public.
I was able to duplicate the problem with not being able to hear the
voicemail greeting by doing the following:
Turn off all the phones, and power cycle my Linksys, then turn on 1
phone. That one phone will then work, and you can hear voicemail
greeting.
The I turn on the second phone. Then voicemail greeting breaks, and you
cant hear it when you dial into voicemail. If I unplug the first phone,
and power cycle the Linksys again, the second phone will begin to work.
So the question is, does this behavior make sense?
I assumed with an STUN server I could have multiple phones behind my
Linksys firewall, now it appears I can only have one. Is it a Linksys
bug, or a general known issue? Do I need to run multiple STUN servers?
Thanks
John
On Jan 31, 2008, at 1:00 PM, Shane D wrote:
> Very odd. Could you try taking the mailbox line out of sip.conf and
> see what happens?
>
> On 1/31/08, John Von Essen <[EMAIL PROTECTED]> wrote:
>> 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 [EMAIL PROTECTED]:1] Ringing("SIP/6001-081d65c8", "") in
>> new stack
>> -- Executing [EMAIL PROTECTED]:2] Wait("SIP/6001-081d65c8", "2") in
>> new
>> stack
>> -- Executing [EMAIL PROTECTED]:3] VoiceMailMain("SIP/6001-081d65c8",
>> "[EMAIL PROTECTED]") in new stack
>> -- 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 '[EMAIL PROTECTED]' 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 <[EMAIL PROTECTED]> 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
>>>
>>
>>
>> ___
>> -- 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
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] pulling my hair out over voicemail
John Von Essen wrote:
> 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 [EMAIL PROTECTED]:1] Ringing("SIP/6001-081d65c8", "") in
> new stack
> -- Executing [EMAIL PROTECTED]:2] Wait("SIP/6001-081d65c8", "2") in new
> stack
> -- Executing [EMAIL PROTECTED]:3] VoiceMailMain("SIP/6001-081d65c8",
> "[EMAIL PROTECTED]") in new stack
> -- 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 '[EMAIL PROTECTED]' 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 <[EMAIL PROTECTED]>
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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Re: [asterisk-users] pulling my hair out over voicemail
John Von Essen wrote: > 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 [EMAIL PROTECTED] Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pulling my hair out over voicemail
Very odd. Could you try taking the mailbox line out of sip.conf and
see what happens?
On 1/31/08, John Von Essen <[EMAIL PROTECTED]> wrote:
> 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 [EMAIL PROTECTED]:1] Ringing("SIP/6001-081d65c8", "") in
> new stack
> -- Executing [EMAIL PROTECTED]:2] Wait("SIP/6001-081d65c8", "2") in new
> stack
> -- Executing [EMAIL PROTECTED]:3] VoiceMailMain("SIP/6001-081d65c8",
> "[EMAIL PROTECTED]") in new stack
> -- 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 '[EMAIL PROTECTED]' 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 <[EMAIL PROTECTED]> 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
> >
>
>
> ___
> -- 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
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Re: [asterisk-users] pulling my hair out over voicemail
Okay, What I ment was you don't have to. On 1/31/08, John Millican <[EMAIL PROTECTED]> wrote: > 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 <[EMAIL PROTECTED]> 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([EMAIL PROTECTED]|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 > > > ___ > -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pulling my hair out over voicemail
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 [EMAIL PROTECTED]:1] Ringing("SIP/6001-081d65c8", "") in
new stack
-- Executing [EMAIL PROTECTED]:2] Wait("SIP/6001-081d65c8", "2") in new
stack
-- Executing [EMAIL PROTECTED]:3] VoiceMailMain("SIP/6001-081d65c8",
"[EMAIL PROTECTED]") in new stack
-- 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 '[EMAIL PROTECTED]' 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 <[EMAIL PROTECTED]> 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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Re: [asterisk-users] pulling my hair out over voicemail
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 <[EMAIL PROTECTED]> 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([EMAIL PROTECTED]|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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pulling my hair out over voicemail
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 <[EMAIL PROTECTED]> 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 > > > ___ > -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pulling my hair out over voicemail
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pulling my hair out over voicemail
On Jan 31, 2008 12:30 AM, John Von Essen <[EMAIL PROTECTED]> 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
Re: [asterisk-users] pulling my hair out over voicemail
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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
[EMAIL PROTECTED]: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
[EMAIL PROTECTED]: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
[EMAIL PROTECTED]:3] VoiceMailMain("SIP/6001-081de7a8", "[EMAIL PROTECTED]") 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: --
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
'[EMAIL PROTECTED]' 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: --
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
'[EMAIL PROTECTED]'
[Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Destroying SIP dialog
[EMAIL PROTECTED]
[Jan 30 21:26:53] VERBOSE[7893] logger.c: Really destroying SIP dialog
'[EMAIL PROTECTED]' Method: REGISTER
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (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: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len =
4)
[Jan 30 21:26
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
[EMAIL PROTECTED]: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
[EMAIL PROTECTED]: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
[EMAIL PROTECTED]:3] VoiceMailMain("SIP/6001-081de7a8", "[EMAIL PROTECTED]") 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: --
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
'[EMAIL PROTECTED]' 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: --
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
'[EMAIL PROTECTED]'
[Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Destroying SIP dialog
[EMAIL PROTECTED]
[Jan 30 21:26:53] VERBOSE[7893] logger.c: Really destroying SIP dialog
'[EMAIL PROTECTED]' Method: REGISTER
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (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: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (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: 0005 (len =
4)
[Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len =
4)
[Jan 30 21:26:57] DEBUG[7893] chan_sip.c: Setting SIP_ALREADYGONE on
dialog [EMAIL PROTECTED]
[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] p
Re: [asterisk-users] pulling my hair out over voicemail
John Von Essen wrote: > 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 > > > ___ > -- 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 > I would suggest showing us the extensions configs for both phones :). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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. 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pulling my hair out over voicemail
You might need Voicemailmain([EMAIL PROTECTED]) PaulH On Thu, 2008-01-31 at 00:30 -0500, John Von Essen wrote: > 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 > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 ___ -- 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
