[asterisk-users] Hardware requirements
I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? /Y.T. Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. http://au.docs.yahoo.com/mail/unlimitedstorage.html ___ --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] AGI with System() ?
On Sat, Oct 13, 2007 at 05:45:26PM -0700, Dominic Son wrote: > Hi. > > You mean to use the AGI funtion in the particular programming > language? yeah. i tried, same results.. : T I guess that this is a permissions issue. Check what you get in the standard error. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Display channels and codecs
Try sip show channels from the CLI - Original Message - From: "Scott Moseman" <[EMAIL PROTECTED]> To: Sent: Friday, October 12, 2007 6:12 PM Subject: [asterisk-users] Display channels and codecs > Is there an easy way to show all active channels AND the codecs > they're using? Other than going through EACH channel individually to > check the codec which is, obviously, not a very efficient process. > > Thanks, > Scott > > ___ > --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] AGI with System() ?
On Saturday 13 October 2007 18:35:28 Dominic Son wrote: > tried both as suggested...though AGI says it's succesful: > > AGI Rx << EXEC System rm /var/lib/asterisk/sounds/abandons.gsm > AGI Tx >> 200 result=0 The EXEC command takes two arguments, an application name and an argument. So, it, in fact, ran: EXEC System rm (ignoring anything beyond the second argument). What you probably meant to send it was: EXEC System "rm /var/lib/asterisk/sounds/abandons.gsm" -- Tilghman ___ --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] AGI with System() ?
Hi.
You mean to use the AGI funtion in the particular programming
language? yeah. i tried, same results.. : T
i guess i'll have to put it in a database, and flag it to remove
manually for now...
On 10/13/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote:
> Dominic Son wrote:
>
> > tried both as suggested...though AGI says it's succesful:
> >
> > AGI Rx << EXEC System rm /var/lib/asterisk/sounds/abandons.gsm
> > AGI Tx >> 200 result=0
> >
> > the abandons.gsm file is still there...
>
> Umm, then I don't know what's going wrong.
>
> > i have to delete it through my agi because i'm recording sounds, and i
> > want users to hear their recording and redo it if they choose to.
> >
> > so there's a 'RECORD FILE', but not a 'DELETE FILE' .. : T
> >
> > On 10/13/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote:
> >> Dominic Son wrote:
> >>
> >>> Uuugh..for the life of me, i cannot delete sound files using
> >>> "EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)"
>
> >> But I don't understand why you would want to do that instead of
> >> just running the command in your script.
>
> You did not get my point. Why not just do something like
> unlink('/var/lib/asterisk/sounds/abandons.gsm');
> or
> exec('rm -f '.escapeShellArg('/var/lib/asterisk/sounds/abandons.gsm'));
> in your script? (examples are for PHP)
>
> Regards,
> Philipp Kempgen
>
> --
> amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
> Let's use IT to solve problems and not to create new ones.
> Asterisk? -> http://www.das-asterisk-buch.de
>
> Geschäftsführer: Stefan Wintermeyer
> Handelsregister: Neuwied B 14998
>
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Re: [asterisk-users] Dock-N-Talk with Asterisk, Anyone?
Jeng Yu wrote: > I would like to hear if anyone out there in Asteriskland has used the > Dock-N-Talk (DNT) box to connect cell phones to Asterisk box. The only problem I noticed is that after a random amount of time the box will lost contact/synch with the cell phone. I'm using DockNTalk for about 2 or 3 years, and this is happening after 1 or 2 weeks. After a power cycle the DNT will work again. ___ --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] Affordable SIP Trunk for Home PBX ?
On Saturday October 13 2007 12:47 pm, Doug Lytle wrote: > John Millican wrote: > > On Saturday October 13 2007 12:09 pm, Lee Jenkins wrote: > > > > Be sure to read the fine print as most of the "unlimited" plans do > > actually have a limit on usage (even the ones I offer). Some are out in > > the open some > > Then don't advertise it as *unlimited* > > Seems simple, doesn't it? > > Doug Doug, You are absolutely correct, it should be simple but... When you are trying to market a product and are competing in a market littered with limited unlimited plans and knowing that these are the key words that a lot of people look for, the other partner in the company said "we have to have an unlimited plan". Long hard battle ensued but we came to an agreement. Okay, but we will call it the Unlimited3000 (yes pretty cheesy I know) plan and spell it out clearly in the contract also. Also if you left the rest of the post in you would see that I said I usually steer people away from these plans into a per minute as most people do not get close to 3000 minutes a month. Yes, there are those that do but not the majority. My apologies if anyone feels this is to close to a commercial post but I felt I should answer. 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] AGI with System() ?
Dominic Son wrote:
> tried both as suggested...though AGI says it's succesful:
>
> AGI Rx << EXEC System rm /var/lib/asterisk/sounds/abandons.gsm
> AGI Tx >> 200 result=0
>
> the abandons.gsm file is still there...
Umm, then I don't know what's going wrong.
> i have to delete it through my agi because i'm recording sounds, and i
> want users to hear their recording and redo it if they choose to.
>
> so there's a 'RECORD FILE', but not a 'DELETE FILE' .. : T
>
> On 10/13/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote:
>> Dominic Son wrote:
>>
>>> Uuugh..for the life of me, i cannot delete sound files using
>>> "EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)"
>> But I don't understand why you would want to do that instead of
>> just running the command in your script.
You did not get my point. Why not just do something like
unlink('/var/lib/asterisk/sounds/abandons.gsm');
or
exec('rm -f '.escapeShellArg('/var/lib/asterisk/sounds/abandons.gsm'));
in your script? (examples are for PHP)
Regards,
Philipp Kempgen
--
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Re: [asterisk-users] 'Start' in extension rules
Turbo Fredriksson wrote:
> Quoting Philipp Kempgen <[EMAIL PROTECTED]>:
>
>>> exten => s,1,Answer()
>>> exten => s,n,Goto(s-${DIALSTATUS},1)
>> This still doesn't make sense because you did not Dial()
>> before jumping based on ${DIALSTATUS}.
>
> Ok, make sense. But still no go:
>
> - s n i p -
> [default]
> exten => s,1,Answer()
> exten => s,2,Dial(SIP/${EXTEN},20,t)
> exten => s,4,Goto(default,s-${DIALSTATUS},1)
>
> exten => s-CANCEL,1,Hangup()
> exten => s-DONTCALL,1,Voicemail(${EXTEN},u)
> exten => s-NOANSWER,1,Voicemail(${EXTEN},u)
> exten => s-BUSY,1,Voicemail(${EXTEN},b)
>
> exten => 2403,1,Hangup()
>
> exten => _X.,1,Playback(pbx-invalid)
> exten => _X.,2,Hangup()
> - s n i p -
> The 's,...' part is STILL not 'executed'...!
No, because if you dial 2403 in your default context it would
match the 2403 extension not the s extension.
exten => _X.,1,Dial(SIP/${EXTEN},20)
exten => _X.,n,Goto(s-${DIALSTATUS},1)
Well, I don't want the first part
>> Then maybe
>>
>> [default]
>> exten => _X.,1,Dial(SIP/${EXTEN},20)
>> exten => _X.,n,Goto(s-${DIALSTATUS},1)
>
> How are these different?
They are not. But the part you left out is different. :)
>> Or you could define every extension and use a macro. There's an example
>> in the default extensions.conf file (if you ran `make samples`).
>
> Ah, got it... A little modifications for my need, and of I went!
[snip]
> One line per extension, exactly how I wanted it originally.
Great!
Regards,
Philipp Kempgen
--
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Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de
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Re: [asterisk-users] AGI with System() ?
tried both as suggested...though AGI says it's succesful: AGI Rx << EXEC System rm /var/lib/asterisk/sounds/abandons.gsm AGI Tx >> 200 result=0 the abandons.gsm file is still there... i have to delete it through my agi because i'm recording sounds, and i want users to hear their recording and redo it if they choose to. so there's a 'RECORD FILE', but not a 'DELETE FILE' .. : T On 10/13/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote: > Dominic Son wrote: > > > Uuugh..for the life of me, i cannot delete sound files using > > "EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)" > > > > through AGI > > agi show exec > Usage: EXEC > Executes with given . > > So I'd try > EXEC System rm foo > or > EXEC System rm\ foo > > But I don't understand why you would want to do that instead of > just running the command in your script. > > Regards, > Philipp Kempgen > > -- > amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de > Let's use IT to solve problems and not to create new ones. > Asterisk? -> http://www.das-asterisk-buch.de > > Geschäftsführer: Stefan Wintermeyer > Handelsregister: Neuwied B 14998 > > ___ > --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] 'Start' in extension rules
On Saturday 13 October 2007 18:08:49 Turbo Fredriksson wrote:
> Quoting Philipp Kempgen <[EMAIL PROTECTED]>:
> >> exten => s,1,Answer()
> >> exten => s,n,Goto(s-${DIALSTATUS},1)
> >
> > This still doesn't make sense because you did not Dial()
> > before jumping based on ${DIALSTATUS}.
>
> Ok, make sense. But still no go:
>
> - s n i p -
> [default]
> exten => s,1,Answer()
> exten => s,2,Dial(SIP/${EXTEN},20,t)
> exten => s,4,Goto(default,s-${DIALSTATUS},1)
1, 2, 4...
--
Tilghman
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Re: [asterisk-users] 'Start' in extension rules
Quoting Philipp Kempgen <[EMAIL PROTECTED]>:
>> exten => s,1,Answer()
>> exten => s,n,Goto(s-${DIALSTATUS},1)
>
> This still doesn't make sense because you did not Dial()
> before jumping based on ${DIALSTATUS}.
Ok, make sense. But still no go:
- s n i p -
[default]
exten => s,1,Answer()
exten => s,2,Dial(SIP/${EXTEN},20,t)
exten => s,4,Goto(default,s-${DIALSTATUS},1)
exten => s-CANCEL,1,Hangup()
exten => s-DONTCALL,1,Voicemail(${EXTEN},u)
exten => s-NOANSWER,1,Voicemail(${EXTEN},u)
exten => s-BUSY,1,Voicemail(${EXTEN},b)
exten => 2403,1,Hangup()
exten => _X.,1,Playback(pbx-invalid)
exten => _X.,2,Hangup()
- s n i p -
If I didn't have the '2403,1,Hangup()' line, I just got
'No such extension'. If I instead put:
- s n i p -
exten => 2403,1,Dial(SIP/${EXTEN},20,t)
exten => 2403,2,Set(GLOBAL(ORIGEXTEN)=${EXTEN})
exten => 2403,3,Goto(default,s-${DIALSTATUS},1)
- s n i p -
And changed the 'EXTEN' to 'ORIGEXTEN' in some places,
added the 'Set()' in some other (mainly as prio 3
in the 's,...' lines) it works. But the whole idea
here was so that I wouldn't have to use so many
lines...
The 's,...' part is STILL not 'executed'...!
>>> exten => _X.,1,Dial(SIP/${EXTEN},20)
>>> exten => _X.,n,Goto(s-${DIALSTATUS},1)
>>
>> > Well, I don't want the first part
>
> Then maybe
>
> [default]
> exten => _X.,1,Dial(SIP/${EXTEN},20)
> exten => _X.,n,Goto(s-${DIALSTATUS},1)
How are these different?
> Or you could define every extension and use a macro. There's an example
> in the default extensions.conf file (if you ran `make samples`).
Ah, got it... A little modifications for my need, and of I went!
For the archive:
- s n i p -
[macro-catchstatus]
exten => s,1,Dial(SIP/${ARG1},20,t) ; Dial the extension,
maximum 20 seconds
exten => s,2,Goto(s-${DIALSTATUS},1); Go to relevant dial
status
exten => s-CANCEL,1,Hangup()
exten => s-DONTCALL,1,Voicemail(${ARG2},u) ; Direct caller to the
voicemail w/ unavailible
exten => s-NOANSWER,1,Voicemail(${ARG2},u) ; Direct caller to the
voicemail w/ unavailible
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten => s-BUSY,1,Voicemail(${ARG2},b) ; Direct caller to the
voicemail w/ busy
exten => s-BUSY,2,Goto(default,s,1) ; If they press #,
return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else
as no answer
exten => a,1,VoicemailMain(${ARG2}) ; If they press *, send
the user into VoicemailMain
[default]
exten => s,1,Answer() ; Can't Playtones
unless we answer the line first
exten => 2403,1,Macro(catchstatus,${EXTEN},${EXTEN})
- s n i p -
One line per extension, exactly how I wanted it originally.
Thanx for the pointers!
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Re: [asterisk-users] Asterisk 1.4.13 build crashed
Kevin P. Fleming wrote: > As the message says, this is a bug in your compiler, and should be > reported to the packager (or in this case, since you are using LFS, > directly to the GCC maintainers). I will tell you that we have been > building Asterisk 1.4 with GCC 4.2.1 for quite a while without problems > on our development systems. As I said, 1.4.12 builds fine. I'll do a bit more digging and if I find a cause I'll report it upstream. Cheers. Al -- The way out is open! http://www.theopensourcerer.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] 'Start' in extension rules
On Sat, 13 Oct 2007, Turbo Fredriksson wrote: >> Turbo Fredriksson wrote: >> >>> I can't seem to get the [s]tart to work in my extensions... When a dialplan doesn't work as you expect, "show dialplan [context]" is your friend. Reply with show dialplan default and you may get a more specific answer. And, on a personal note, I prefer "sip" to "SIP" and would like to shoot whoever coded the parser to consider leading or trailing spaces to be significant :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] AGI with System() ?
Dominic Son wrote: > Uuugh..for the life of me, i cannot delete sound files using > "EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)" > > through AGI agi show exec Usage: EXEC Executes with given . So I'd try EXEC System rm foo or EXEC System rm\ foo But I don't understand why you would want to do that instead of just running the command in your script. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] How to use an Application from inside an Application?
Thx a lot for response. pbx_exec is very useful.
Pirlouwi.
2007/10/12, Tilghman Lesher <[EMAIL PROTECTED]>:
>
> On Friday 12 October 2007 04:28:42 Pirlouwi wrote:
> > I wonder if there is a way to build my own asterisk application (let us
> say
> > apps/app_myappl.c),
> > and to launch other existing applications from it (for example, doing an
> > apps/app_dial.c, or others).
>
> Both the Page app and VoicemailMain do this, respectively for MeetMe and
> Directory, so you can look at their source for examples.
>
> In the case of Page, the guts of it is:
> app = pbx_findapp("MeetMe");
> pbx_exec(chan, app, meetmeopts);
>
> Please check out app_page.c for the rest of the syntax.
>
> --
> Tilghman
>
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[asterisk-users] AGI with System() ?
Uuugh..for the life of me, i cannot delete sound files using
"EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)"
through AGI
the AGI debug log indicates the command executes successful ( equals 0)
but my files are clearly still there.
If i try System(rm ...) in my extensions.conf diaplan it'll work there.
Is there a bug in the AGI to use "System" ?
because i tried to copy files ('cp') as a test, and that didn't work either..
I'm running Trixbox v 1.1.0
Asterisk 1.2.9.1 svn rev 34876
with RAGI
--
Anything else, let me know.
- Dominic
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Re: [asterisk-users] Help 60Hz Hum?
Yes, Jay and Philip, you are right, but you can also have hums if ground cables for chassis protection against electrical hazards are making loops, if certain of them are in parallel and if they have different length between the equipments to protect. Many audio stages (unbalanced side, not balanced line side) are sensitive to that case. You can also encounter this case with PCB ground design. You can also have hums (often harmonics) due to a poor filtering from the power supplies (too old or poor quality chimical capacitors). And also, in rare specific cases, due to a strong radio field, near high power transmitters. It's not so easy to found what is the exact cause. Use an oscilloscope to check the different signals (audio, power lines, earth cable, etc...). Best Regards, Francois BERGERET France ___ --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] Distributed FAX - How to best complement asterisk ?
> "PvK" == Philipp von Klitzing <[EMAIL PROTECTED]> writes: PvK> Some of the bigger MFC printer/copy/fax combo devices by Brother PvK> (and maybe also other vendors?) provide a fax-via-smtp feature PvK> and can built fax networks that way. As far as I can tell, the Brother boxes require the user to enter fax-over-internet addresses as actual email addresses. There is no way to tell the machine to always use fax-over-internet, and there is also no way to configure a default domain. Asking fax users to select "email" and then type in [EMAIL PROTECTED] on a tiny little keyboard just won't fly. I don't understand why no fax vendors offer an easy fax-via-email user interface. /Benny ___ --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] 'Start' in extension rules
Turbo Fredriksson wrote:
> Quoting Philipp Kempgen <[EMAIL PROTECTED]>:
>
>> Turbo Fredriksson wrote:
>>
>>> I can't seem to get the [s]tart to work in my extensions...
>>>
>>> - s n i p -
>>> [default]
>>> exten => s,n,Goto(s-${DIALSTATUS},1)
>> The first priority in an extension must be 1 not n.
>
> Actually, I did. I just had it commented out because I didn't understand
> it's use:
>
> exten => s,1,Answer()
> exten => s,n,Goto(s-${DIALSTATUS},1)
This still doesn't make sense because you did not Dial()
before jumping based on ${DIALSTATUS}.
>> exten => _X.,1,Dial(SIP/${EXTEN},20)
>> exten => _X.,n,Goto(s-${DIALSTATUS},1)
>
> Well, I don't want the first part because I want it to return 'invalid
> extension, try again' (Playback(pbx-invalid)) for extensions not
> configured... In other words, I want every single (local) extension
> to be defined.
Then maybe
[default]
exten => _X.,1,Dial(SIP/${EXTEN},20)
exten => _X.,n,Goto(s-${DIALSTATUS},1)
exten => s-ANSWER,1,Hangup()
exten => s-BUSY,1,VoiceMail(${EXTEN},b)
exten => s-NOANSWER,1,VoiceMail(${EXTEN},u)
exten => s-CHANUNAVAIL,1,Playback(pbx-invalid)
exten => _s-.,1,Goto(s-NOANSWER,1)
Or you could define every extension and use a macro. There's an example
in the default extensions.conf file (if you ran `make samples`).
Regards,
Philipp Kempgen
--
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Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
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Re: [asterisk-users] 'Start' in extension rules
Quoting Philipp Kempgen <[EMAIL PROTECTED]>:
> Turbo Fredriksson wrote:
>
>> I can't seem to get the [s]tart to work in my extensions...
>>
>> - s n i p -
>> [default]
>> exten => s,n,Goto(s-${DIALSTATUS},1)
>
> The first priority in an extension must be 1 not n.
Actually, I did. I just had it commented out because I didn't understand
it's use:
exten => s,1,Answer()
exten => s,n,Goto(s-${DIALSTATUS},1)
>> exten => s-BUSY,1,Voicemail(${EXTEN}, b)
>
> Don't put any spaces in the app data.
Darn, forgot that part. I've removed other spaces (in the Dial()'s I
have).
>> exten => 2403,1,Dial(sip/${EXTEN},20,t)
>
> Replace sip by SIP.
Saw that it's case sencetive while I tried other things, so already
fixed.
> exten => _X.,1,Dial(SIP/${EXTEN},20)
> exten => _X.,n,Goto(s-${DIALSTATUS},1)
Well, I don't want the first part because I want it to return 'invalid
extension, try again' (Playback(pbx-invalid)) for extensions not
configured... In other words, I want every single (local) extension
to be defined.
And I guess the second line is 'dependent' on this, the first one
line... ?
> See http://www.the-asterisk-book.com/unstable/voicemail-beispiele.html
Nice one, but can't do it that way...
But it doesn't work even with these changes... Still get:
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/2401-081cea88",
"SIP/2403|20|t") in new stack
[...]
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [EMAIL PROTECTED]:2] Hangup("SIP/2401-081cea88", "") in new
stack
== Spawn extension (default, 2403, 2) exited non-zero on 'SIP/2401-081cea88'
I.e., it still doesn't do the Goto()...
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Re: [asterisk-users] Help 60Hz Hum?
Jay R. Ashworth wrote: > On Fri, Oct 12, 2007 at 11:09:59PM +0200, F6HQZ wrote: > >> Check if you have a ground loop. >> If yes, this is probably the cause of this hum. >> Open the loop. >> > > Actually, hum involving analog POTS lines is usually the result of the > line becoming unbalanced to ground. > Or else running your phone wiring in parallel with and too close to electrical (line voltage) wiring, resulting in induction (crosstalk, as it were). ___ --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] Asterisk 1.4.13 build crashed
Alan Lord wrote: > [CC] chan_zap.c -> chan_zap.o > chan_zap.c: In function ‘process_zap’: > chan_zap.c:11149: internal compiler error: Segmentation fault > Please submit a full bug report, > with preprocessed source if appropriate. > See http://gcc.gnu.org/bugs.html> for instructions. > make[1]: *** [chan_zap.o] Error 1 > make: *** [channels] Error 2 > If I can help further just let me know. For now, I will go back to 1.4.12. As the message says, this is a bug in your compiler, and should be reported to the packager (or in this case, since you are using LFS, directly to the GCC maintainers). I will tell you that we have been building Asterisk 1.4 with GCC 4.2.1 for quite a while without problems on our development systems. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Asterisk Experience" (TM) ___ --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] Help 60Hz Hum?
On Fri, Oct 12, 2007 at 11:09:59PM +0200, F6HQZ wrote: > Check if you have a ground loop. > If yes, this is probably the cause of this hum. > Open the loop. Actually, hum involving analog POTS lines is usually the result of the line becoming unbalanced to ground. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth & Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --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] file.c: File digits/ett does not exist in any format
> "Philipp" == Philipp Kempgen <[EMAIL PROTECTED]> writes: >> files come from? I.e. who recorded them/whos voice it is? Philipp> Only you can tell where you got the sound files you use. I thought they came with Asterisk (v1.4.13).. Sorry, that was a separate package Got the supplier, thanx. ___ --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] Bridging in Asterisk
How do I know when bridging is effective, Francis? A. Anthony Francis <[EMAIL PROTECTED]> wrote: That is brought to you by the sip reinvite, in short yes, unless you set canreinvite = no to either side of that. Apa Minerala wrote: > Am I correct in understanding that if the call comes in g729 and it is > ended in g729 ( by the provider ) , asterisk does only bridging, > therefore using very few CPU ressources ? > > Am I correct in understanding that this "bridging" means that calls ( > rtp ) pass from one provider to another, therefore using low bandwith? > > Thank you > > A. > > > Helping businesses save money worldwide > www.sunapemobil.ca > Numar de acces in .ro : +40318107430 (se taxeaza la pretul unui numar > 031) > Romania 3.5 c/min (USD) > Moldova 10c/min > > > Looking for a deal? Find great prices on flights and hotels > > with Yahoo! FareChase. > > > ___ > --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 -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP ___ --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 - Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'Start' in extension rules
Turbo Fredriksson wrote:
> I can't seem to get the [s]tart to work in my extensions...
>
> - s n i p -
> [default]
> exten => s,n,Goto(s-${DIALSTATUS},1)
> exten => s-BUSY,1,Voicemail(${EXTEN}, b)
>
>
The extensions 's' needs to start with a priority of 1. For example:
exten => s,1,Answer()
exten => s,n,Wait(1)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-BUSY,1,Voicemail(${EXTEN}, b)
Doug
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Re: [asterisk-users] 'Start' in extension rules
On 10/13/07, Turbo Fredriksson <[EMAIL PROTECTED]> wrote: > > Setting debug shows that it skipps the 's' parts... > > Why? because you don't seem to have exten => s,1,something. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'Start' in extension rules
Turbo Fredriksson wrote:
> I can't seem to get the [s]tart to work in my extensions...
>
> - s n i p -
> [default]
> exten => s,n,Goto(s-${DIALSTATUS},1)
The first priority in an extension must be 1 not n.
> exten => s-BUSY,1,Voicemail(${EXTEN}, b)
Don't put any spaces in the app data.
> exten => 2403,1,Dial(sip/${EXTEN},20,t)
Replace sip by SIP.
> exten => _X.,2,Playback(pbx-invalid)
> - s n i p -
> In case I've missunderstood all the docs I've seen
> on the issue, the thing I'm trying to do is have
> the call sent to the voicemail of the extension
> called if the extension is busy. If not, dial the
> extension...
exten => _X.,1,Dial(SIP/${EXTEN},20)
exten => _X.,n,Goto(s-${DIALSTATUS},1)
exten => s-ANSWER,1,Hangup()
exten => s-BUSY,1,VoiceMail(${EXTEN},b)
exten => s-NOANSWER,1,VoiceMail(${EXTEN},u)
exten => _s-.,1,Goto(s-NOANSWER,1)
See http://www.the-asterisk-book.com/unstable/voicemail-beispiele.html
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
That's kinda high then. I wouldn't be happy about that either. You shouldn't be over 30% ever for anything real time. Instantaneous spikes can really start to make your life miserable at that point. -Original Message- From: Erik Anderson [mailto:[EMAIL PROTECTED] Sent: Friday, October 12, 2007 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk? On 10/12/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote: > > I wouldn't be too happy about a system with a > loadavg of 3. The system he mentioned had 8 cores, though. So a load average of 3 is less than 50% usage. -erik ___ --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] Affordable SIP Trunk for Home PBX ?
John Millican wrote: > On Saturday October 13 2007 12:09 pm, Lee Jenkins wrote: > > Be sure to read the fine print as most of the "unlimited" plans do actually > have a limit on usage (even the ones I offer). Some are out in the open some > Then don't advertise it as *unlimited* Seems simple, doesn't it? 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
[asterisk-users] 'Start' in extension rules
I can't seem to get the [s]tart to work in my extensions...
- s n i p -
[default]
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-BUSY,1,Voicemail(${EXTEN}, b)
exten => 2403,1,Dial(sip/${EXTEN},20,t)
exten => _X.,2,Playback(pbx-invalid)
- s n i p -
If I dial '2403' with is off-hook, I don't get
to the voice mail, I get the playback...
Setting debug shows that it skipps the 's' parts...
Why?
In case I've missunderstood all the docs I've seen
on the issue, the thing I'm trying to do is have
the call sent to the voicemail of the extension
called if the extension is busy. If not, dial the
extension...
I've managed to write the same thing, but a LOT more
complex. And I'm going to have about thirty local
extension, and I want it simplified...
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Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?
I used packet 8 with ATA's and a 4 port card. $19.99 a month and works great. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of John covici > Sent: Saturday, 13 October 2007 12:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Affordable SIP Trunk for Home PBX ? > > And my experience with the "unlimited" plans is after a certain point > -- which is sometimes quite obscure -- they start charging -- > sometimes at a rather high rate, so be careful with those. Unlimited > means whatever I want it to mean! > > > on Saturday 10/13/2007 Lee Jenkins([EMAIL PROTECTED]) wrote > > Steve Edwards wrote: > > > On Sat, 13 Oct 2007, Lee Jenkins wrote: > > > > > >> I have been using axVoice.com for some about 9 month to a year now and > > >> their service is pretty damn good. For home users they have unlimited > > >> plan for around 22.00-24.00 U.S. per month. > > > > > > I think the "pay as you go" plans make more sense for most people -- why > > > do you think the vendors push the flat rate plans? > > > > > > At $25.00 per month, you'd have to be on the phone for about an hour a day > > > for it to be cheaper than a $0.015 per minute plan. > > > > > > > True, but I work from home, have a wife and 4 kids with friends and > > family all over the U.S. so it makes more sense for me. > > > > Good point though, Steve. > > > > --- > > > > Lee > > > > > > ___ > > --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 > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > [EMAIL PROTECTED] > > ___ > --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] Affordable SIP Trunk for Home PBX ?
On Saturday October 13 2007 12:09 pm, Lee Jenkins wrote: > Steve Edwards wrote: > > On Sat, 13 Oct 2007, Lee Jenkins wrote: > >> I have been using axVoice.com for some about 9 month to a year now and > >> their service is pretty damn good. For home users they have unlimited > >> plan for around 22.00-24.00 U.S. per month. > > > > I think the "pay as you go" plans make more sense for most people -- why > > do you think the vendors push the flat rate plans? > > > > At $25.00 per month, you'd have to be on the phone for about an hour a > > day for it to be cheaper than a $0.015 per minute plan. > > True, but I work from home, have a wife and 4 kids with friends and > family all over the U.S. so it makes more sense for me. > > Good point though, Steve. > > --- > > Lee Be sure to read the fine print as most of the "unlimited" plans do actually have a limit on usage (even the ones I offer). Some are out in the open some are very well hidden and some others do not even publish the number of minutes that will get you moved to a business rate or possibly even canceled. Think of it from a business perspective, You would not want your clients to use so many minutes that it ended up costing you money, would you? So what the provider has to do is settle on an average of all the customers so that some can use a few more minutes than they pay for and some use less than they pay for, the group in the middle make up the profits. Even under this scenerio there is still a point where the provider starts to loose money. This is why I usually guide customers to a per minute rate so that it is fairer to both sides. Everybody knows what the rules are. Then again there are those that like the convenience of writting the same check every month. 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] Affordable SIP Trunk for Home PBX ?
And my experience with the "unlimited" plans is after a certain point -- which is sometimes quite obscure -- they start charging -- sometimes at a rather high rate, so be careful with those. Unlimited means whatever I want it to mean! on Saturday 10/13/2007 Lee Jenkins([EMAIL PROTECTED]) wrote > Steve Edwards wrote: > > On Sat, 13 Oct 2007, Lee Jenkins wrote: > > > >> I have been using axVoice.com for some about 9 month to a year now and > >> their service is pretty damn good. For home users they have unlimited > >> plan for around 22.00-24.00 U.S. per month. > > > > I think the "pay as you go" plans make more sense for most people -- why > > do you think the vendors push the flat rate plans? > > > > At $25.00 per month, you'd have to be on the phone for about an hour a day > > for it to be cheaper than a $0.015 per minute plan. > > > > True, but I work from home, have a wife and 4 kids with friends and > family all over the U.S. so it makes more sense for me. > > Good point though, Steve. > > --- > > Lee > > > ___ > --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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] Affordable SIP Trunk for Home PBX ?
Steve Edwards wrote: > On Sat, 13 Oct 2007, Lee Jenkins wrote: > >> I have been using axVoice.com for some about 9 month to a year now and >> their service is pretty damn good. For home users they have unlimited >> plan for around 22.00-24.00 U.S. per month. > > I think the "pay as you go" plans make more sense for most people -- why > do you think the vendors push the flat rate plans? > > At $25.00 per month, you'd have to be on the phone for about an hour a day > for it to be cheaper than a $0.015 per minute plan. > True, but I work from home, have a wife and 4 kids with friends and family all over the U.S. so it makes more sense for me. Good point though, Steve. --- Lee ___ --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] file.c: File digits/ett does not exist in any format
Turbo Fredriksson wrote: > Any idea where the current [swedish] > files come from? I.e. who recorded > them/whos voice it is? Only you can tell where you got the sound files you use. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Affordable SIP Trunk for Home PBX ?
On Sat, 13 Oct 2007, Lee Jenkins wrote: > I have been using axVoice.com for some about 9 month to a year now and > their service is pretty damn good. For home users they have unlimited > plan for around 22.00-24.00 U.S. per month. I think the "pay as you go" plans make more sense for most people -- why do you think the vendors push the flat rate plans? At $25.00 per month, you'd have to be on the phone for about an hour a day for it to be cheaper than a $0.015 per minute plan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] file.c: File digits/ett does not exist in any format
> "Anselm" == Anselm Martin Hoffmeister <[EMAIL PROTECTED]> writes: Anselm> You could also copy the file "en.gsm" which should exist Anselm> there over to "ett.gsm" - wrong reading will result, but I Anselm> guess people understand what is meant, like they would Anselm> understand "you have _an_ messages" instead of "one Anselm> message" (details of swedish numbers are a mistery to me Anselm> ;-) Thanx, you got the general idea just perfect :). It WOULD sound like 'an message' :) Anselm> As soon as _any_ file can be played the voicemail will Anselm> work. You could also record a file yourself - if you do Anselm> not mind having a single digit read by a different person Anselm> than the surrounding text. Any idea where the current files come from? I.e. who recorded them/whos voice it is? Solved it with a symlink for now. ___ --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] Combining Flags in Dial()
Jeng Yu wrote: > Hi All, > > I have a quick one for you. Is there a way to mask > (i.e. combine) the flags in the Dial() application? In > other words, a way to do something like > > Dial(Zap/1,10,d|t|f) > > to get the effects of the three flags together in one > shot? I have a need to combine the effects of the "o" > and "A" flags in a dialplan. Just about every example of using Dial anywhere shows you how to combine options. Dial(Zap/1,10,dtf) ___ --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] Affordable SIP Trunk for Home PBX ?
D4rk F1ber wrote: > So I have my asterisk box up and working internally at home and all is > good so far. The next thing I wanted to do was make and recieve calls > to regular land lines now. > > I don't have a POTS line and was looking for probably a SIP trunk. > > I have seen mentions of Skype integration with Asterisk, but does that > include say Skype IN and Skype OUT ? Or is that integration component > really just for being able to contact skype users? > > Looking for the easiest and cheapest way to reach the PSTN, and well > the options out there are plenty regarding SIP trunks, but most tend > to be geared towards businesses for obvious reasons. > > Curious what others are using, and if anyone can make some > recommendations? Not sure if this has been covered already on the > list, and not sure if recommending companies are allowed, so maybe I > need get replies off list? > I have been using axVoice.com for some about 9 month to a year now and their service is pretty damn good. For home users they have unlimited plan for around 22.00-24.00 U.S. per month. --- Warm Regards, Lee ___ --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] file.c: File digits/ett does not exist in any format
Turbo Fredriksson wrote: > I'm using Swedish on version 1.4.13. The full part of the > log is: > > [Oct 13 12:51:16] WARNING[7810] file.c: File digits/ett does not exist in any > format > [Oct 13 12:51:16] WARNING[7810] file.c: Unable to open digits/ett (format 0x8 > (alaw)): No such file or directory > > > The word 'ett' means 'one'. We have two words for one: 'en' and 'ett'. > > Any idea how to fix/solve this problem - can't listen to my > voicemail if I only have one message? It works if I have > 1 > message... You could either create a symlink cd /digits/ && ln -s en.alaw ett.alaw (replace alaw by whatever format your files are in) or record a prompt for ett. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Is there real benefits on a SMP machine for Asterisk?
On Sat, Oct 13, 2007 at 03:31:09PM +0200, Philipp Kempgen wrote:
> Tzafrir Cohen wrote:
>
> > The loadavg is the average number of threads[0] ready to run (or running).
>
> To me it seems that there are important differences between
> systems, especially Linux/Unix, as of which of the states in
> following are counted in:
> - running (i.e. using the CPU)
> - runnable (i.e. waiting for their turn)
> - uninterruptible sleep (i.e. waiting for disk/network I/O)
Load average is for both "running" and "runnable".
The test is simple:
a 100% cpu loop is easy to create (while :; do :; done in bourne
shells such as bash, pick your favorite environment. Run as many as you
want).
Strangely enough in Linux the load avarage includes also processes in a
"uninterruptable sleep" (state "D" in top and ps, as opposed to "S",
which is where processes normally are).
Processes staying long in uninterruptable sleep usually mean a trouble
of some sort.
>
> Without knowing how your kernel calculates the loadavg the
> usefulness of this value is very limited.
>
> > We are all well
> > familiar with a single CPU and single core systems. In those systems
> > only one thread can execute at each time. If the load average is greater
> > than 1 it means that there on the average[1] at least one process
> > waiting for the CPU and not getting executed immediately.
>
> So what you're implying is that only runnable (i.e. waiting)
> threads are counted, not running threads.
Running threads sure are counted.
> The question is if there are any differences on a multi-CPU
> system. Waiting threads would still be waiting threads, no
> matter how many CPUs /could/ run the thread /if they were idle/.
> Is that correct?
>
> The mistake people often seem to make is to assume that
> loadavg == cpu usage.
It is a good indication. Even a better indicaton to the ammount of
threads ("processes") starved for CPU time.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
Tzafrir Cohen wrote: > The loadavg is the average number of threads[0] ready to run (or running). To me it seems that there are important differences between systems, especially Linux/Unix, as of which of the states in following are counted in: - running (i.e. using the CPU) - runnable (i.e. waiting for their turn) - uninterruptible sleep (i.e. waiting for disk/network I/O) Without knowing how your kernel calculates the loadavg the usefulness of this value is very limited. > We are all well > familiar with a single CPU and single core systems. In those systems > only one thread can execute at each time. If the load average is greater > than 1 it means that there on the average[1] at least one process > waiting for the CPU and not getting executed immediately. So what you're implying is that only runnable (i.e. waiting) threads are counted, not running threads. The question is if there are any differences on a multi-CPU system. Waiting threads would still be waiting threads, no matter how many CPUs /could/ run the thread /if they were idle/. Is that correct? The mistake people often seem to make is to assume that loadavg == cpu usage. > So what we really want to know is if a certain thread of Asterisk was > waiting in the run queue too long. Asterisk should not need to wait when > presented with a voice frame to move around. Is there any more direct > way of checking that? Not that I know of. But from your posts on this list I got the impression that your knowledge of what happens inside the kernel is a bit deeper anyway. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] file.c: File digits/ett does not exist in any format
Am Samstag, den 13.10.2007, 15:01 +0200 schrieb Turbo Fredriksson: > I'm using Swedish on version 1.4.13. The full part of the > log is: > > [Oct 13 12:51:16] WARNING[7810] file.c: File digits/ett does not exist in any > format > [Oct 13 12:51:16] WARNING[7810] file.c: Unable to open digits/ett (format 0x8 > (alaw)): No such file or directory > > > The word 'ett' means 'one'. We have two words for one: 'en' and 'ett'. > > Any idea how to fix/solve this problem - can't listen to my > voicemail if I only have one message? It works if I have > 1 > message... The easiest solution "for the moment" is to create a file named "ett.gsm" in the appropriate directory- that should be /usr/share/asterisk/sounds/se/ in your case. You can either create an empty file (with "touch", for example) which will of course result in no number to be read out, which could be annoying. You could also copy the file "en.gsm" which should exist there over to "ett.gsm" - wrong reading will result, but I guess people understand what is meant, like they would understand "you have _an_ messages" instead of "one message" (details of swedish numbers are a mistery to me ;-) As soon as _any_ file can be played the voicemail will work. You could also record a file yourself - if you do not mind having a single digit read by a different person than the surrounding text. Or you could try and find the file "ett.gsm" somewhere; I have no idea why it does not exist, but it probably should. BR Anselm ___ --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] [1.4] lookup_user : specified user 'asterisk' unknown? installing zaptel?
On Fri, Oct 12, 2007 at 03:55:39PM +0200, Vincent wrote: > Hello > > 1. I don't have deep knowledge of either Linux or Asterisk, but I seem > to have successfully installed 1.4 with Zaptel (for support for an > OpenVox PCI FXO card) on a stock Ubuntu 7.04 Server Edition: I assume that this is a X100P clone handled by wcfxo . > > dmesg == > [ 25.990943] Zapata Telephony Interface Registered on major 196 > [ 25.990948] Zaptel Version: 1.4.5.1 > [ 25.990950] Zaptel Echo Canceller: MG2 > > [ 27.523605] ztdummy: RTC rate is 1024 > > [ 34.720147] Zaptel Transcoder support loaded > dmesg == > > One thing though: When I boot up, I see the following error message on > the screen (no trace dmesg or /var/log/messages): > > === > udevd : lookup_user : specified user 'asterisk' unknown > === I suspect Ubuntu have the same udev rules as Debian (but I'm not sure), and thus adding the Zaptel udev rules is simply unnecessary. > > By looking at /etc/udev/rules.d/zaptel.rules, I guess there was a step > missing in the instructions I read on how to compile Zaptel, and > assume I have to add a user/group for "asterisk"? > > 2. More generally, I found missing or possibly outdated information on > how to go and install the Zaptel module to support PCI cards, so I'm > not positive I did everything right: > > === > 1. Compile and install Zaptel: > cd zaptel-1.4.5.1 > ./configure > make clean > make > make zttool > make install > make config "make zttool" is implied by "make" (if you have newt. Otherwise it will fail). > => (here, says "If you have any zaptel hardware it is now recommended > to edit /etc/default/zaptel or /etc/sysconfig/zaptel and set there an > optimal value for the variable MODULES") MODULES="wcfxo" in your case. > > 2. Compile and install Asterisk: > cd /usr/src/asterisk-1.4.2 > ./configure > make clean > make > make install > make samples > make config > > cd /usr/src/asterisk-addons > ./configure > make clean > make > make install > make samples > > 3. create /etc/zaptel.conf > edit /etc/asterisk/zapata.conf > edit /etc/asterisk/extensions.conf > > 4. install ztdummy: > echo "ztdummy" >> /etc/modules > modprobe ztdummy echo "wcfxo" >>/etc/modules No need for ztdummy . 'genzaptelconf -sdvM' would have done that. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] really sorry about this - E1 vs T1
On Sat, Oct 13, 2007 at 04:41:11AM +0200, Philipp Kempgen wrote: > > Yes, see the t1e1override module parameter in wct4xxp/base.c. IIRC, it's > > 0xff to hard code to E1 mode, and set it to 0 for T1 mode. -1 is to use > > the jumper settings. > > ztcfg -vv > will tell you the number of channels, so you can tell whether > the card is in T1 or E1 mode. Actually this will tell you the number of channels you wrote in zaptel.conf . If you got no error, then chances are the configuration is connect. It is simpler to look directly at the representation of the span in under /proc/zaptel . 24 channels means T1 (or J1?) and 31 channels mean E1. This is what genzaptelconf and zapconf do . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] file.c: File digits/ett does not exist in any format
I'm using Swedish on version 1.4.13. The full part of the log is: [Oct 13 12:51:16] WARNING[7810] file.c: File digits/ett does not exist in any format [Oct 13 12:51:16] WARNING[7810] file.c: Unable to open digits/ett (format 0x8 (alaw)): No such file or directory The word 'ett' means 'one'. We have two words for one: 'en' and 'ett'. Any idea how to fix/solve this problem - can't listen to my voicemail if I only have one message? It works if I have > 1 message... ___ --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] Strange behaviour afetr update from 1.2 to 1.4
Il Neofita wrote: > In the extensions I used DIAL(SIP/100&SIP/101,30,tTr) if I receive a > call from an external providers Remove the r 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] Other apps checking Day/Night
On Fri, Oct 12, 2007 at 11:34:39AM -0400, C. Duncan Hudson wrote: > I'm fairly new to Asterisk, so please bear with me if this is silly > question. I'd like to run a script on my server that would take the > "Call now to order" banner off my website automatically when I put my > phone system on night. I can handle the webserver side of things, but I > don't know where to begin on the Asterisk side of things - can a simple > script be run to check the value of the day/night condition, or is that > value written somewhere that I could check or poll? Any help / ideas > are really appreciated. Thanks in advance, If your dialplan does not poll the filesystem / database for each call, consider actively flipping a flag in Asterisk. This can be done by setting a value in the Asterisk database or setting a global variable through the manager interface or or a call file. Someone mentioned poking a hole. Unless you use sudo or something euivalent then yes, the ability to generate a call today allows quite too much. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] [1.4] lookup_user : specified user 'asterisk' unknown? installing zaptel?
On Fri, 12 Oct 2007 09:42:47 -0800, "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]> wrote: >Since you are using the OpenVOX FXO card, don't you need another >module? I'm guessing you'd need wctdm INSTEAD of ztdummy. Thanks. I've seen it mentionned in some articles, but I'm still in the dark at what is needed to run an FXO card. There's a lot of articles on setting up Asterisk, but they're not necessarily complete and/or up-to-date. ___ --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] Asterisk Redundancy
Adrian Marsh wrote: interested in how people are "clustering" Asterisk, if that's possible, or how you might be achieving a redundant solution. > I've a single Asterisk server driving the company. Its well backed-up, and > I've a cloned machine that (in theory) with a DNS change could take over > operations. > > However I'd like to achieve something more automated if possible. The Carrier Class project, which is based on Asterisk and OpenSER, looks promising: http://www.carrierclass.net/ ___ --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] Is there real benefits on a SMP machine for Asterisk?
On Fri, Oct 12, 2007 at 05:29:24PM +0200, Philipp Kempgen wrote: > Atis Lezdins wrote: > > > I have 8-core system that has web interface + sql + java + some other stuff > > running, and at 30 simultenous calls i get loadavg maximum of 3. > > I wouldn't be too happy about a system with a > loadavg of 3. The loadavg is the average number of threads[0] ready to run (or running). In your typical desktop system it is 0, because the CPU is mostly idle. But why do we care so much about the load average? We are all well familiar with a single CPU and single core systems. In those systems only one thread can execute at each time. If the load average is greater than 1 it means that there on the average[1] at least one process waiting for the CPU and not getting executed immediately. Maybe one thread of Asterisk uses the CPU and another thread is waiting for it. So what we really want to know is if a certain thread of Asterisk was waiting in the run queue too long. Asterisk should not need to wait when presented with a voice frame to move around. Is there any more direct way of checking that? [0] Recall that the CPU executes threads, regardless to which process they belong. Two different threads of he same process and two different processes are the same for this discussion. [1] There are three different loadavg numbers: for the last minute, for the last five minutes and for the last 15 minutes. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] AEL2 Syntax Highlighting
On Fri, Oct 12, 2007 at 05:24:29PM -0500, Perssy Llamosas wrote: > Hi, > > I am looking for a syntax highlighter for AEL2. Google is not helping, > so I thought you guys could help me. > > I found this vim syntax highlighter for AEL but it doesn't help if you > want to code in AEL2: > http://vim.sourceforge.net/scripts/script.php?script_id=1900 How is AEL2 syntax different from AEL? Can you please give examples where the above fails for AEL2? (or for AEL, for that matter) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Strange behaviour afetr update from 1.2 to 1.4
Hi, I update from asterisk 1.2 to 1.4 and I have some problems. In the extensions I used DIAL(SIP/100&SIP/101,30,tTr) if I receive a call from an external providers now in 1.4 I recieve only one ring What can I do to solve this problem? ___ --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] aastra 9133i and autoanswer with headset
Just in case anyone is looking at these for their call-centre or similar, you *cannot* put an auto-answer call through to the headset - it always is directed to the speakerphone. This has been confirmed by AAstra support. Obviously, this causes problems in a call centre ! Julian Julian Lyndon-Smith wrote: > I am *really* sorry about hijacking this thread, but the only way I can > post to the -user list is by replying to another thread. (btw, this is > getting really annoying. Please, Digium, sort the filters out!) > > I've added the auto-answer header in my dialplan, and it works great. > However, there is a problem with a headset - If I use a headset, then no > matter what the settings on the phone are, the auto-answer *always* > defaults to the speakerphone rather than the headset. > > Has anyone else encountered this, and have a solution ? > > Many thanks > > Julian > > ___ > --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 > > __ > This email has been scanned by the MessageLabs Email Security System. > For more information please visit http://www.messagelabs.com/email > __ > > __ > This email for dotr.com has been scanned by MessageLabs > __ > > ___ --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] Set up two PSTN calls and then join them
I wish to set up two PSTN calls and then connect them similar to Jajah (is this called 3pcc?). The PSTN interconnect is handled by a third party SIP provider. I can do this using the manager or call files. An example (using php) would be: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret: $strSecret\r\n\r\n"); fputs($oSocket, "Action: originate\r\n"); fputs($oSocket, "Channel: $strChannel\r\n"); fputs($oSocket, "WaitTime: $strWaitTime\r\n"); fputs($oSocket, "CallerId: $strCallerId\r\n"); fputs($oSocket, "Exten: $strExten\r\n"); fputs($oSocket, "Context: $strContext\r\n"); fputs($oSocket, "Priority: $strPriority\r\n\r\n"); fputs($oSocket, "Action: Logoff\r\n\r\n"); This simulates one extension calling another and has some limitations e.g. the CDR records show one call which doesn't meet my requirements for billing purposes. What's the best way to achieve this join up two calls scenario? Any suggestions appreciated. Cameron ___ --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
