[asterisk-users] RT voicemail greetings not played
Hi all. I've got an odd situation with my RT asterisk server. I've got a number of users who are reporting that their voicemail greeting isn't being played anymore. This used to work before a recent asterisk restart. The dialplan is in AGI, so it wasn't changed. I'm storing voicemail in a mysql database and that is working properly. It's just the greeting message that isn't working properly. And, there are not "file not found" type errors on the console with verbose=25. Any ideas as to where I should look? -- Mike Diehl -- _ -- 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] RT voicemail greetings not played
Hi all. I've got an odd situation with my RT asterisk server. I've got a number of users who are reporting that their voicemail greeting isn't being played anymore. This used to work before a recent asterisk restart. The dialplan is in AGI, so it wasn't changed. I'm storing voicemail in a mysql database and that is working properly. It's just the greeting message that isn't working properly. And, there are not "file not found" type errors on the console with verbose=25. Any ideas as to where I should look? -- Mike -- _ -- 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] High resident memory with 11.14.0 ?
On Wed, Nov 26, 2014 at 3:20 PM, James Lamanna wrote:
>
> On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna
> wrote:
>
>>
>> On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan
>> wrote:
>>
>>> On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna
>>> wrote:
>>> > Also, how big does the cache in frame.c grow to?
>>> > I've recompiled with MALLOC_DEBUG on that server:
>>> >
>>> > asterisk -rx "memory show summary"
>>> >
>>> >
>>> > 1780466242 bytes (1780181594 cache) in2352909 allocations in file
>>> > frame.c
>>> > ...
>>> >
>>> > Seems like a ridiculous cache.
>>> >
>>>
>>> I'm not going to respond to your new thread, since it is the same
>>> discussion as this one.
>>>
>>> The frame cache is a per-thread local cache of frames that prevents
>>> having to re-allocate frames as they pass through Asterisk. Clearly,
>>> something is abusing it.
>>>
>>> I think you'll need to provide some more information on how you're
>>> producing this situation. Specifically:
>>> * Channel technologies involved, and the formats on the channels
>>> * Dialplan that reproduces the problem
>>>
>>> Are you using any non-core dialplan applications or channel drivers?
>>>
>>>
>> This PBX has about 100 registered SIP clients, along with 23 PRI
>> channels, 2 inbound/outbound SIP trunks and around 100 IAXModems registered
>> to it. It primarily handles faxing.
>> I am not using any non-standard channel drivers. I am using the T.38
>> gateway funcionality.
>>
>> The jist of the dialplan is this: (example of the PRI and a SIP trunk,
>> inbound)
>>
>> [pri-in]
>> exten => _X.,1,Set(__FROM_DID=${EXTEN})
>> exten => _X.,n,Set(FAX_IDX=700)
>> exten => _X.,n,Set(MAX_IDX=719)
>> exten => _X.,n,Goto(dial-hylafax,s,1)
>>
>> [sip-trunk-in]
>> exten => _X.,1(normal),Set(__FROM_DID=${EXTEN})
>> exten => _X.,n,Set(FAX_IDX=950)
>> exten => _X.,n,Set(MAX_IDX=959)
>> exten => _X.,n,Set(FAXOPT(gateway)=yes)
>> exten => _X.,n,Goto(dial-hylafax,s,1)
>>
>> [dial-hylafax]
>> exten => s,1,GotoIf($["${FROM_DID:0:1}" = "1"]?prune:cont)
>> exten => s,n(prune),Set(__FROM_DID=${FROM_DID:1})
>> exten => s,n(cont),GotoIf($[${FAX_IDX} <= ${MAX_IDX}]?tryfax:nofax)
>> exten => s,n(tryfax),Set(STATE=${DEVICE_STATE(Custom:iaxmodem${FAX_IDX})})
>> exten => s,n,NoOp(${STATE})
>> exten => s,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=INUSE)
>> exten => s,n,Dial(IAX2/iaxmodem${FAX_IDX}/${FROM_DID},60,g)
>> exten => s,n,Goto(s-${DIALSTATUS},1)
>> exten => s,n(nofax),Playtones(busy)
>> exten => s,n,NoOp(NO MODEMS AVAILABLE)
>> exten => s,n,Wait(20)
>> exten => s,n,Hangup()
>> exten => s-ANSWER,1,NoOp(IAXMODEM HANGUP)
>> exten => s-ANSWER,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
>> exten => s-ANSWER,n,Hangup()
>> exten => _s-.,1,Set(FAX_IDX=${MATH(1+${FAX_IDX},i)})
>> exten => _s-.,n,Goto(s,1)
>> exten => h,1,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
>>
>> The current state requires me to restart Asterisk almost every day.
>> I'm also seeing this on a completely different machine after upgrading
>> from Asterisk10 to 11.
>>
>>
> I'm wondering if this is a problem in the SLIN converter?
> I do use SLIN with iaxmodem.
>
>
Also of note,
A quick valgrind run and attempting to send a few faxes produces a bunch of
these in the valgrind output:
==30640== 217,259 bytes in 287 blocks are definitely lost in loss record
1,778 of 1,789
==30640==at 0x4C267CC: calloc (vg_replace_malloc.c:467)
==30640==by 0x4DC50E: ast_frdup (utils.h:523)
==30640==by 0x47125F: __ast_queue_frame (channel.c:1284)
==30640==by 0x1EF75589: __do_deliver (chan_iax2.c:3102)
==30640==by 0x1EF76C5A: schedule_delivery (chan_iax2.c:4374)
==30640==by 0x1EF8F497: socket_process_helper (chan_iax2.c:12010)
==30640==by 0x1EF99C37: iax2_process_thread (chan_iax2.c:12030)
==30640==by 0x56C458: dummy_start (utils.c:1192)
==30640==by 0x5E359C9: start_thread (pthread_create.c:300)
==30640==by 0x270326FF: ???
--
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Re: [asterisk-users] day night service toggle
On Fri, 28 Nov 2014, A J Stiles wrote: And note that this really should be done by dialling separate numbers for "in" and "out", because toggle actions are annoying as hell in practice -- it's easier to remember two different numbers, than to remember what state you are currently in. According to NANPA's 'Vertical Service Codes, Code Definitions' (http://www.nanpa.com/number_resource_info/vsc_definitions.html), *72 and *73 are for call forwarding activation and deactivation (respectively). Of course, the OP could use execiftime() instead and just make sure to go to lunch and return at the exact same time every day. The dialplan coding is easier and no pesky codes to remember. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] day night service toggle
On Thursday 27 Nov 2014, Control Oye wrote:
> Hi,
>
> I need dialplan to set INCOMING call forwarding during lunch break to my
> secretary.
>
> I want that I can set call forwarding by dialing an extension number to
> turn it ON or OFF.
>
> I am using asterisk 11.
What you need to do is, set a global variable from within your dialplan, to
indicate whether you are at your desk (meaning calls to your number should go
to your phone) or on lunch (meaning calls to your number should go somewhere
else).
And note that this really should be done by dialling separate numbers for "in"
and "out", because toggle actions are annoying as hell in practice -- it's
easier to remember two different numbers, than to remember what state you are
currently in.
If 101 is you and 102 is the person who fields your calls while you are on
lunch, you need something like this;
; extension 101
exten => 101,1,GotoIf($[${ON_LUNCH}] ? on_lunch : at_desk)
exten => 101,n(at_desk),Dial(SIP/101)
exten => 101,n,Hangup()
exten => 101,n(on_lunch),Dial(SIP/102)
exten => 101,n,Hangup()
; These extensions should be in some context which is only callable from 101
; extension 771 sets "on lunch"
exten => 771,1,Set(GLOBAL(ON_LUNCH)=1)
exten => 771,n,Playback(ajs-set_on_lunch)
exten => 771,n,Hangup()
; extension 770 sets "at desk"
exten => 770,1,Set(GLOBAL(ON_LUNCH)=)
exten => 770,n,Playback(ajs-set_at_desk)
exten => 770,n,Hangup()
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
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Re: [asterisk-users] day night service toggle
Hello Control,
On Thursday, November 27, 2014, 9:48:30 PM, you wrote:
> I need dialplan to set INCOMING call forwarding during lunch break to my
> secretary.
>
> I want that I can set call forwarding by dialing an extension number to turn
> it ON or OFF.
There are probably lots of ways to do this, and it would be worth
checking to see whether your handset can be programmed to do the
forwarding, which is simpler than setting up a dialplan to handle it.
I pass incoming calls to a mobile phone based on a toggle. There are
three parts to this:
[1] I have a Global Variable (MDIVERT) that holds the current Divert status.
[2] I have an extension (*70) that toggles the value of the Global variable
[3] I have a GotoIf in my incoming call handler that selects different
dial strings depending on whether the is set or not.
I added a two voice prompts to announce the status of MDIVERT.
Dialling *70 the first time sets the variable and dialling *70 again
unsets it.
[1] Create the Global Variable
[globals]
MDIVERT=0
[2] Toggle the variable.
exten => *70,1,Set(GLOBAL(MDIVERT)=${IF($[ ${MDIVERT} = 1]?0:1)})
exten => *70,n,GotoIf($[ $[ "${MDIVERT}" = "0" ] ]?notset)
exten => *70,n,playback(divertactive)
exten => *70,n,Hangup()
exten => *70,n(notset),playback(divertinactive)
exten => *70,n,Hangup()
[3] GotoIf in call handler
exten => 100,1,GotoIf($["${MDIVERT}" = "1"]?divert) ; check variable
exten => 100,n,Dial(SIP/ManagerPhone,30,ctkx) ;Call your phone
exten => 100,n,Hangup()
exten => 100,n(divert),Dial(SecretaryPhone,30,ctkx) ; call Secretary
exten => 100,n,Hangup()
I have cut out a lot from the dialplan to give you an idea, but I am
sure that others will chip in with their ideas and point out the
failings in my approach :-)
J
--
Best regards,
Julianmailto:jb_s...@trink.co.uk
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