Ryan,

I get...

pbx ~ # upgrade-run-image check http://mirror.astlinux.org/ast18-firmware-1.x
Current version is: astlinux-1.0-6352,  Newest available version is: 
astlinux-1.1.3

Your error indicates a DNS error or connectivity problem.

But, yes "upgrade-run-image upgrade ..." should work from your 1.0.4 .

Browse through the ChangeLog to see what has changed, but it should be a smooth 
upgrade.
http://svn.code.sf.net/p/astlinux/code/branches/1.0/docs/ChangeLog.txt

BTW, we plan on tagging 1.1.4 this week and official images sometime after that.

Lonnie


On Jan 14, 2014, at 8:53 PM, Ryan Bantz wrote:

> Hello Everyone,
> 
> I tried adding Answer before the whole recording statement and that didn't 
> have any affect.
> 
> I did grab the latest stable image from the site and loaded up. I copied the 
> contents of the /etc/asterisk directory to the new instance and the issue is 
> gone. Looks like I will just have to upgrade the live box I am using.
> 
> Should I be able to upgrade to the latest version from 1.0.4? Running the 
> update results in the following.
> 
> upgrade-run-image check http://mirror.astlinux.org/ast18-firmware-1.x
> 
> wget: bad address 'mirror.astlinux.org'
> 
> No version available.
> 
> -Ryan
> 
> 
> On Tue, Jan 14, 2014 at 12:31 AM, Ioan Indreias <[email protected]> 
> wrote:
> Hello Ryan,
> 
> I second Michael's suggestion for using 'Answer' as the first step in 903 
> extension dialplan.
> 
> Additionally, my suggestion is to set 'maxduration' parameter in the Record 
> application?
> 
> Something like:
> 
> exten => 903,n,Record(custom-sounds/ivr.gsm,,60,y)
> 
> should prevent the record to have a bigger length than 1 minute.
> 
> HTH,
> Ioan
> 
> PS: I also add option 'y' in case you would like to test "Terminate recording 
> if *any* DTMF digit is received."
> asterisk -rx "core show application Record"
> should reveal all options you may use...
> 
> 
> 
> On 14-Jan-14 1:41 AM, Michael Keuter wrote:
>> Maybe there's some kind of internal timeout.
>> 
>> 2 more ideas : 
>> 1st try an "Answer()" as first step in your 903 extension.
>> 2nd try the latest AstLinux beta from our http://build.astlinux.org/ site, 
>> which includes Asterisk 1.8.25.0.
>> 
>> 
>> Sent from my iPad
>> 
>> Michael
>> 
>> Am 14.01.2014 um 00:10 schrieb Ryan Bantz <[email protected]>:
>> 
>>> Something must be keeping the channel open that I just haven't found yet. I 
>>> have attached a copy of the entire dialplan. I don't have any AGI scripts. 
>>> The channel appears to say up after the user has hung up.
>>> 
>>> Channel              Location             State   Application(Data)         
>>>     
>>> 
>>> SIP/00907a0dea94-000 903@hos-devices:2    Up      
>>> Record(custom-sounds/ivr.gsm) 
>>> 
>>> 1 active channel
>>> 
>>> 1 active call
>>> 
>>> 4 calls processed
>>> 
>>> [Jan 13 15:47:55]
>>> 
>>> -Ryan
>>> 
>>> On Mon, Jan 13, 2014 at 1:38 PM, Lonnie Abelbeck 
>>> <[email protected]> wrote:
>>> Ryan,
>>> 
>>> I tried your example with Asterisk 1.8.25.0 on a Jetway Atom box...
>>> --
>>>     -- Executing [4@default:2] Wait("SIP/10.10.50.1-00000004", "2") in new 
>>> stack
>>>     -- Executing [4@default:3] Record("SIP/10.10.50.1-00000004", 
>>> "custom-sounds/ivr.gsm") in new stack
>>>     -- <SIP/10.10.50.1-00000004> Playing 'beep.ulaw' (language 'en')
>>>   == Spawn extension (default, 4, 3) exited non-zero on 
>>> 'SIP/10.10.50.1-00000004'
>>> --
>>> A hangup works properly as well as deleting "custom-sounds/ivr.gsm" if it 
>>> existed.
>>> 
>>> Possibly something keeping "custom-sounds/ivr.gsm" from being removed on a 
>>> hangup ?  Grasping...
>>> 
>>> Lonnie
>>> 
>>> 
>>> On Jan 13, 2014, at 1:15 PM, Lonnie Abelbeck wrote:
>>> 
>>> > Ryan,
>>> >
>>> > This link might also help...
>>> >
>>> > Autodestruct on dialog 'with owner in place' - help needed
>>> > http://forums.asterisk.org/viewtopic.php?f=1&t=86221
>>> >
>>> > Also if you have some sort of AGI that is not completing or such it could 
>>> > be causing this issue.
>>> > http://stackoverflow.com/questions/7599934/asterisk-issue-autodestruct-on-dialog-with-owner-in-place-method-bye
>>> >
>>> > Something is keeping the SIP session from terminating, so this is a 
>>> > warning, not an error.
>>> >
>>> > This appears to be an Asterisk issue and not an AstLinux problem, per-se .
>>> >
>>> > Lonnie
>>> >
>>> >
>>> > On Jan 13, 2014, at 12:58 PM, Lonnie Abelbeck wrote:
>>> >
>>> >> Ryan,
>>> >>
>>> >> What version of Asterisk are you using ?
>>> >>
>>> >> There was a bug in early versions of Asterisk 1.8 ...
>>> >>
>>> >> Calls not released after BYE
>>> >> https://issues.asterisk.org/jira/browse/ASTERISK-19425
>>> >>
>>> >> Lonnie
>>> >>
>>> >>
>>> >> On Jan 13, 2014, at 12:28 PM, Ryan Bantz wrote:
>>> >>
>>> >>> Here is a snippet of hanging up without pressing # to end the recording.
>>> >>>
>>> >>> pbx*CLI>
>>> >>>
>>> >>> == Using SIP RTP CoS mark 5
>>> >>>
>>> >>>   -- Executing [903@devices:1] Wait("SIP/00907a0dea94-00000001", "2") 
>>> >>> in new stack
>>> >>>
>>> >>>   -- Executing [903@devices:2] Record("SIP/00907a0dea94-00000001", 
>>> >>> "custom-sounds/ivr.gsm") in new stack
>>> >>>
>>> >>>   -- <SIP/00907a0dea94-00000001> Playing 'beep.ulaw' (language 'en')
>>> >>>
>>> >>> [Jan 13 12:17:12] WARNING[382]: chan_sip.c:3870 __sip_autodestruct: 
>>> >>> Autodestruct on dialog '[email protected]' with 
>>> >>> owner in place (Method: BYE). Rescheduling destruction for 10000 ms
>>> >>>
>>> >>> [Jan 13 12:17:26] WARNING[382]: chan_sip.c:3870 __sip_autodestruct: 
>>> >>> Autodestruct on dialog '[email protected]' with 
>>> >>> owner in place (Method: BYE). Rescheduling destruction for 10000 ms
>>> >>>
>>> >>> pbx*CLI>
>>> >>>
>>> >>>
>>> >>>
>>> >>> -Ryan
>>> >>>
>>> >>>
>>> >>>
>>> >>> On Mon, Jan 13, 2014 at 12:06 PM, Lonnie Abelbeck 
>>> >>> <[email protected]> wrote:
>>> >>> Ryan,
>>> >>>
>>> >>> Can you post a snippet of "asterisk -rvvvv" output (sanitize if 
>>> >>> necessary) starting when extension 903 is called and the end user 
>>> >>> hangs-up.
>>> >>>
>>> >>> Lonnie
>>> >>>
>>> >>>
>>> >>> On Jan 13, 2014, at 11:43 AM, Ryan Bantz wrote:
>>> >>>
>>> >>>> I have verified that the custom directory is created on the flash card.
>>> >>>>   lrwxrwxrwx    1 root     root            21 Jan 13 11:20 
>>> >>>> /var/lib/asterisk/sounds/custom-sounds -> /mnt/kd/custom-sounds
>>> >>>>
>>> >>>> I didn't have also updated the record function to read:
>>> >>>>   exten => 903,n,Record(custom-sounds/ivr.gsm) ; Press # to stop 
>>> >>>> recording
>>> >>>>
>>> >>>> There wasn't a hangup extension defined the so I added the following 
>>> >>>> to the recording context.
>>> >>>>   exten => h,1,Hangup
>>> >>>>
>>> >>>> I still seem to have the issue were hanging up before pressing # to 
>>> >>>> end the recording locks the box up. It appears that something is 
>>> >>>> preventing the call from ending. I turned on some more logging and the 
>>> >>>> console fills up with the following.
>>> >>>>
>>> >>>> [Jan 13 11:12:05] WARNING[380]: chan_sip.c:3870 __sip_autodestruct: 
>>> >>>> Autodestruct on dialog '[email protected]' 
>>> >>>> with owner in place (Method: BYE). Rescheduling destruction for 10000 
>>> >>>> ms
>>> >>>>
>>> >>>> Any other thoughts?
>>> >>>>
>>> >>>> -Ryan
>>> >>>>
>>> >>>>
>>> >>>> On Sun, Jan 12, 2014 at 9:03 AM, Lonnie Abelbeck 
>>> >>>> <[email protected]> wrote:
>>> >>>>
>>> >>>> On Jan 12, 2014, at 5:35 AM, Michael Keuter wrote:
>>> >>>>
>>> >>>>>
>>> >>>>> Am 12.01.2014 um 12:31 schrieb Michael Keuter 
>>> >>>>> <[email protected]>:
>>> >>>>>
>>> >>>>>>
>>> >>>>>> Am 11.01.2014 um 21:30 schrieb Ryan Bantz 
>>> >>>>>> <[email protected]>:
>>> >>>>>>
>>> >>>>>>> Hello,
>>> >>>>>>>
>>> >>>>>>> I am running an embedded version of AstLinux on an Alix board. I 
>>> >>>>>>> have the following code in my dial plan. If a user is leaving a 
>>> >>>>>>> recording and hangs up prior to pressing # to end the recording the 
>>> >>>>>>> box will lock up. The box is inaccessible via SSH and needs to be 
>>> >>>>>>> power cycled. As far as I can tell it looks like it fills up the 
>>> >>>>>>> ram disk. Is there a way around this to prevent the system from 
>>> >>>>>>> locking up?
>>> >>>>>>>
>>> >>>>>>> ; Record ivr
>>> >>>>>>>
>>> >>>>>>> exten => 903,1,Wait(2) ; Call 903 to record new sound files
>>> >>>>>>>
>>> >>>>>>> exten => 903,n,Record(custom-sounds/ivr:gsm) ; Press # to stop 
>>> >>>>>>> recording
>>> >>>>>>>
>>> >>>>>>> exten => 903,n,Wait(2)
>>> >>>>>>>
>>> >>>>>>> exten => 903,n,Playback(custom-sounds/ivr) ; Listen to your 
>>> >>>>>>> recording
>>> >>>>>>>
>>> >>>>>>> exten => 903,n,wait(2)
>>> >>>>>>>
>>> >>>>>>> exten => 903,n,Hangup
>>> >>>>>>>
>>> >>>>>>>
>>> >>>>>>>
>>> >>>>>>> -Ryan
>>> >>>>>>
>>> >>>>>> Hi Ryan,
>>> >>>>>>
>>> >>>>>> I use nearly the same dialplan stuff to record custom sounds, and it 
>>> >>>>>> works fine for me. Which AstLinux version are you using?
>>> >>>>>> "custom-sounds" is normally a symlink to "/mnt/kd/custom-sounds", so 
>>> >>>>>> the file should be recorded on your CompactFlash card and not in RAM.
>>> >>>>>> Can you verify this?
>>> >>>>>>
>>> >>>>>> ls -l /var/lib/asterisk/sounds/custom-*
>>> >>>>>> lrwxrwxrwx    1 root     root            21 Jan  9 08:36 
>>> >>>>>> /var/lib/asterisk/sounds/custom-sounds -> /mnt/kd/custom-sounds
>>> >>>>>
>>> >>>>>
>>> >>>>> Update: the colon (":") in your record statement needs to be a dot 
>>> >>>>> (".") => ivr.gsm
>>> >>>>> Maybe this is the problem.
>>> >>>>>
>>> >>>>> Syntax (1.8):
>>> >>>>> Record(filename.format[,silence[,maxduration[,options]]])
>>> >>>>>
>>> >>>>> Michael
>>> >>>>
>>> >>>> Hi Ryan,
>>> >>>>
>>> >>>> I think Michael has it covered above, but do you have the 'h' (hangup) 
>>> >>>> extension defined ?  Something simple...
>>> >>>> --
>>> >>>> exten => h,1,Hangup
>>> >>>> --
>>> >>>> Or possibly something else ?
>>> >>>>
>>> >>>> Lonnie
>>> >>
>>> >>
> 
> 
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