Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
Hello Jerry, when you run asterisk using su, ownership of audio device files does not get updated. When you login, you get the permissions. You can verify by ls -l and getfacl on the device file. Marek --- Original Message --- On Thursday, September 14th, 2023 at 14:33, Jerry Geis wrote: > On Wed, Sep 13, 2023 at 5:20 PM Jerry Geis wrote: > >>>An issue[1] was already created by asterisk at phreaknet.org and they also >>>put >>>a fix up for review and inclusion[2]. >> >>>[1] https://github.com/asterisk/asterisk/issues/308 >>>[2] https://github.com/asterisk/asterisk/pull/309 >> >> The change "seems" to be working. >> Will test more tomorrow - had to leave. >> THANKS! >> Jerry > > Yes - this fix is working for me. > > Only issue I have now is, I used to run asterisk like this: > su silentm -c "/usr/sbin/asterisk -fn" > I also tried > su silentm -l -c "/usr/sbin/asterisk -fn" > > these do not work for the chan_console. I have to actually login as silentm > and then run asterisks - to HEAR the audio. > doing su above I do not hear the audio - but the CLI looks the same - no > errors. > > Thoughts? > > Jerry-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
On Wed, Sep 13, 2023 at 5:20 PM Jerry Geis wrote: > >An issue[1] was already created by asterisk at phreaknet.org and they > also put > >a fix up for review and inclusion[2]. > > >[1] https://github.com/asterisk/asterisk/issues/308 > >[2] https://github.com/asterisk/asterisk/pull/309 > > > The change "seems" to be working. > Will test more tomorrow - had to leave. > THANKS! > > Jerry > Yes - this fix is working for me. Only issue I have now is, I used to run asterisk like this: su silentm -c "/usr/sbin/asterisk -fn" I also tried su silentm -l -c "/usr/sbin/asterisk -fn" these do not work for the chan_console. I have to actually login as silentm and then run asterisks - to HEAR the audio. doing su above I do not hear the audio - but the CLI looks the same - no errors. Thoughts? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>An issue[1] was already created by asterisk at phreaknet.org and they also put >a fix up for review and inclusion[2]. >[1] https://github.com/asterisk/asterisk/issues/308 >[2] https://github.com/asterisk/asterisk/pull/309 The change "seems" to be working. Will test more tomorrow - had to leave. THANKS! Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
An issue[1] was already created by aster...@phreaknet.org and they also put a fix up for review and inclusion[2]. [1] https://github.com/asterisk/asterisk/issues/308 [2] https://github.com/asterisk/asterisk/pull/309 On Wed, Sep 13, 2023 at 4:27 PM Jerry Geis wrote: > > I have found that I can add the restart of asterisk (killall -9 asterisk) > to the h extension- BOY is that UGLY. > > chan_console is not a testing device - how can we get this nasty bug fixed > ? > > Jerry > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
I have found that I can add the restart of asterisk (killall -9 asterisk) to the h extension- BOY is that UGLY. chan_console is not a testing device - how can we get this nasty bug fixed ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
> After a hung call, can you run core restart now from the asterisk console? Doing a "killall -9 asterisk" is the only thing that works I tried killall asterisk - does not free up the channel the asterisk "core restart now" takes like a good 20 seconds to return but does work. The issue is I cannot run it after teh Dial() as the Dial(Console/default,20,g) never returns to the dial plan. jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
It worked with my test. I'm on Asterisk 18.19.0
-- Executing [517xxx@voipms:4] System("IAX2/voipms-15815", "asterisk
-rx 'core restart now'") in new stack
-- Remote UNIX connection
Asterisk uncleanly ending (0).
Executing last minute cleanups
== Destroying musiconhold processes
== Manager unregistered action DBGet
== Manager unregistered action DBGetTree
== Manager unregistered action DBPut
== Manager unregistered action DBDel
== Manager unregistered action DBDelTree
Preparing for Asterisk restart...
Asterisk is now restarting...
asterisk*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
After a hung call, can you run core restart now from the asterisk console?
Doug
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>Using system() you could issue a asterisk -rx 'core restart now' So I tried this exten => s,1,Playback(beep) exten => s,n,Dial(Console/default,20,g) exten => s,n,Hangup exten => s,n,System(asterisk -rx 'core restart now') But it does not continue. Last thing I see is "Exited non zero" so its not doing the hangup or the system. jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
I have noticed that once my message speaks - the server thinks its done and HUNGUP, the endpoint STILL thinks the channel is active - the last message says "Rx: BYE" on sip show channels I tried ADDING to Dial() ,20,g and then had a Hangup after teh dial. Its NOT getting there to hangup. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>> Is there a dial plan call that can "exit asterisk" or completely reload >> everything - killall active calls and start over ? Using system() you could issue a asterisk -rx 'core restart now' Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
Is there a dial plan call that can "exit asterisk" or completely reload everything - killall active calls and start over ? seems the console/dummy (chan_console) is holding some resource. How do I just "exit" and start over after call came in ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
I don't know if this will help you, but looking back through an old config I have for an older version of Asterisk, I had used chan_console with the old and now defunct app_rpt app to listen to audio on various nodes via the console for testing. Here is what I did: In console.conf, I defined this: [default] input_device = default output_device = default autoanswer = no context = extension = callerid = language = en overridecontext = no mohintrepret = default active = yes In modules.conf I loaded the audio module (in this case it was chan_alsa.so, but I also could use chan_oss.so). I made sure noload was commented out for chan_alsa.so In alsa.conf, I defined some of the same things as in console.conf: [general] autoanswer=no context= extension= inputdevice=plughw:0,1 otuputdevice=plughw:0,0 mute=true You'll need to check your ALSA device to see what the input and output devices are. That last line is important, since on the console you may not have a mic that works to talk, you just want to listen, In extensions.conf, I defined a dialplan that instead of trying to dial out, it just answered the call and then threw me into the app. Then to dial from the console, I woudl use: console dial And it woudl use the context I defined and launch the Rpt app. What you could do is define somehting like this ,but have the extension use DISA so that you can then get dumped into your normal dialplan logic where you could use "console dial xxx". No guarantees that this will work with a newer version of Asterisk, but this did work with a 1.8 setup I used to have (that I have the configs saved for). -Stacy On 9/8/23 10:28 AM, Jerry Geis wrote: So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>>> How do we get this working For the time being, go back to 18.14.0 Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>
>
> Not sure if this is the same thing you're seeing, but chan_console
> currently has a known deadlock issue that has not been resolved:
> https://issues-archive.asterisk.org/ASTERISK-30481
> It's quite easy to reproduce, so it may be the case that the module is
> currently unusable.
>
Well this is a bummer
[Sep 8 08:45:28] WARNING[313684][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
--- <("<) --- Hangup on Console --- (>")> ---
[Sep 8 08:45:28] ERROR[313683]: utils.c:1027 lock_info_destroy: Thread
'stream_monitor started at [ 390] chan_console.c start_stream()'
still has a lock! - 'pvt' (0x55e647a245e0) from 'stream_monitor' in
chan_console.c:281!
How do we get this working
Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
On 9/8/2023 8:18 AM, Jerry Geis wrote:
But when I do a second test. Asterisk HANGS on ChanIsAvail()
Then I thought lets SKIP that - and just let it do the Dial() - I
stopped everything - got it running again. - and then the Dial() hangs
on the second call.
So both ChanIsAvail() or Dial() both hang on the second call in.
So only 1 call in will work.
Below is the CLI report of the call that works.
This is my context
[smvoice-mediacontroller-public-address]
exten => s,1,ChanIsAvail(Console/default)
exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/default)
exten => s,n,Hangup
Not sure if this is the same thing you're seeing, but chan_console
currently has a known deadlock issue that has not been resolved:
https://issues-archive.asterisk.org/ASTERISK-30481
It's quite easy to reproduce, so it may be the case that the module is
currently unusable.
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
Some progress to report:
I had to run asterisk as the user logged in - actually not even that. I
could not "su user -c " to that user - I had to run it as that user.
Then I did a test and got audio! Great...
But when I do a second test. Asterisk HANGS on ChanIsAvail()
Then I thought lets SKIP that - and just let it do the Dial() - I stopped
everything - got it running again. - and then the Dial() hangs on the
second call.
So both ChanIsAvail() or Dial() both hang on the second call in.
So only 1 call in will work.
Below is the CLI report of the call that works.
This is my context
[smvoice-mediacontroller-public-address]
exten => s,1,ChanIsAvail(Console/default)
exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/default)
exten => s,n,Hangup
Now what ???
Jerry
onnected to Asterisk 18.18.0 currently running on nuc11cdev2 (pid = 282195)
== Using SIP RTP CoS mark 5
> 0x7feeec0086b0 -- Strict RTP learning after remote address set to:
192.168.1.8:17526
-- Executing [public_address@smvoice-mediacontroller:1]
SoftHangup("SIP/devgeis_to_nuc11cdev2-", "ALSA/dummy") in new stack
-- Executing [public_address@smvoice-mediacontroller:2]
Goto("SIP/devgeis_to_nuc11cdev2-",
"smvoice-mediacontroller-public-address,s,1") in new stack
-- Goto (smvoice-mediacontroller-public-address,s,1)
-- Executing [s@smvoice-mediacontroller-public-address:1]
NoOp("SIP/devgeis_to_nuc11cdev2-", "JERRY") in new stack
-- Executing [s@smvoice-mediacontroller-public-address:2]
Playback("SIP/devgeis_to_nuc11cdev2-", "beep") in new stack
> 0x7feeec0086b0 -- Strict RTP switching to RTP target address
192.168.1.8:17526 as source
-- Playing 'beep.gsm' (language
'en')
-- Executing [s@smvoice-mediacontroller-public-address:3]
Dial("SIP/devgeis_to_nuc11cdev2-", "Console/default") in new stack
--- <("<) --- Call to device 'default' on console from 'MyName Here'
<2564286000> --- (>")> ---
--- <("<) --- Auto-answered --- (>")> ---
-- Called Console/default
-- Console/default answered SIP/devgeis_to_nuc11cdev2-
-- Channel Console/default joined 'simple_bridge' basic-bridge
[Sep 8 08:07:10] WARNING[282457][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
-- Channel SIP/devgeis_to_nuc11cdev2- joined 'simple_bridge'
basic-bridge
> 0x7feeec0086b0 -- Strict RTP learning complete - Locking on source
address 192.168.1.8:17526
-- Channel SIP/devgeis_to_nuc11cdev2- left 'simple_bridge'
basic-bridge
-- Channel Console/default left 'simple_bridge' basic-bridge
[Sep 8 08:07:17] WARNING[282457][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
--- <("<) --- Hangup on Console --- (>")> ---
== Spawn extension (smvoice-mediacontroller-public-address, s, 3) exited
non-zero on 'SIP/devgeis_to_nuc11cdev2-'
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
In the old days when I was using console/dsp, I would have to use
alsamix from the console to verify that the output wasn't muted. Maybe
it's still the same.
Doug
On 9/7/23 03:43 PM, Jerry Geis wrote:
ok switching to "Console/default" does show the text
--- <("<) --- Call to device 'default' on console from 'default'
<2564286000> --- (>")> ---
--- <("<) --- Auto-answered --- (>")> ---
However I don't hear any audio.
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
ok switching to "Console/default" does show the text
--- <("<) --- Call to device 'default' on console from 'default'
<2564286000> --- (>")> ---
--- <("<) --- Auto-answered --- (>")> ---
However I don't hear any audio.
Thanks
Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
On Thu, Sep 7, 2023 at 3:27 PM Jerry Geis wrote: > > I found "console list available" > > === > === - > === Device Name: default > === ---> Default Input Device > === ---> Default Output Device > === - > === > === - > === Device Name: dmix > === ---> Output Device > === - > === > = > > dmix is there and default is there > I tried both - and get the same error > Console device "dmix" not found . etc. > Yes, because that lists the available devices. You have to configure it in console.conf in order to be able to dial it. If you haven't configured a thing named "dmix" in console.conf, then it's not going to work. "console list available" show available devices that you can use in the configuration "console list devices" show what is actually configured -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
I found "console list available" === === - === Device Name: default === ---> Default Input Device === ---> Default Output Device === - === === - === Device Name: dmix === ---> Output Device === - === = dmix is there and default is there I tried both - and get the same error Console device "dmix" not found . etc. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
On Thu, Sep 7, 2023 at 3:20 PM Joshua C. Colp wrote: > On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis wrote: > >> Joshua >> >> Asterisk 18.14.0 with chan_alsa and Console/dsp works. >> This does not work in 18.18.0 with chan_console enabled. >> I am on Ubuntu 20.04 LTS. >> >> Is there a howto for the new chan_console ? >> > > I'm not aware of one. The module itself has existed since at least > Asterisk 1.8 > > >> how can I get this working again ? >> I am trying to just play audio on pulse audio. >> > > I don't have anything additional to add beyond what I've said and the > config file I've provided. > I can say that with the default configuration file it would be Console/default though, and would use the system default input and output devices according to PortAudio. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis wrote: > Joshua > > Asterisk 18.14.0 with chan_alsa and Console/dsp works. > This does not work in 18.18.0 with chan_console enabled. > I am on Ubuntu 20.04 LTS. > > Is there a howto for the new chan_console ? > I'm not aware of one. The module itself has existed since at least Asterisk 1.8 > how can I get this working again ? > I am trying to just play audio on pulse audio. > I don't have anything additional to add beyond what I've said and the config file I've provided. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
Joshua Asterisk 18.14.0 with chan_alsa and Console/dsp works. This does not work in 18.18.0 with chan_console enabled. I am on Ubuntu 20.04 LTS. Is there a howto for the new chan_console ? how can I get this working again ? I am trying to just play audio on pulse audio. Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
