Re: [asterisk-users] Asterisk vs c-client issues

[Tzafrir Cohen]

On Fri, Aug 15, 2008 at 12:49:02PM -0700, Lee Lundrigan wrote:

> londo*CLI> module load app_voicemail.so
> [Aug 15 12:45:24] WARNING[14459]: loader.c:363 load_dynamic_module: 
> Error loading module 'app_voicemail.so': 
> /usr/lib/asterisk/modules/app_voicemail.so: cannot restore segment prot 
> after reloc: Permission denied
> [Aug 15 12:45:24] WARNING[14459]: loader.c:657 load_resource: Module 
> 'app_voicemail.so' could not be loaded.

Try: ldd /usr/lib/asterisk/modules/app_voicemail.so

-- 
   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] Maybe a crazy idea, but are there Asterisk hoster outside there?

[Gordon Henderson]

On Sat, 16 Aug 2008, Ronald Wiplinger wrote:

> I used to run an Asterisk server in the office, ... was looking for a
> small replacement. I am not sure if that one is a good idea yet either.

Many people here (myself included :) will tell you it's an excellent idea!

> How about this one:
>
> I have VoIP phones, I have a Welgate 3804 (=2 FXO), all what I need is
> an Asterisk server.
>
> Is there a Asterisk hoster out there? Maybe as a virtual machine?
>
> The mini solution does not have all features, but maybe this would still
> allow me to turn off another machine here.

There are lots of 'hosters' - however you don't say what country you're in 
- your domain & website suggests Taiwan though, so you need to find one 
local to you, as it's no pint myself offering the facility as I'm in the 
UK, nor the multitude of others on the list unless they're based in Taiwan 
themselves!

However, with 2 analogue lines to deal with, unless you can port the 
numbers to a local VoIP provider then you'll probably be better off with a 
PBX local to you with 2 x FXO interfaces (TDM410/OpenVox, etc.) to handle 
those lines.

If you want something small and ready-ish built, look at a miniITX box and 
maybe "pbx in a flash" or trixbox if you don't want to completely roll 
your own...

Gordon

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Re: [asterisk-users] Fwd: USA Lata AreaCode Database

[John Faubion]

> Why everybody charge so much for this information, 
> why this information could not be free ?

Probably because Telecordia makes a decent amount of money from selling it.
If they gave it away that revenue stream would dry up.

John


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Re: [asterisk-users] AMI and extensions.conf

[Tzafrir Cohen]

On Sat, Aug 16, 2008 at 01:46:54AM +, Vadim Lebedev wrote:
> Vadim Lebedev  mbdsys.com> writes:
> 
> > 
> > Tzafrir Cohen  xorcom.com> writes:
> > 
> > > 
> > > On Thu, Aug 14, 2008 at 04:06:27PM +, Vadim Lebedev wrote:
> > > > Hello
> > > > 
> > > > I'm looking for a wayy to modify extensions.conf
> > > > It seems that PutConfig AMI command is not 
> > > > supposed to work on extensionsq.conf
> > > 
> > > It should. Do you have a test case where it doesn't?
> > > 
> 
> > Action: updateconfig
> > reload: no
> > srcfilename: extensions.conf
> > dstfilename:  extensions.conf
> > Action-: append
> > Var-: exten
> > Value-: 999,1,Noop(999)
> > Cat-: ami-test
> > 
> > Response: Success
> > 
> > But the file itself is not modified
> > 
> > ___
> 
> Ok i've fixed the problem (actually there was two of them
> 1) space in after colon in "dstfilename:  extensions.conf"  and

Does this really matter?

> 2) numiric id have to be in XX (6 digit) format

Is there a simple way to find "incorrect headers" in manager messages?
e.g.: a debug message about "header Action- does nothing"?

-- 
   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] asterisk-users Digest, Vol 49, Issue 40

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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Re: [asterisk-users] ANSI terminal colors

[Tilghman Lesher]

On Friday 15 August 2008 22:31:18 Jay R. Ashworth wrote:
> On Fri, Aug 15, 2008 at 07:08:33PM +0200, Philipp Kempgen wrote:
> > True. But lines of varying length with a black background in a
> > white terminal window don't make it any better. Just causes the
> > ragged margins to stand out.
>
> Run it into a file with screen(1l) and use less -r to watch it.  That's
> what I do, anyway...

Or one could simply go try the patch which has been posted for the past
24 hours.

-- 
Tilghman

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[asterisk-users] Maybe a crazy idea, but are there Asterisk hoster outside there?

[Ronald Wiplinger]

I used to run an Asterisk server in the office, ... was looking for a
small replacement. I am not sure if that one is a good idea yet either.

How about this one:

I have VoIP phones, I have a Welgate 3804 (=2 FXO), all what I need is
an Asterisk server.

Is there a Asterisk hoster out there? Maybe as a virtual machine?

The mini solution does not have all features, but maybe this would still
allow me to turn off another machine here.

bye

Ronald

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Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?

[Tilghman Lesher]

On Friday 15 August 2008 22:16:34 Jay R. Ashworth wrote:
> On Fri, Aug 15, 2008 at 02:49:17PM -0500, Tilghman Lesher wrote:
> > > To be more clear, what I'm after is to have *someone else besides me*
> > > place calls out their PRI, and then TBCT those placed calls to my DN.
> > >
> > > By the time the calls get to me, they should just be standard phone
> > > calls.
> > >
> > > So I expect the call-placing-party to need TBCT, but not me.
> > >
> > > > I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are
> > > > capable of TBCT with the current zaptel code-base. Also, the two B
> > > > channels involved in the TBCT have to use the same D channel.
> > >
> > > And I'm probably not concerned with whether Asterisk can deal with
> > > TBCT, because Asterisk probably won't be involved at that stage; just
> > > once the call's transferred to me.
> > >
> > > But before I inquire of said second party whether they *can* do that, I
> > > wanted to confirm it was possible.
> >
> > 2BCT works when the telco originates the call and Asterisk is hairpinning
> > the call back out the same PRI circuit.  However, Asterisk does not
> > support the opposite direction.  That is, a call originated from Asterisk
> > that comes back in via the same PRI circuit cannot be 2BCT.  I'm not
> > certain whether this is a limitation of Asterisk alone or of the
> > protocol, but it cannot be done.
>
> I'm not sure we're not talking at cross purposes, here, Tilghman...
>
> but TBCT is an instruction to an end-office that sent you a call to
> yank it back off your timeslot and forward it along to someone else.
>
> There's no hairpin involved: the point of TBCT is that you tie up *0*
> timeslots instead of 2, to forward a call.

There is a hairpin involved.  The call (for several milliseconds at least) is
using two channels on the PRI before the 2BCT succeeds, and then the call
no longer takes up any channels.  It is only when the PRI detects the hairpin,
through the native bridge code that it is able to detect that the call is
eligible for 2BCT.

> Why would an Asterisk instance call itself on the same span?

Very simple.  Call your main number, and if you don't have special logic
in your internal dialplan context to handle that, the call will go out to the
telco and dutifully come right back in on the same circuit.

> > Similarly, Asterisk cannot complete a 2BCT request, if Asterisk is on the
> > NET side of the PRI circuit.  That might could be added in the future,
> > but it is not supported now.
> >
> > So in summary, Asterisk can request 2BCT, but it cannot perform a 2BCT if
> > requested from the other side.
>
> Nothing can perform a TBCT unless it's a PRI server, not a client; it's
> function of 5ESS's and DMSen; you have to be an SS7 speaker to do it in
> the first case.

I don't think that's the case.  Matt would know more, and his reply suggests
that it certainly would be possible for Asterisk to do this.  SS7 is not at
all required here.

-- 
Tilghman

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Re: [asterisk-users] zaptel timing

[Nhadie]

thank you. would it not limit the number of conference? or maybe the 
number of user in a conference?

regards,
ron

Tzafrir Cohen wrote:
> On Sat, Aug 16, 2008 at 12:52:33AM +0800, Nhadie wrote:
>> Hi,
>>
>> Just wondering if i can use a card just for timing instead of just using 
>> ztdummy? maybe use 2 port fxs or fxo card so it would not be costly.
> 
> Yes, it should work well.
> 
>> will there be any difference on  the meetme performance if i used a 
>> hardware compared to just using ztdummy?
>>
>> is there a limit on conference number using ztdummy? will it be more 
>> limited if i used hardware?
>>
>> i've also read about ztrtc, is this better than ztdummy?
> 
> ztrtc is not needed, as ztdummy can use the RTC.
> 

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Re: [asterisk-users] ANSI terminal colors

[Jay R. Ashworth]

On Fri, Aug 15, 2008 at 07:08:33PM +0200, Philipp Kempgen wrote:
> True. But lines of varying length with a black background in a
> white terminal window don't make it any better. Just causes the
> ragged margins to stand out.

Run it into a file with screen(1l) and use less -r to watch it.  That's
what I do, anyway...

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

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?

[Jay R. Ashworth]

On Fri, Aug 15, 2008 at 03:03:23PM -0500, Matthew Fredrickson wrote:
> Under no circumstances can Asterisk receive a TBCT request.  We just 
> ignore them.  We can initiate them however.
> 
> There are different TBCT implementations, dependent on which switch type 
> is used, with different restrictions associated with each switch type 
> selected.
> 
> For true TBCT (on switchtypes of NI2 and 5ESS, AFAIK), you can have any 
> combination of inbound and/or outbound channels (one inbound/one 
> outbound, two inbound, two outbound) and transfer them to the upstream 
> switch.  The protocol doesn't care.
> 
> For DMS100's version of TBCT, called RLT, one leg *must* be inbound and 
> the other *must* be outbound.  No other combination is going to work. 
> This is explicitly mentioned in the protocol in RLT.

Oddly, I learned about TBCT *from the feature planning guide concerning
the DMS100*, to which I had a subscription 10 years or so ago; I don't
recall it having a different name or limitations.

I can lay my hands on that issue; I will.

But, again, I wasn't concerned with whether Asterisk could do anything
specific with TBCT, except catch calls sent to me by someone else
performing one.

Which my original message was pretty clear on.  :-)

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

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?

[Jay R. Ashworth]

On Fri, Aug 15, 2008 at 02:49:17PM -0500, Tilghman Lesher wrote:
> > To be more clear, what I'm after is to have *someone else besides me*
> > place calls out their PRI, and then TBCT those placed calls to my DN.
> >
> > By the time the calls get to me, they should just be standard phone
> > calls.
> >
> > So I expect the call-placing-party to need TBCT, but not me.
> >
> > > I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are
> > > capable of TBCT with the current zaptel code-base. Also, the two B
> > > channels involved in the TBCT have to use the same D channel.
> >
> > And I'm probably not concerned with whether Asterisk can deal with
> > TBCT, because Asterisk probably won't be involved at that stage; just
> > once the call's transferred to me.
> >
> > But before I inquire of said second party whether they *can* do that, I
> > wanted to confirm it was possible.
> 
> 2BCT works when the telco originates the call and Asterisk is hairpinning
> the call back out the same PRI circuit.  However, Asterisk does not support
> the opposite direction.  That is, a call originated from Asterisk that comes
> back in via the same PRI circuit cannot be 2BCT.  I'm not certain whether this
> is a limitation of Asterisk alone or of the protocol, but it cannot be done.

I'm not sure we're not talking at cross purposes, here, Tilghman...

but TBCT is an instruction to an end-office that sent you a call to
yank it back off your timeslot and forward it along to someone else.

There's no hairpin involved: the point of TBCT is that you tie up *0*
timeslots instead of 2, to forward a call.

Why would an Asterisk instance call itself on the same span?

> Similarly, Asterisk cannot complete a 2BCT request, if Asterisk is on the NET
> side of the PRI circuit.  That might could be added in the future, but it is
> not supported now.
> 
> So in summary, Asterisk can request 2BCT, but it cannot perform a 2BCT if
> requested from the other side.

Nothing can perform a TBCT unless it's a PRI server, not a client; it's
function of 5ESS's and DMSen; you have to be an SS7 speaker to do it in
the first case.

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

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] Basic outbound calling issue : a lot closer

[Brad]

I figure it out, asterisk is using the wrong ip address.
I have bind address set to the correct ip address. How to I force asterisk to 
use the correct ip address?



--- On Fri, 8/15/08, Brad <[EMAIL PROTECTED]> wrote:

> From: Brad <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Basic outbound calling issue : a lot closer
> To: asterisk-users@lists.digium.com
> Date: Friday, August 15, 2008, 9:33 PM
> This what they sent me
> You need to send: 
> - 11-digit originating # (i.e., 1-NPA-NXX-) 
> - 10-digit terminating #
> 
> This got me a lot further in extensions.conf
> 
> exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
> 
> I am getting a 503 error on the phone and asterisk is
> giving me:
> 
>  == Auto fallthrough, channel 'SIP/100-09ef2cc0'
> status is 'CONGESTION'
> -- Executing [EMAIL PROTECTED]:1]
> Dial("SIP/100-09f2ee18",
> "SIP/[EMAIL PROTECTED]|30|r") in new stack
> -- Called [EMAIL PROTECTED]
> -- Got SIP response 503
> "NoCircuitChannelAvailable" back from
> 64.211.41.115
> -- SIP/64.211.41.115-09ef2cc0 is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
>   == Auto fallthrough, channel 'SIP/100-09f2ee18'
> status is 'CONGESTION'
> 
> 
> 
> --- On Fri, 8/15/08, Brad <[EMAIL PROTECTED]> wrote:
> 
> > From: Brad <[EMAIL PROTECTED]>
> > Subject: Re: [asterisk-users] Basic outbound calling
> issue
> > To: asterisk-users@lists.digium.com
> > Cc: "Felippe Silvestre"
> <[EMAIL PROTECTED]>
> > Date: Friday, August 15, 2008, 9:06 PM
> > extensions.conf
> > 
> > [To_Airspring]
> > exten => 55,1,Playback(demo-echotest) ; Let them
> know
> > what's going on
> > exten => 55,2,Echo ; Do the echo test
> > exten => 55,3,Playback(demo-echodone) ; Let them
> know
> > it's over
> > 
> > exten => 100,1,Dial(SIP/100,20)
> > 
> > sip.conf
> > 
> > ;; twinkle softphone
> > [100]
> > user=100
> > nat=yes
> > type=friend
> > secret=andreasd
> > host=dynamic
> > context=To_Airspring
> > 
> > 
> > This should ba all I need
> > 
> > exten => 100,1,Dial(SIP/100,20) should catch it and
> send
> > it to Sip
> > 
> > 
> > --- On Fri, 8/15/08, Felippe Silvestre
> > <[EMAIL PROTECTED]> wrote:
> > 
> > > From: Felippe Silvestre
> > <[EMAIL PROTECTED]>
> > > Subject: RE: [asterisk-users] Basic outbound
> calling
> > issue
> > > To: [EMAIL PROTECTED], "Asterisk Users
> Mailing
> > List - Non-Commercial Discussion"
> > 
> > > Date: Friday, August 15, 2008, 12:25 PM
> > > Check if you have some rule to dial under brad1
> > context
> > > 
> > > dialplan [EMAIL PROTECTED]
> > > 
> > > Regards
> > > 
> > > Felippe Silvestre
> > >  
> > > 
> > > -Original Message-
> > > From: [EMAIL PROTECTED] 
> > > [mailto:[EMAIL PROTECTED]
> On
> > Behalf
> > > Of Brad
> > > Sent: Friday, August 15, 2008 12:09
> > > To: Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > > Subject: [asterisk-users] Basic outbound calling
> issue
> > > 
> > > I am trying to lauch a first outbound call.
> > > I am connected to my telco via a peer which is a
> > little 
> > > different from what I consider the norm.
> > > 
> > > extinsions.conf
> > > 
> > > [To_Bandwidth]
> > > ignorepat => 9
> > > exten => 9,1,Dial(Sip/g2/)
> > > exten => 9,2,Congestion
> > > 
> > > sip.conf
> > > 
> > > [To_Bandwidth]
> > > canreinvite=yes
> > > context=from-pstn
> > > dtmfmode=rfc2833
> > > host=.com
> > > nat=no
> > > outboundproxy=xxx.com
> > > qualify=no
> > > type=peer
> > > 
> > > 
> > > error
> > > 
> > > [Aug 15 11:02:27] NOTICE[25791]: chan_sip.c:14035
> 
> > > handle_request_invite: Call from 'brad1'
> to
> > > extension 
> > > '919544790554' rejected because extension
> not
> > > found.
> > > 
> > > 
> > >   
> > > 
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> > >   
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> > 
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Re: [asterisk-users] dahdi and ztdummy

[Jerry Geis]

Tzafrir Cohen wrote:

>/ That's part of Asteirsk. Not of Zaptel/DAHDI itself . Asterisk has
/>/ renamed chan_zap.so to chan_dahdi.so . It now supports DAHDI . It also
/>/ supports Zap/ for the moment for backward compatibility .
/
And the relevant information is in the Zaptel-to-DAHDI.txt file,
specifically written so that users will know what to expect when
converting from Zaptel to DAHDI.

  

kevin,

Where is this file Zaptel-to-DAHDI.txt?
I have searching svn asterisk, voip-info.org and dahdi linux complete 
and I did not see it.


Jerry
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Re: [asterisk-users] Basic outbound calling issue : a lot closer

[Brad]

I get congestion (same error) with
exten =>  _NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
not dialing 1
exten =>  _1NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
dialing 1
exten =>  _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
dialing 9

All the same

  == Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/100-b7c03ce8", "SIP/[EMAIL 
PROTECTED]|30|r") in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 503 "NoCircuitChannelAvailable" back from 64.211.41.115
-- SIP/64.211.41.115-09f2ee18 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/100-b7c03ce8' status is 'CONGESTION'




--- On Fri, 8/15/08, Brad <[EMAIL PROTECTED]> wrote:

> From: Brad <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Basic outbound calling issue : a lot closer
> To: asterisk-users@lists.digium.com
> Date: Friday, August 15, 2008, 9:33 PM
> This what they sent me
> You need to send: 
> - 11-digit originating # (i.e., 1-NPA-NXX-) 
> - 10-digit terminating #
> 
> This got me a lot further in extensions.conf
> 
> exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
> 
> I am getting a 503 error on the phone and asterisk is
> giving me:
> 
>  == Auto fallthrough, channel 'SIP/100-09ef2cc0'
> status is 'CONGESTION'
> -- Executing [EMAIL PROTECTED]:1]
> Dial("SIP/100-09f2ee18",
> "SIP/[EMAIL PROTECTED]|30|r") in new stack
> -- Called [EMAIL PROTECTED]
> -- Got SIP response 503
> "NoCircuitChannelAvailable" back from
> xxx.xxx.xxx
> -- SIP/xxx.xxx.xxx-09ef2cc0 is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
>   == Auto fallthrough, channel 'SIP/100-09f2ee18'
> status is 'CONGESTION'
> 
> 
> 
> --- On Fri, 8/15/08, Brad <[EMAIL PROTECTED]> wrote:
> 
> > From: Brad <[EMAIL PROTECTED]>
> > Subject: Re: [asterisk-users] Basic outbound calling
> issue
> > To: asterisk-users@lists.digium.com
> > Cc: "Felippe Silvestre"
> <[EMAIL PROTECTED]>
> > Date: Friday, August 15, 2008, 9:06 PM
> > extensions.conf
> > 
> > [To_Airspring]
> > exten => 55,1,Playback(demo-echotest) ; Let them
> know
> > what's going on
> > exten => 55,2,Echo ; Do the echo test
> > exten => 55,3,Playback(demo-echodone) ; Let them
> know
> > it's over
> > 
> > exten => 100,1,Dial(SIP/100,20)
> > 
> > sip.conf
> > 
> > ;; twinkle softphone
> > [100]
> > user=100
> > nat=yes
> > type=friend
> > secret=andreasd
> > host=dynamic
> > context=To_Airspring
> > 
> > 
> > This should ba all I need
> > 
> > exten => 100,1,Dial(SIP/100,20) should catch it and
> send
> > it to Sip
> > 
> > 
> > --- On Fri, 8/15/08, Felippe Silvestre
> > <[EMAIL PROTECTED]> wrote:
> > 
> > > From: Felippe Silvestre
> > <[EMAIL PROTECTED]>
> > > Subject: RE: [asterisk-users] Basic outbound
> calling
> > issue
> > > To: [EMAIL PROTECTED], "Asterisk Users
> Mailing
> > List - Non-Commercial Discussion"
> > 
> > > Date: Friday, August 15, 2008, 12:25 PM
> > > Check if you have some rule to dial under brad1
> > context
> > > 
> > > dialplan [EMAIL PROTECTED]
> > > 
> > > Regards
> > > 
> > > Felippe Silvestre
> > >  
> > > 
> > > -Original Message-
> > > From: [EMAIL PROTECTED] 
> > > [mailto:[EMAIL PROTECTED]
> On
> > Behalf
> > > Of Brad
> > > Sent: Friday, August 15, 2008 12:09
> > > To: Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > > Subject: [asterisk-users] Basic outbound calling
> issue
> > > 
> > > I am trying to lauch a first outbound call.
> > > I am connected to my telco via a peer which is a
> > little 
> > > different from what I consider the norm.
> > > 
> > > extinsions.conf
> > > 
> > > [To_Bandwidth]
> > > ignorepat => 9
> > > exten => 9,1,Dial(Sip/g2/)
> > > exten => 9,2,Congestion
> > > 
> > > sip.conf
> > > 
> > > [To_Bandwidth]
> > > canreinvite=yes
> > > context=from-pstn
> > > dtmfmode=rfc2833
> > > host=.com
> > > nat=no
> > > outboundproxy=xxx.com
> > > qualify=no
> > > type=peer
> > > 
> > > 
> > > error
> > > 
> > > [Aug 15 11:02:27] NOTICE[25791]: chan_sip.c:14035
> 
> > > handle_request_invite: Call from 'brad1'
> to
> > > extension 
> > > '919544790554' rejected because extension
> not
> > > found.
> > > 
> > > 
> > >   
> > > 
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> Arizona
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> > > To UNSUBSCRIBE or update options visit:
> > >   
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
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Re: [asterisk-users] AMI and extensions.conf

[Vadim Lebedev]

Vadim Lebedev  mbdsys.com> writes:

> 
> Tzafrir Cohen  xorcom.com> writes:
> 
> > 
> > On Thu, Aug 14, 2008 at 04:06:27PM +, Vadim Lebedev wrote:
> > > Hello
> > > 
> > > I'm looking for a wayy to modify extensions.conf
> > > It seems that PutConfig AMI command is not 
> > > supposed to work on extensionsq.conf
> > 
> > It should. Do you have a test case where it doesn't?
> > 

> Action: updateconfig
> reload: no
> srcfilename: extensions.conf
> dstfilename:  extensions.conf
> Action-: append
> Var-: exten
> Value-: 999,1,Noop(999)
> Cat-: ami-test
> 
> Response: Success
> 
> But the file itself is not modified
> 
> ___

Ok i've fixed the problem (actually there was two of them
1) space in after colon in "dstfilename:  extensions.conf"  and
2) numiric id have to be in XX (6 digit) format

I've also fixed text here:
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+UpdateConfig

which erronously stated that thi command is unable to handle extensions.conf
agents.conf , etc  (where you have var => value form)

Thanks
Vadim


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Re: [asterisk-users] Fwd: USA Lata AreaCode Database

[Saul Bejarano]

I know what you mean, however it's all about business.
Why to give it free?

Saul

roberto wrote:
> Dear All,
> 
> Why everybody charge so much for this information, why this information
> could not be free ?
> Thanks
> 
> -- Forwarded message --
> From: *Saul Bejarano* <[EMAIL PROTECTED] >
> Date: Fri, Aug 15, 2008 at 3:04 PM
> Subject: Re: [asterisk-users] USA Lata AreaCode Database
> To: asterisk-users@lists.digium.com 
> 
> 
> I got the file for 1999 in space separated value with the NPA, NXX,
> Lat-Long , State and City information, if you want to download it I
> placed it in my ftp server.
> 
> http://www.procomm100.com/files/npanxx99.txt
> 
> Kind regards,
> 
> Saul
> 
> Anthony Messina wrote:
>> On Thursday 14 August 2008 03:09:42 pm roberto wrote:
>>> I'm looking for some "free" LATA X Area Code database.
>>>
>>> Anyone have any idea where can i found?
>>
>> this site has lots of info: http://www.localcallingguide.com/
>>
>>
>>
>> 
>>
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> 
> 
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Re: [asterisk-users] Basic outbound calling issue : a lot closer

[Brad]

This what they sent me
You need to send: 
- 11-digit originating # (i.e., 1-NPA-NXX-) 
- 10-digit terminating #

This got me a lot further in extensions.conf

exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)

I am getting a 503 error on the phone and asterisk is giving me:

 == Auto fallthrough, channel 'SIP/100-09ef2cc0' status is 'CONGESTION'
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/100-09f2ee18", "SIP/[EMAIL 
PROTECTED]|30|r") in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 503 "NoCircuitChannelAvailable" back from 64.211.41.115
-- SIP/64.211.41.115-09ef2cc0 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/100-09f2ee18' status is 'CONGESTION'



--- On Fri, 8/15/08, Brad <[EMAIL PROTECTED]> wrote:

> From: Brad <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Basic outbound calling issue
> To: asterisk-users@lists.digium.com
> Cc: "Felippe Silvestre" <[EMAIL PROTECTED]>
> Date: Friday, August 15, 2008, 9:06 PM
> extensions.conf
> 
> [To_Airspring]
> exten => 55,1,Playback(demo-echotest) ; Let them know
> what's going on
> exten => 55,2,Echo ; Do the echo test
> exten => 55,3,Playback(demo-echodone) ; Let them know
> it's over
> 
> exten => 100,1,Dial(SIP/100,20)
> 
> sip.conf
> 
> ;; twinkle softphone
> [100]
> user=100
> nat=yes
> type=friend
> secret=andreasd
> host=dynamic
> context=To_Airspring
> 
> 
> This should ba all I need
> 
> exten => 100,1,Dial(SIP/100,20) should catch it and send
> it to Sip
> 
> 
> --- On Fri, 8/15/08, Felippe Silvestre
> <[EMAIL PROTECTED]> wrote:
> 
> > From: Felippe Silvestre
> <[EMAIL PROTECTED]>
> > Subject: RE: [asterisk-users] Basic outbound calling
> issue
> > To: [EMAIL PROTECTED], "Asterisk Users Mailing
> List - Non-Commercial Discussion"
> 
> > Date: Friday, August 15, 2008, 12:25 PM
> > Check if you have some rule to dial under brad1
> context
> > 
> > dialplan [EMAIL PROTECTED]
> > 
> > Regards
> > 
> > Felippe Silvestre
> >  
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] On
> Behalf
> > Of Brad
> > Sent: Friday, August 15, 2008 12:09
> > To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> > Subject: [asterisk-users] Basic outbound calling issue
> > 
> > I am trying to lauch a first outbound call.
> > I am connected to my telco via a peer which is a
> little 
> > different from what I consider the norm.
> > 
> > extinsions.conf
> > 
> > [To_Bandwidth]
> > ignorepat => 9
> > exten => 9,1,Dial(Sip/g2/)
> > exten => 9,2,Congestion
> > 
> > sip.conf
> > 
> > [To_Bandwidth]
> > canreinvite=yes
> > context=from-pstn
> > dtmfmode=rfc2833
> > host=.com
> > nat=no
> > outboundproxy=xxx.com
> > qualify=no
> > type=peer
> > 
> > 
> > error
> > 
> > [Aug 15 11:02:27] NOTICE[25791]: chan_sip.c:14035 
> > handle_request_invite: Call from 'brad1' to
> > extension 
> > '919544790554' rejected because extension not
> > found.
> > 
> > 
> >   
> > 
> > ___
> > -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> > 
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register
> > 
> > Now: http://www.astricon.net
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
>   
> 
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Re: [asterisk-users] Basic outbound calling issue

[Brad]

extensions.conf

[To_Airspring]
exten => 55,1,Playback(demo-echotest) ; Let them know what's going on
exten => 55,2,Echo ; Do the echo test
exten => 55,3,Playback(demo-echodone) ; Let them know it's over

exten => 100,1,Dial(SIP/100,20)

sip.conf

;; twinkle softphone
[100]
user=100
nat=yes
type=friend
secret=andreasd
host=dynamic
context=To_Airspring


This should ba all I need

exten => 100,1,Dial(SIP/100,20) should catch it and send it to Sip


--- On Fri, 8/15/08, Felippe Silvestre <[EMAIL PROTECTED]> wrote:

> From: Felippe Silvestre <[EMAIL PROTECTED]>
> Subject: RE: [asterisk-users] Basic outbound calling issue
> To: [EMAIL PROTECTED], "Asterisk Users Mailing List - Non-Commercial 
> Discussion" 
> Date: Friday, August 15, 2008, 12:25 PM
> Check if you have some rule to dial under brad1 context
> 
> dialplan [EMAIL PROTECTED]
> 
> Regards
> 
> Felippe Silvestre
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf
> Of Brad
> Sent: Friday, August 15, 2008 12:09
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Basic outbound calling issue
> 
> I am trying to lauch a first outbound call.
> I am connected to my telco via a peer which is a little 
> different from what I consider the norm.
> 
> extinsions.conf
> 
> [To_Bandwidth]
> ignorepat => 9
> exten => 9,1,Dial(Sip/g2/)
> exten => 9,2,Congestion
> 
> sip.conf
> 
> [To_Bandwidth]
> canreinvite=yes
> context=from-pstn
> dtmfmode=rfc2833
> host=.com
> nat=no
> outboundproxy=xxx.com
> qualify=no
> type=peer
> 
> 
> error
> 
> [Aug 15 11:02:27] NOTICE[25791]: chan_sip.c:14035 
> handle_request_invite: Call from 'brad1' to
> extension 
> '919544790554' rejected because extension not
> found.
> 
> 
>   
> 
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Re: [asterisk-users] dahdi and ztdummy

[Kevin P. Fleming]

Tzafrir Cohen wrote:

> That's part of Asteirsk. Not of Zaptel/DAHDI itself . Asterisk has
> renamed chan_zap.so to chan_dahdi.so . It now supports DAHDI . It also
> supports Zap/ for the moment for backward compatibility .

And the relevant information is in the Zaptel-to-DAHDI.txt file,
specifically written so that users will know what to expect when
converting from Zaptel to DAHDI.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] AMI and extensions.conf

[Vadim Lebedev]

Tzafrir Cohen  xorcom.com> writes:

> 
> On Thu, Aug 14, 2008 at 04:06:27PM +, Vadim Lebedev wrote:
> > Hello
> > 
> > I'm looking for a wayy to modify extensions.conf
> > It seems that PutConfig AMI command is not 
> > supposed to work on extensionsq.conf
> 
> It should. Do you have a test case where it doesn't?
> 

Look at this trace:
Action: login
Username: castel
Secret: castel

Asterisk Call Manager/1.0
Response: Success
Message: Authentication accepted

Action: updateconfig
reload: no
srcfilename: extensions.conf
dstfilename:  extensions.conf
Action-: newcat
Cat-: ami-test

Response: Success

Action: updateconfig
reload: no
srcfilename: extensions.conf
dstfilename:  extensions.conf
Action-: append
Var-: exten
Value-: 999,1,Noop(999)
Cat-: ami-test

Response: Success

But the file itself is not modified



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[asterisk-users] eloqua (was: Re: dahdi link broken)

[Philipp Kempgen]

bkruse schrieb:
> Works for me, try again.

> Jerry Geis wrote:
>> This links: http://downloads.digium.com/pub/telephony/dahdi-linux-complete/
>> appear broken. thy just take me back to /pub
>> nothing downloads.

downloads.digium.com features redirecting links via a shady
marketing company called Eloqua.

Grüße,
Philipp Kempgen
-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998

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Re: [asterisk-users] Problems assigning mISDN Trunk using the DIGIUM Asterisk GUI

[Klaus Ruebsam]

Bk,

Thanks a lot for this immediate fix. mISDN trunk config now works like a
charm! 

Best regards,

Klaus

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von bkruse
Gesendet: Freitag, 15. August 2008 21:43
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Problems assigning mISDN Trunk using the
DIGIUM Asterisk GUI

Now this, even though not on the bug tracker, is a great example of a bug
report.

Because of that error code, I was able to easily fix the bug.

Fixed in revision 3671

-bk

Klaus Ruebsam wrote:
>
> Hi list!
>
> The topic-line already describes it all. I´m in the process of setting 
> up Asterisk with mISDN.
> versions used: Asterisk: 1.4.21.2, Asterisk GUI: 2.0 as obtained from 
> DIGIUM just today, mISDN: 1.1.7.2, OS: Debian 4.01r3 (Kernel 2.6.18 ).
>
> With the GUI I can not manage to set up mISDN trunks. I ticked "mISDN 
> Config" then it says "You donot have any mISDN trunks defined". So I 
> klicked on "Add". The Dialog-box appears asking for "TrunkName" and 
> "Port(s)". But whatever I enter in these fields the "Update"-function 
> does not take it.
>
> Using Internet Explorer as a browser it even complains about an error 
> in Line 429, character 3 with cause "´c´is undefined". FireFox does 
> not prompt for any errors with the code.
>
> Any ideas?
>
> Thanks in advance,
>
> Klaus
>
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[asterisk-users] Asterisk AGI and php problem....

[Gerard A. Matthew]

I am having a difficulty with AGI and PHP. The following is my script 
agi file


#!/usr/bin/php
verbose("$argv[1]\n");
?> 


However, my Asterisk CLI returns this


AGI Tx >> agi_request: cid-to-acct.php
AGI Tx >> agi_channel: SIP/5073-0821eda0
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1218834584.3
AGI Tx >> agi_callerid: 5305
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: *86
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: a2billing
AGI Tx >> agi_extension: *86
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode: 5073
AGI Tx >>
AGI Rx << verbose "Failed to execute 
'/var/lib/asterisk/agi-bin/cid-to-acct.php': No such file or directory" 2
 ==  cid-to-acct.php: Failed to execute 
'/var/lib/asterisk/agi-bin/cid-to-acct.php': No such file or directory

AGI Tx >> 200 result=1


I have both #!/usr/binphp and #!/usr/bin/php5 tried out with the same 
errortried searching online with no help.


Any ideas?
begin:vcard
fn:Gerard A. Matthew
n:Matthew;Gerard A.
email;internet:[EMAIL PROTECTED]
tel;home:1 (206) 203-7608
tel;cell:1 (940) 337-3739
note;quoted-printable:DM Tel: +1 (767) 440-3940 ext 2103=0D=0A=
	US Tel: +1 (206) 203-7608=0D=0A=
	VoIP: [EMAIL PROTECTED]
	MSN: [EMAIL PROTECTED]
	Y!: calibishie2002=0D=0A=
	=0D=0A=
	CONFIDENTIALITY NOTICE:  This electronic message contains information whi=
	ch may be legally confidential and/or privileged.  The information is int=
	ended solely for the individual or entity named above and access by anyon=
	e else is unauthorized.  If you are not the intended recipient, any discl=
	osure, copying, distribution, or use of the contents of this informationi=
	s prohibited and may be unlawful.  If you have received this electronictr=
	ansmission in error, please reply immediately to the sender that youhave=
	 received the message in error, and delete it.
version:2.1
end:vcard

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Re: [asterisk-users] dahdi link broken

[bkruse]

Works for me, try again.

-bk

Jerry Geis wrote:
> This links: http://downloads.digium.com/pub/telephony/dahdi-linux-complete/
> appear broken. thy just take me back to /pub
> nothing downloads.
>
> Jerry
>
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[asterisk-users] dahdi link broken

[Jerry Geis]

This links: http://downloads.digium.com/pub/telephony/dahdi-linux-complete/
appear broken. thy just take me back to /pub
nothing downloads.

Jerry

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Re: [asterisk-users] SIP Callerid Question

[Philipp Kempgen]

Adam Robins schrieb:
> I have two Asterisk 1.2 boxes across a WAN.  Calls between them are sent
> via SIP g729a.   The issue is that the original calleridnum is
> overwritten by the value of the "fromuser" parameter in sip.conf on the
> originating server.  Is there any way to preserve the original
> calleridnum value?

Don't use fromuser?

Grüße,
Philipp Kempgen
-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998

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Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?

[Matthew Fredrickson]

Tilghman Lesher wrote:
> On Friday 15 August 2008 13:45:11 Jay R. Ashworth wrote:
>> On Fri, Aug 15, 2008 at 02:37:46PM -0400, Matt Florell wrote:
>>> Most carrier sales people don't know what TBCT is unfortunately, and
>>> even if a carrier is capable of doing it, it is a possiblity that not
>>> all of their equipment is capable of doing it. One client of mine
>>> tried to get TBCT working across all 16 of their PRIs(all on the same
>>> carrier) and it only worked on 4 of them, supposedly because not all
>>> of the telco equipment was capable of the feature.
>> I expect to fight this battle, yes.  :-)
>>
>>> This actually depends on the kind of PRI service you have. For
>>> instance with DMS100 circuits you can only do TBCT with calls that
>>> come in to your circuit, not with outgoing calls.
>>>
>>> As for connecting two incoming calls, since that is not possible in
>>> Asterisk(to natively bridge two incoming calls together) I can't see
>>> how you would get that to work even if it is possible in TBCT.
>> To be more clear, what I'm after is to have *someone else besides me*
>> place calls out their PRI, and then TBCT those placed calls to my DN.
>>
>> By the time the calls get to me, they should just be standard phone
>> calls.
>>
>> So I expect the call-placing-party to need TBCT, but not me.
>>
>>> I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are
>>> capable of TBCT with the current zaptel code-base. Also, the two B
>>> channels involved in the TBCT have to use the same D channel.
>> And I'm probably not concerned with whether Asterisk can deal with
>> TBCT, because Asterisk probably won't be involved at that stage; just
>> once the call's transferred to me.
>>
>> But before I inquire of said second party whether they *can* do that, I
>> wanted to confirm it was possible.
> 
> 2BCT works when the telco originates the call and Asterisk is hairpinning
> the call back out the same PRI circuit.  However, Asterisk does not support
> the opposite direction.  That is, a call originated from Asterisk that comes
> back in via the same PRI circuit cannot be 2BCT.  I'm not certain whether this
> is a limitation of Asterisk alone or of the protocol, but it cannot be done.
> 
> Similarly, Asterisk cannot complete a 2BCT request, if Asterisk is on the NET
> side of the PRI circuit.  That might could be added in the future, but it is
> not supported now.
> 
> So in summary, Asterisk can request 2BCT, but it cannot perform a 2BCT if
> requested from the other side.
> 


Let me clarify some of this.

Under no circumstances can Asterisk receive a TBCT request.  We just 
ignore them.  We can initiate them however.

There are different TBCT implementations, dependent on which switch type 
is used, with different restrictions associated with each switch type 
selected.

For true TBCT (on switchtypes of NI2 and 5ESS, AFAIK), you can have any 
combination of inbound and/or outbound channels (one inbound/one 
outbound, two inbound, two outbound) and transfer them to the upstream 
switch.  The protocol doesn't care.

For DMS100's version of TBCT, called RLT, one leg *must* be inbound and 
the other *must* be outbound.  No other combination is going to work. 
This is explicitly mentioned in the protocol in RLT.

Hope that helps a bit.

Matthew Fredrickson
Digium, Inc.



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Re: [asterisk-users] SIP Callerid Question

[Adam Robins]

Nevermind, I just answered my own question.  Used "username" instead of
"fromuser".

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Friday, August 15, 2008 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP Callerid Question

 

I have two Asterisk 1.2 boxes across a WAN.  Calls between them are sent
via SIP g729a.   The issue is that the original calleridnum is
overwritten by the value of the "fromuser" parameter in sip.conf on the
originating server.  Is there any way to preserve the original
calleridnum value?  Calleridname is not affected.  I suppose I could
concatenate the number into the name field and then parse it out at the
other end, but . . . 

I know this issue has been around for a while, and is documented.  I'm
wondering if anything has changed or there are any new solutions:

Thanks,

Adam

 

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Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?

[Tilghman Lesher]

On Friday 15 August 2008 13:45:11 Jay R. Ashworth wrote:
> On Fri, Aug 15, 2008 at 02:37:46PM -0400, Matt Florell wrote:
> > Most carrier sales people don't know what TBCT is unfortunately, and
> > even if a carrier is capable of doing it, it is a possiblity that not
> > all of their equipment is capable of doing it. One client of mine
> > tried to get TBCT working across all 16 of their PRIs(all on the same
> > carrier) and it only worked on 4 of them, supposedly because not all
> > of the telco equipment was capable of the feature.
>
> I expect to fight this battle, yes.  :-)
>
> > This actually depends on the kind of PRI service you have. For
> > instance with DMS100 circuits you can only do TBCT with calls that
> > come in to your circuit, not with outgoing calls.
> >
> > As for connecting two incoming calls, since that is not possible in
> > Asterisk(to natively bridge two incoming calls together) I can't see
> > how you would get that to work even if it is possible in TBCT.
>
> To be more clear, what I'm after is to have *someone else besides me*
> place calls out their PRI, and then TBCT those placed calls to my DN.
>
> By the time the calls get to me, they should just be standard phone
> calls.
>
> So I expect the call-placing-party to need TBCT, but not me.
>
> > I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are
> > capable of TBCT with the current zaptel code-base. Also, the two B
> > channels involved in the TBCT have to use the same D channel.
>
> And I'm probably not concerned with whether Asterisk can deal with
> TBCT, because Asterisk probably won't be involved at that stage; just
> once the call's transferred to me.
>
> But before I inquire of said second party whether they *can* do that, I
> wanted to confirm it was possible.

2BCT works when the telco originates the call and Asterisk is hairpinning
the call back out the same PRI circuit.  However, Asterisk does not support
the opposite direction.  That is, a call originated from Asterisk that comes
back in via the same PRI circuit cannot be 2BCT.  I'm not certain whether this
is a limitation of Asterisk alone or of the protocol, but it cannot be done.

Similarly, Asterisk cannot complete a 2BCT request, if Asterisk is on the NET
side of the PRI circuit.  That might could be added in the future, but it is
not supported now.

So in summary, Asterisk can request 2BCT, but it cannot perform a 2BCT if
requested from the other side.

-- 
Tilghman

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Re: [asterisk-users] Asterisk vs c-client issues

[Lee Lundrigan]

Russell Bryant wrote:
> Lee Lundrigan wrote:
>   
>> Hi everyone,
>>
>> Are there any incompatibility issues between asterisk and the c-client 
>> using SSL?
>> When I enable SSL I get the error:
>> *pbx.c:1832 pbx_extension_helper: No application 'VoiceMailMain'
>> *whenever I am trying to access voicemail.
>>
>> But when SSL is disabled everything works great, just like its supposed 
>> to with imap.
>>
>> Any ideas?
>> 
>
> It looks like app_voicemail is failing to load when you build c-client 
> with SSL support.  Try running "module load app_voicemail.so" from the 
> Asterisk CLI to see what the error message is.
>
>   
This is everything:

londo*CLI> module load app_voicemail.so
[Aug 15 12:45:24] WARNING[14459]: loader.c:363 load_dynamic_module: 
Error loading module 'app_voicemail.so': 
/usr/lib/asterisk/modules/app_voicemail.so: cannot restore segment prot 
after reloc: Permission denied
[Aug 15 12:45:24] WARNING[14459]: loader.c:657 load_resource: Module 
'app_voicemail.so' could not be loaded.

Thanks for the Help!

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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 39

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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Re: [asterisk-users] Problems assigning mISDN Trunk using the DIGIUM Asterisk GUI

[bkruse]

Now this, even though not on the bug tracker, is a great example of a 
bug report.

Because of that error code, I was able to easily fix the bug.

Fixed in revision 3671

-bk

Klaus Ruebsam wrote:
>
> Hi list!
>
> The topic-line already describes it all. I´m in the process of setting 
> up Asterisk with mISDN.
> versions used: Asterisk: 1.4.21.2, Asterisk GUI: 2.0 as obtained from 
> DIGIUM just today, mISDN: 1.1.7.2, OS: Debian 4.01r3 (Kernel 2.6.18 ).
>
> With the GUI I can not manage to set up mISDN trunks. I ticked "mISDN 
> Config" then it says "You donot have any mISDN trunks defined". So I 
> klicked on "Add". The Dialog-box appears asking for "TrunkName" and 
> "Port(s)". But whatever I enter in these fields the "Update"-function 
> does not take it.
>
> Using Internet Explorer as a browser it even complains about an error 
> in Line 429, character 3 with cause "´c´is undefined". FireFox does 
> not prompt for any errors with the code.
>
> Any ideas?
>
> Thanks in advance,
>
> Klaus
>
> 
>
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[asterisk-users] SIP Callerid Question

[Adam Robins]

I have two Asterisk 1.2 boxes across a WAN.  Calls between them are sent
via SIP g729a.   The issue is that the original calleridnum is
overwritten by the value of the "fromuser" parameter in sip.conf on the
originating server.  Is there any way to preserve the original
calleridnum value?  Calleridname is not affected.  I suppose I could
concatenate the number into the name field and then parse it out at the
other end, but . . . 

I know this issue has been around for a while, and is documented.  I'm
wondering if anything has changed or there are any new solutions:

Thanks,

Adam

 

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[asterisk-users] Problems assigning mISDN Trunk using the DIGIUM Asterisk GUI

[Klaus Ruebsam]

Hi list!

The topic-line already describes it all. I´m in the process of setting up
Asterisk with mISDN.
versions used: Asterisk: 1.4.21.2, Asterisk GUI: 2.0 as obtained from DIGIUM
just today, mISDN: 1.1.7.2, OS: Debian 4.01r3 (Kernel 2.6.18 ).

With the GUI I can not manage to set up mISDN trunks. I ticked "mISDN
Config" then it says "You donot have any mISDN trunks defined". So I klicked
on "Add". The Dialog-box appears asking for "TrunkName" and "Port(s)". But
whatever I enter in these fields the "Update"-function does not take it. 

Using Internet Explorer as a browser it even complains about an error in
Line 429, character 3 with cause "´c´is undefined". FireFox does not prompt
for any errors with the code.

Any ideas?

Thanks in advance,

Klaus

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Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?

[Jay R. Ashworth]

On Fri, Aug 15, 2008 at 02:37:46PM -0400, Matt Florell wrote:
> Most carrier sales people don't know what TBCT is unfortunately, and
> even if a carrier is capable of doing it, it is a possiblity that not
> all of their equipment is capable of doing it. One client of mine
> tried to get TBCT working across all 16 of their PRIs(all on the same
> carrier) and it only worked on 4 of them, supposedly because not all
> of the telco equipment was capable of the feature.

I expect to fight this battle, yes.  :-)

> This actually depends on the kind of PRI service you have. For
> instance with DMS100 circuits you can only do TBCT with calls that
> come in to your circuit, not with outgoing calls.
> 
> As for connecting two incoming calls, since that is not possible in
> Asterisk(to natively bridge two incoming calls together) I can't see
> how you would get that to work even if it is possible in TBCT.

To be more clear, what I'm after is to have *someone else besides me*
place calls out their PRI, and then TBCT those placed calls to my DN.

By the time the calls get to me, they should just be standard phone
calls.

So I expect the call-placing-party to need TBCT, but not me.

> I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are
> capable of TBCT with the current zaptel code-base. Also, the two B
> channels involved in the TBCT have to use the same D channel.

And I'm probably not concerned with whether Asterisk can deal with
TBCT, because Asterisk probably won't be involved at that stage; just
once the call's transferred to me.

But before I inquire of said second party whether they *can* do that, I
wanted to confirm it was possible.

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

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] Fwd: USA Lata AreaCode Database

[Alex Balashov]

On Fri, August 15, 2008 2:31 pm, roberto wrote:

> Why everybody charge so much for this information, why this information
> could not be free ?

Welcome to incumbent communication industry in the US - and, to a some
extent, pretty much anywhere.  Everything is ridiculously expensive,
viciously proprietary, and extremely obscurantist and abstruse.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599


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Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?

[Matt Florell]

On 8/15/08, Don Kelly <[EMAIL PROTECTED]> wrote:
> 1. The carrier you are connected to must be licensed for it and have the
>  necessary software, if the carrier requires, your circuit(s) must be
>  provisioned for it. The originating/destination carriers shouldn't matter.

Most carrier sales people don't know what TBCT is unfortunately, and
even if a carrier is capable of doing it, it is a possiblity that not
all of their equipment is capable of doing it. One client of mine
tried to get TBCT working across all 16 of their PRIs(all on the same
carrier) and it only worked on 4 of them, supposedly because not all
of the telco equipment was capable of the feature.

>  2. Both incoming and outgoing calls can be transferred to a second outgoing
>  call; I think it's theoretically possible to connect two incoming calls, but
>  I haven't done that.

This actually depends on the kind of PRI service you have. For
instance with DMS100 circuits you can only do TBCT with calls that
come in to your circuit, not with outgoing calls.

As for connecting two incoming calls, since that is not possible in
Asterisk(to natively bridge two incoming calls together) I can't see
how you would get that to work even if it is possible in TBCT.

I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are
capable of TBCT with the current zaptel code-base. Also, the two B
channels involved in the TBCT have to use the same D channel.


MATT---

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[asterisk-users] Fwd: USA Lata AreaCode Database

[roberto]

Dear All,

Why everybody charge so much for this information, why this information
could not be free ?
Thanks

-- Forwarded message --
From: Saul Bejarano <[EMAIL PROTECTED]>
Date: Fri, Aug 15, 2008 at 3:04 PM
Subject: Re: [asterisk-users] USA Lata AreaCode Database
To: asterisk-users@lists.digium.com


I got the file for 1999 in space separated value with the NPA, NXX,
Lat-Long , State and City information, if you want to download it I
placed it in my ftp server.

http://www.procomm100.com/files/npanxx99.txt

Kind regards,

Saul

Anthony Messina wrote:
> On Thursday 14 August 2008 03:09:42 pm roberto wrote:
>> I'm looking for some "free" LATA X Area Code database.
>>
>> Anyone have any idea where can i found?
>
> this site has lots of info: http://www.localcallingguide.com/
>
>
>
> 
>
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Re: [asterisk-users] Asterisk stress call test

[Saul Bejarano]

The term DIE is not correct sorry, it will be better say CRASH if that 
works for you, if you stress call an asterisk the moment you reach the 
treshold of that stress test the application will stop responding, 
initially it will start popping errors on the log then it will just stop 
responding having to reboot the asterisk session.
The one crashing is asterisk as an application not the box itself.


Now, is there something you have to help this user? or it is just about 
making a point?


Al Baker wrote:
> 
> Saul Bejarano wrote:
>> Remember the rule of 30Mhz per call when you kill the machine and also 
>> the bandwidth usage on connected calls.
>>
>> Kind regards,
>>
>> Saul Bejarano
>>
>> aby azid wrote:
>>   
>>> Hi everyone,
>>>
>>> I'm required to make  a stress call on Asterisk server ( > 2000 calls 
>>> per seconds). Are there tools for me to do this sort of test. I was 
>>> thinking of sending loads of Asterisk call files simultaneously 
>>> (starting with 100 call files). Really appreciate if anyone can come up 
>>> with ideas or tools for me to achieve this.
>>>
>>> Cheers,
>>> Aby Azid
>>> Vyke Asia
>>> 
> Where did you get the "Rule" of 30Mhz per call ???
> Wouldn't this be highly dependent on whether it to TDM over a T1 or 
> whether it was in SIP , and which CODEC it was using.
> 
> And why would a properly configured machine "Die", have a HIGH Load 
> Average - YesDIED - Sounds like WinBlows to me
> 
> Yoa - Aby - You need to define your test scenario more fully.  Are you 
> making a call OUT of the box , into the box,  a MIX
> How Long are the calls ?
> Net, net , how many simultaneous call are you going to have ?
> How man Call Originations are there ?
> How may Call Answers ?
> 
> 
> 
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Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?

[Don Kelly]

1. The carrier you are connected to must be licensed for it and have the
necessary software, if the carrier requires, your circuit(s) must be
provisioned for it. The originating/destination carriers shouldn't matter.

2. Both incoming and outgoing calls can be transferred to a second outgoing
call; I think it's theoretically possible to connect two incoming calls, but
I haven't done that.

3. This may rely on your carrier. My carrier allows me to include anything I
choose as outbound ANI--I don't abuse it.

4. To the best of my knowledge, the originating caller and destination can
be anywhere in the world.

Your scenario sounds workable to me.

My experience (non-Asterisk) is using NI2 on DMS and 5E switches in North
America. Carrier personnel are generally unfamiliar with TBCT and your
initial installation will probably involve a little frustration.

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651 842-1000
888 Don Kell(y)
651 842-1001 fax

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay R.
Ashworth
Sent: Friday, August 15, 2008 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] PRI TBCT - Practical Experience, Anybody?

I may have to do some work with TBCT, and probably cross-carrier TBCT,
here shortly, and I haven't ever worked with it.  If anyone on the list
ever has, I'd be interested to know:

1) Only the carrier first involved with the call has to
actually be provisioned for it, correct?

2) Both incoming and outgoing calls can be TBCT'd?

3) If a placed call is transferred to me via TBCT, can I get
the DN of the original target call sent to me as CNID?

4) Does it, in fact, matter if the call placer, and the TBCT
target, are on the same IXC?

I want someone to place calls for me, talk to the people for a
while, and then do an unsupervised transfer to me wherein I can
capture the other party's number off the call itself and feed
the calls into my VICIdial/Asterisk instance.

All my incoming lines are Zap/PRI.

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


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Re: [asterisk-users] USA Lata AreaCode Database

[Saul Bejarano]

I got the file for 1999 in space separated value with the NPA, NXX, 
Lat-Long , State and City information, if you want to download it I 
placed it in my ftp server.

http://www.procomm100.com/files/npanxx99.txt

Kind regards,

Saul

Anthony Messina wrote:
> On Thursday 14 August 2008 03:09:42 pm roberto wrote:
>> I'm looking for some "free" LATA X Area Code database.
>>
>> Anyone have any idea where can i found?
> 
> this site has lots of info: http://www.localcallingguide.com/
> 
> 
> 
> 
> 
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Re: [asterisk-users] DID's needed for Reston Virginia - + hosted asterisk

[Saul Bejarano]

I have worked with a company out of Virginia, IPCOMMS www.ipcomms.net, 
Donald Handsil, you may want to shoot him an email, quality good, 
experience with asterisk and DID from all over the country.

[EMAIL PROTECTED]

Kind regards,

Saul Bejarano

Dean Collins wrote:
> I’ve just started consulting for a SME client based in Reston Virginia.
> 
>  
> 
> They don’t know it yet but they are going to need a hosted asterisk 
> service and some DID’s.
> 
>  
> 
> Email me if you are able to provide 10 DID’s in Reston (must be able to 
> be ported away!!) and hosted Asterisk with end user configurable IVR 
> etc. Probably only 5-8 users at the moment BUT… they’ll be growing real 
> soon now. Will also need some Polycom handsets and  laptop softphone 
> extensions etc.
> 
>  
> 
> Email me your pricing and marketing etc in the next 24-48 hours and I’ll 
> propose my choice on Monday.
> 
>  
> 
>  
> 
>  
> 
>  
> 
> Regards,
> 
> Dean Collins
> [EMAIL PROTECTED] 
> 
> +1-212-203-4357 (New York)
> +61-2-9016-5642 (Sydney)
> http://www.Cognation.net 
> 
>  
> 
> 
> 
> 
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Re: [asterisk-users] asterisk realtime and creating "new" contexts

[Saul Bejarano]

I took it out of the field name used by FreePBX on the database.
If you install FreePBX you can see it on the mysql table list.

Saul

Todd Fulton wrote:
> hi saul,
> 
> could you give me more info on the "VMX-CONTEXT" concept?  i tried to 
> google it, but could find nothing.
> 
> i am trying to do exactly what you state in terms of creating a virtual 
> slice of the box for each user.  thanks!
> 
> 
> todd
> 
>  Original Message 
> Subject: Re: [asterisk-users] asterisk realtime and creating "new"
> contexts
> From: Saul Bejarano <[EMAIL PROTECTED]>
> Date: Thu, August 14, 2008 8:01 pm
> To: asterisk-users@lists.digium.com
> 
> 
> The VMX-CONTEXT can be managed from database
> 
> mysql> select * from globals;
> But you will have to specify the same on each extension under sip
> once the extension under table sip is the one that calls the CONTEXT to
> route the call.
> 
> | PARKNOTIFY | SIP/200 |
> | RECORDEXTEN | "" |
> | RINGTIMER | 15 |
> | DIRECTORY | last |
> | AFTER_INCOMING | |
> | IN_OVERRIDE | forcereghours |
> | REGTIME | 7:55-17:05 |
> | REGDAYS | mon-fri |
> | DIRECTORY_OPTS | |
> | DIALOUTIDS | 1/2/3/4/5/6/ |
> | OUT_1 | ZAP/g0 |
> | VM_PREFIX | * |
> | VM_OPTS | |
> | VM_GAIN | |
> | VM_DDTYPE | u |
> | TIMEFORMAT | kM |
> | TONEZONE | us |
> | ALLOW_SIP_ANON | yes |
> | VMX_CONTEXT | from-internal
> 
> It will make it sort of complicated thought because it was build to be
> the generic element routing the calls out of the SIP Registrar, by
> having individual Context what you are trying to do is partition one
> Asterisk box to function as multiple offering invidual termination to
> each one of your users or complete management of a virtual slice of the
> box :)
> 
> Todd Fulton wrote:
>  > Hi,
>  >
>  > I'm trying to create a multi-tennant asterisk installation 
> where
>  > each of my customers has its own context. Well, I've got asterisk
>  > realtime working, and I can add/update extensions to existing
> contexts
>  > in extensions.conf without a problem. However, when I attempt to
> create
>  > database entries with a context that is NOT in extensions.conf, I
> get an
>  > error "invalid extension".
>  >
>  > I've found several posts around the net asking this question, but no
>  > answers. Has anyone out there dealt with this problem?
>  >
>  > Any help would be great!
>  >
>  >
>  > Todd
>  >
>  >
>  >
> 
>  >
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> 
> 
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Re: [asterisk-users] Asterisk stress call test

[Saul Bejarano]

Hi Atis:
Based on your experience.
How many calls can be handled by a single Pentium 3.0Ghz processor on a 
2GB RAM machine spining a 10Krpm disk?

Thanks for the repply, great tool.

Atis Lezdins wrote:
> On Fri, Aug 15, 2008 at 8:31 AM, aby azid <[EMAIL PROTECTED]> wrote:
>> Hi everyone,
>>
>> I'm required to make  a stress call on Asterisk server ( > 2000 calls per
>> seconds). Are there tools for me to do this sort of test. I was thinking of
>> sending loads of Asterisk call files simultaneously (starting with 100 call
>> files). Really appreciate if anyone can come up with ideas or tools for me
>> to achieve this.
>>
> 
> Hi,
> 
> I've written test framework, you'll need another machine with Asterisk
> (+php) on it to generate calls. It allows to write scripts in PHP to
> emulate random customer actions, etc.. You can download it here
> http://ftp.iq-labs.net/pbx-test/
> If you find it useful, or get into some problems, don't hesitate to write me.
> 
> If you need just bunch of identical calls, you may also try out SIPp.
> 
> Regards,
> Atis
> 
> 
> 
> 


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Re: [asterisk-users] zaptel timing

[Steve Totaro]

On Fri, Aug 15, 2008 at 12:52 PM, Nhadie <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Just wondering if i can use a card just for timing instead of just using
> ztdummy? maybe use 2 port fxs or fxo card so it would not be costly.
>
> will there be any difference on  the meetme performance if i used a
> hardware compared to just using ztdummy?
>
> is there a limit on conference number using ztdummy? will it be more
> limited if i used hardware?
>
> i've also read about ztrtc, is this better than ztdummy?
>
> thank you
>
> regards,
> ron
>

If you just need zaptel timing and do not care about FXP/FXO ports,
you can just use a TDM400P card with not modules, just the card
itself.

This is usually what do for boxen to avoid using ztdummy.  Then it is
simple enough to add modules as you needs change.

Thanks,
Steve Totaro

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Re: [asterisk-users] Asterisk stress call test

[Saul Bejarano]

Well lets go by parts:
In regards to your question about the CPU processor usage I am 
evaluating the scenario posted by the customer which is not at all the 
generic one you are talking about.
He wants STRESS CALL TEST which is SIP to SIP based on the software he 
is trying to use:

http://www.voip-info.org/wiki/view/Asterisk+dimensioning
You can find answers to your first question here, please follow the link.

Al Baker wrote:
> 
> Saul Bejarano wrote:
>> Remember the rule of 30Mhz per call when you kill the machine and also 
>> the bandwidth usage on connected calls.
>>
>> Kind regards,
>>
>> Saul Bejarano
>>
>> aby azid wrote:
>>   
>>> Hi everyone,
>>>
>>> I'm required to make  a stress call on Asterisk server ( > 2000 calls 
>>> per seconds). Are there tools for me to do this sort of test. I was 
>>> thinking of sending loads of Asterisk call files simultaneously 
>>> (starting with 100 call files). Really appreciate if anyone can come up 
>>> with ideas or tools for me to achieve this.
>>>
>>> Cheers,
>>> Aby Azid
>>> Vyke Asia
>>> 
> Where did you get the "Rule" of 30Mhz per call ???
> Wouldn't this be highly dependent on whether it to TDM over a T1 or 
> whether it was in SIP , and which CODEC it was using.
> 
> And why would a properly configured machine "Die", have a HIGH Load 
> Average - YesDIED - Sounds like WinBlows to me
> 
> Yoa - Aby - You need to define your test scenario more fully.  Are you 
> making a call OUT of the box , into the box,  a MIX
> How Long are the calls ?
> Net, net , how many simultaneous call are you going to have ?
> How man Call Originations are there ?
> How may Call Answers ?
> 
> 
> 
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[asterisk-users] PRI TBCT - Practical Experience, Anybody?

[Jay R. Ashworth]

I may have to do some work with TBCT, and probably cross-carrier TBCT,
here shortly, and I haven't ever worked with it.  If anyone on the list
ever has, I'd be interested to know:

1) Only the carrier first involved with the call has to
actually be provisioned for it, correct?

2) Both incoming and outgoing calls can be TBCT'd?

3) If a placed call is transferred to me via TBCT, can I get
the DN of the original target call sent to me as CNID?

4) Does it, in fact, matter if the call placer, and the TBCT
target, are on the same IXC?

I want someone to place calls for me, talk to the people for a
while, and then do an unsupervised transfer to me wherein I can
capture the other party's number off the call itself and feed
the calls into my VICIdial/Asterisk instance.

All my incoming lines are Zap/PRI.

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

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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[asterisk-users] Incoming Bogota DID

[Fred Posner]

Anyone know where I can get an incoming DID for Bogota, Colombia?


Fred Posner
[EMAIL PROTECTED]

Tel: +1 (212) 937-7844 x501
Fax: +1 (954) 252-4187

www.teamforrest.com







smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] dahdi and ztdummy

[Tzafrir Cohen]

On Fri, Aug 15, 2008 at 12:43:39PM -0400, Jerry Geis wrote:
> In the past there was ztdummy - what is the new equivalent in dahdi?

You can start with the README. And it's called dahdi_dummy

> 
> Also it used to be Zap/X what is the new channel name?

That's part of Asteirsk. Not of Zaptel/DAHDI itself . Asterisk has
renamed chan_zap.so to chan_dahdi.so . It now supports DAHDI . It also
supports Zap/ for the moment for backward compatibility .

-- 
   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] zaptel timing

[Tzafrir Cohen]

On Sat, Aug 16, 2008 at 12:52:33AM +0800, Nhadie wrote:
> Hi,
> 
> Just wondering if i can use a card just for timing instead of just using 
> ztdummy? maybe use 2 port fxs or fxo card so it would not be costly.

Yes, it should work well.

> 
> will there be any difference on  the meetme performance if i used a 
> hardware compared to just using ztdummy?
> 
> is there a limit on conference number using ztdummy? will it be more 
> limited if i used hardware?
> 
> i've also read about ztrtc, is this better than ztdummy?

ztrtc is not needed, as ztdummy can use the RTC.

-- 
   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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[asterisk-users] Cisco 7960 audible hold reminder?

[dgray]

Hello,

I have recently setup my first PBX and am wondering if there might be a
way to send audible notification to the cisco 7960 phone when a call is
put on hold. We lost a call due to a customer being on hold and
forgotten about (yikes). Is there a way to get the phone to beep or ring
down the same or other SIP channels after a certain amount of time on
hold?

Thanks!

Dayton Gray

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Re: [asterisk-users] ANSI terminal colors

[Philipp Kempgen]

Jay R. Ashworth schrieb:
> On Thu, Aug 14, 2008 at 03:23:57PM -0500, Tilghman Lesher wrote:
>> On Thursday 14 August 2008 13:59:37 Philipp Kempgen wrote:
>> > Jared Smith schrieb:
>> > > On Thu, 2008-08-14 at 20:35 +0200, Philipp Kempgen wrote:
>> > >> Whenever something spits out lines with a different background
>> > >> color (of a varying runlength!) my eyes start to hurt.
>> > >
>> > > You can turn of the ANSI color support completely by adding
>> > > "nocolor=yes" to the [options] section of asterisk.conf and then
>> > > restarting Asterisk.
>> >
>> > Sure. But I want colored output on my default background color.
>> >
>> > :-)
>> >
>> > ls with dircolors works perfectly. So I was curious if there
>> > is a reason for Asterisk to behave differently (forcing the
>> > background color to black).
>> 
>> Because nobody else ever asked, perhaps?
> 
> No, I would suspect that it's because there's probably a subconscious
> perception that colorcoded text "works better" on black than white, and
> indeed, it does: you often have to fiddle the colors chosen to get it
> to be readable against black: the default red has too little luminance
> to stand out from black, for instance.
> 
> It's the same problem as backlit keyboards with silver coatings, or the
> display on a Palm IIIe: in certain lighting conditions, you simply can't
> read it at all.

True. But lines of varying length with a black background in a
white terminal window don't make it any better. Just causes the
ragged margins to stand out.
If someone sets their default background color to something other
than black it's their choice and they have to live with it.
It's a consious decision and trying to outsmart the user is not
good.

Actually I would like my terminal app to be able to translate
ANSI colors to whatever I want them to be, e.g. map
black bg -> white bg
white fg -> black fg
blue  fg -> red   fg
But that's a different problem which needs a global approach
rather than different approaches by single applications.

Grüße,
Philipp Kempgen
-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998

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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 38

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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Re: [asterisk-users] Asterisk stress call test

[Mik Cheez]

Aby,

Assuming you're building Asterisk from source, you can change the 
following in the scan_thread function:

change-
 sleep(1);

to-
 /*sleep(1);*/
 usleep(10);


This will change the delay from 1 second to 10 microseconds (0.1 
second).

Of course Al's question is more relevant, as placing a thousand SIP 
calls is much different than sending calls through a ZAP channel.  If 
SIP, then what codec?  Also, you're going to find that Asterisk isn't 
the bottleneck in the scenario, but rather the computer (or computers) 
itself.  You can look around www.voip-info.org to see how other users 
performance tests look before trying your own stress test.  This will 
allow you to estimate your own performance before making call 1.

Best regards,

Mik

aby azid wrote:
> Hi,
> 
> Thanks for the reply mates, to Al Baker, It's a stress test for Asterisk 
> outgoing calls, this is to see how Asterisk cope when thousands(1000 - 
> 2000) of calls made simultaneously from the server.
> 
> To Mik, where do I find the pbx_spool.c ?, really appreciate if u can 
> explain more details on the method you used.
> 
> Cheers,
> Aby Azid
> Vyke Asia
> 
> On Fri, Aug 15, 2008 at 1:45 PM, Saul Bejarano <[EMAIL PROTECTED] 
> > wrote:
> 
> Remember the rule of 30Mhz per call when you kill the machine and also
> the bandwidth usage on connected calls.
> 
> Kind regards,
> 
> Saul Bejarano
> 
> aby azid wrote:
>  > Hi everyone,
>  >
>  > I'm required to make  a stress call on Asterisk server ( > 2000 calls
>  > per seconds). Are there tools for me to do this sort of test. I was
>  > thinking of sending loads of Asterisk call files simultaneously
>  > (starting with 100 call files). Really appreciate if anyone can
> come up
>  > with ideas or tools for me to achieve this.
>  >
>  > Cheers,
>  > Aby Azid
>  > Vyke Asia
>  >
>  >
>  >
> 
>  >
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[asterisk-users] zaptel timing

[Nhadie]

Hi,

Just wondering if i can use a card just for timing instead of just using 
ztdummy? maybe use 2 port fxs or fxo card so it would not be costly.

will there be any difference on  the meetme performance if i used a 
hardware compared to just using ztdummy?

is there a limit on conference number using ztdummy? will it be more 
limited if i used hardware?

i've also read about ztrtc, is this better than ztdummy?

thank you

regards,
ron

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Re: [asterisk-users] Cisco 7960

[David Backeberg]

An educated guess is:
reverse the SIP trunk buttons, so the preferred provider is the top
button, and voila, your speed dial going to the first trunk is now
what you want.

On Wed, Aug 13, 2008 at 7:44 PM, Shawn L <[EMAIL PROTECTED]> wrote:
> This one is a little off-topic, it's more about the phone than asterisk
> itself.
>
> I have a cisco 7960 configured with 2 lines to 2 different sip providers
> (cant get
> asterisk to register with the 2nd provider, but that's another story).  Is
> there a
> way yo determine which direction speed-dial buttons will go out?  I'd like
> to have
> speed-dial buttons that will go out on line2 instead of line 1.  Anyone know
> if this
> is possible?
>
> Thanks
>
>
> Shawn
>
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[asterisk-users] dahdi and ztdummy

[Jerry Geis]

In the past there was ztdummy - what is the new equivalent in dahdi?

Also it used to be Zap/X what is the new channel name?

searching voip-info.org for dahdi didnt show me anything about that...

Thanks,

Jerry

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Re: [asterisk-users] noise cancelling headset vs handset

[Jay R. Ashworth]

On Fri, Aug 15, 2008 at 04:59:26AM -0700, Vikas wrote:
> Observation: When the agents talk using the polycom 650 handset the
> voice quality on the other end is much better compared to if they talk
> using the plantronics noise cancelling headset. If you would like a
> recorded phone call when a agent is taking using the headset vs when a
> agent is talking using the polycom 650 handset let me know and I can
> post a URL to the file under this thread.
> 
> Is this happening because they are using a noise cancelling headset.
> It would seem reasonable to expect that the noise cancelling headset
> is cancelling out some of the tones in the voice.

Correct: a noise cancelling headset is going to have substantially
altered audio reproduction characteristics vs. a non-noise cancelling
handset, partially because it's a headset, partially because it's noise
cancelling, and partially because the Polycom desksets have
substantially the best audio I have ever come across.

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

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] AstDB/Berkely DB - Hash function? Balanced-Tree? b-Tree? Linked List?

[Jay R. Ashworth]

On Fri, Aug 15, 2008 at 12:56:49AM -0500, Karl Fife wrote:
> The key-space is ideal.  It's just npa/nxx lookups so it's UNIQUE and
> EVENLY DISTRIBUTED

Based on my knowledge of the NPA/NXX space, I wouldn't expect that
either

a) A given batch of random DNs would have either or both NPA/NXX
components evenly distributed over all the valid NPA/NXXs in the NANPA,
or

b) that the assigned NPA/NXXs in the NANPA are themselves evenly
distributed over all the valid NPA/NXXs.

Could you clarify the background that brings you to that assumption?

Do you have empirical data?

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

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] Basic outbound calling issue

[Felippe Silvestre]

Check if you have some rule to dial under brad1 context

dialplan [EMAIL PROTECTED]

Regards

Felippe Silvestre
 

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Brad
Sent: Friday, August 15, 2008 12:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Basic outbound calling issue

I am trying to lauch a first outbound call.
I am connected to my telco via a peer which is a little 
different from what I consider the norm.

extinsions.conf

[To_Bandwidth]
ignorepat => 9
exten => 9,1,Dial(Sip/g2/)
exten => 9,2,Congestion

sip.conf

[To_Bandwidth]
canreinvite=yes
context=from-pstn
dtmfmode=rfc2833
host=.com
nat=no
outboundproxy=xxx.com
qualify=no
type=peer


error

[Aug 15 11:02:27] NOTICE[25791]: chan_sip.c:14035 
handle_request_invite: Call from 'brad1' to extension 
'919544790554' rejected because extension not found.


  

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Re: [asterisk-users] ANSI terminal colors

[Jay R. Ashworth]

On Thu, Aug 14, 2008 at 03:23:57PM -0500, Tilghman Lesher wrote:
> On Thursday 14 August 2008 13:59:37 Philipp Kempgen wrote:
> > Jared Smith schrieb:
> > > On Thu, 2008-08-14 at 20:35 +0200, Philipp Kempgen wrote:
> > >> Whenever something spits out lines with a different background
> > >> color (of a varying runlength!) my eyes start to hurt.
> > >
> > > You can turn of the ANSI color support completely by adding
> > > "nocolor=yes" to the [options] section of asterisk.conf and then
> > > restarting Asterisk.
> >
> > Sure. But I want colored output on my default background color.
> >
> > :-)
> >
> > ls with dircolors works perfectly. So I was curious if there
> > is a reason for Asterisk to behave differently (forcing the
> > background color to black).
> 
> Because nobody else ever asked, perhaps?

No, I would suspect that it's because there's probably a subconscious
perception that colorcoded text "works better" on black than white, and
indeed, it does: you often have to fiddle the colors chosen to get it
to be readable against black: the default red has too little luminance
to stand out from black, for instance.

It's the same problem as backlit keyboards with silver coatings, or the
display on a Palm IIIe: in certain lighting conditions, you simply can't
read it at all.

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

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] USA Lata AreaCode Database

[Anthony Messina]

On Thursday 14 August 2008 03:09:42 pm roberto wrote:
> I'm looking for some "free" LATA X Area Code database.
>
> Anyone have any idea where can i found?

this site has lots of info: http://www.localcallingguide.com/

-- 
Anthony -  http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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[asterisk-users] Basic outbound calling issue

[Brad]

I am trying to lauch a first outbound call.
I am connected to my telco via a peer which is a little different from what I 
consider the norm.

extinsions.conf

[To_Bandwidth]
ignorepat => 9
exten => 9,1,Dial(Sip/g2/)
exten => 9,2,Congestion

sip.conf

[To_Bandwidth]
canreinvite=yes
context=from-pstn
dtmfmode=rfc2833
host=.com
nat=no
outboundproxy=xxx.com
qualify=no
type=peer


error

[Aug 15 11:02:27] NOTICE[25791]: chan_sip.c:14035 handle_request_invite: Call 
from 'brad1' to extension '919544790554' rejected because extension not found.


  

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Re: [asterisk-users] Problem with Aastra 480ci and qualify=yes

[Drew Gibson]

James Lamanna wrote:
> Hi,
> We have a few Aastra 480ci phones and we've noticed that in order to
> get the phone to receive a call, qualify must be = no.
> Apparently the Aastras do not respond to the qualify message (or
> respond in a way Asterisk doesn't understand) and Asterisk thinks the
> phone is unreachable.
> However, this now prevents MWI from working properly on the phones.
>
> Does anyone know how to get MWI working without qualify? Or how to get
> qualify working again with the Aastras?
>
>   

We have a number of 480i and one 480ct all setup with "qualify=yes" 
(Asterisk 1.2.24)
Our inbound call centre seems to be pretty busy and my own MWI lights up 
far too often. Never had a problem with either.

Which version of firmware?
Which version of Asterisk?
What's in your sip.conf?
What error messages show on the console?
Anything relevant in the logs?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


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[asterisk-users] DID's needed for Reston Virginia - + hosted asterisk

[Dean Collins]

I've just started consulting for a SME client based in Reston Virginia.

 

They don't know it yet but they are going to need a hosted asterisk
service and some DID's. 

 

Email me if you are able to provide 10 DID's in Reston (must be able to
be ported away!!) and hosted Asterisk with end user configurable IVR
etc. Probably only 5-8 users at the moment BUT... they'll be growing
real soon now. Will also need some Polycom handsets and  laptop
softphone extensions etc.

 

Email me your pricing and marketing etc in the next 24-48 hours and I'll
propose my choice on Monday.

 

 

 

 

Regards,

Dean Collins
[EMAIL PROTECTED] 

+1-212-203-4357 (New York) 
+61-2-9016-5642 (Sydney)
http://www.Cognation.net  

 

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[asterisk-users] Problem with Aastra 480ci and qualify=yes

[James Lamanna]

Hi,
We have a few Aastra 480ci phones and we've noticed that in order to
get the phone to receive a call, qualify must be = no.
Apparently the Aastras do not respond to the qualify message (or
respond in a way Asterisk doesn't understand) and Asterisk thinks the
phone is unreachable.
However, this now prevents MWI from working properly on the phones.

Does anyone know how to get MWI working without qualify? Or how to get
qualify working again with the Aastras?

Thanks.

-- James

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Re: [asterisk-users] asterisk realtime and creating "new" contexts

[Todd Fulton]

thanks!  this definitely helps.  now, i'm trying to think of a way to make this happen on multiple asterisk nodes at once.  i really wish realtime would simply read new contexts from the db.  i know that the dialplan is core to the system, but i think this core aspect should be ultimately configurable from a database using realtime.  sigh.todd 

 Original Message 
Subject: Re: [asterisk-users] asterisk realtime and creating "new"
contexts
From: Mike Clark <[EMAIL PROTECTED]>
Date: Fri, August 15, 2008 4:49 am
To: Asterisk Users Mailing List - Non-Commercial Discussion



Todd Fulton wrote:
> Hi,
>
> I'm trying to create a multi-tennant asterisk installation  where 
> each of my customers has its own context.  Well, I've got asterisk 
> realtime working, and I can add/update extensions to existing contexts 
> in extensions.conf without a problem.  However, when I attempt to 
> create database entries with a context that is NOT in extensions.conf, 
> I get an error "invalid extension".
>
> I've found several posts around the net asking this question, but no 
> answers.  Has anyone out there dealt with this problem?
>
> Any help would be great!
>
>
> Todd
>
Todd:

Unfortunately, new contexts don't seem to show up in "real time". I 
solved this in RAGUI by putting #exec statements in the extensions.cong 
file that scan the extensions table and generate the proper contexts. 
However, you still have to do a reload to get the contexts to be 
available in Asterisk.

Here is an example:

in extensions.conf

#exec /opt/pointcall/asterisk/scripts/load_extensions.rb

I used Ruby, but it could be Perl , PHP or whatever

load_extensions.rb

#!/usr/local/bin/ruby
#

require 'mysql'

hostname = "host"
username = "user"
password = "pass"
database = "rtdb"

my = Mysql.new(hostname, username, password, database)

res = my.query("SELECT DISTINCT context FROM extensions ORDER by context")
#
res.each do |row|
context = row[0]
print "\n"
print '[' + context + "]\n"
print "Switch => Realtime/" + context + "\n"
end


Thanks,

Mike

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Re: [asterisk-users] AstDB/Berkely DB - Hash function? Balanced-Tree? b-Tree? Linked List?

[Atis Lezdins]

On Fri, Aug 15, 2008 at 8:56 AM, Karl Fife
<[EMAIL PROTECTED]> wrote:
> Does anyone know enough about the implementation of AstDB to know
> whether the data structure is a Hash function, a Balanced-Tree, a
> b-Tree, or a Linked List?
>
> I'm trying to estimate the lookup 'cost' of a AstDB with around 160,000
> keys?  Obviously I already know that it WILL WORK, but the question is
> whether the data structure is optimal in the Berkeley DB AS IMPLEMENTED
> in Asterisk.  AstDB just like CURL is missing some of its features as
> implemented, so the generic Berkeley Doc doesn't help much.
>
> The key-space is ideal.  It's just npa/nxx lookups so it's UNIQUE and
> EVENLY DISTRIBUTED--a perfect for a hash function (or even a balanced
> tree).  What I do NOT want is a 150k member linked list, or even a
> standard b-tree that ends up being 160k entries tall because the values
> were inserted in order etc.
>
> In terms of Databases, I know that 160K keys is very small potatoes, but
> I want to make sure I understand what's going on under the hood so that
> my lookup costs are as low as possible.
>
> Lookups on my Oracle database with tens of millions of records are
> instantaneous and inexpensive when implemented properly, and about ten
> thousand times slower (actually) when done improperly.  If a database
> has only 160K keys, it can be tough to tell whether the data structures
> are being used efficiently, because even WILDLY INEFFICIENT lookups seem
> fast.
>
> Can anyone speak to this?
> What is the default data structure?
> How many records have you stuck into the AstDB?
>

Hi,

I can share my experience only. I'm storing call variables in astdb,
daily ~2000 calls, ~20 variables per call. Stored in way like this:

call_variables/${uniqueid}/var=value

Previously i did cleanup of this only nightly, but sometimes i noticed
that lookup in evenings get extremely slow - ~5-10 seconds per
variable. So, i would guess that it's not very optimal, as every
lookup from 2000 keys takes several seconds (sub-keys shouldn't affect
this).

Anyway, my solution was to keep it small by deleting everything upon
hangup of call. A little clutter, but it works more or less.

Regards,
Atis


-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 37

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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Re: [asterisk-users] Asterisk vs c-client issues

[Russell Bryant]

Lee Lundrigan wrote:
> Hi everyone,
> 
> Are there any incompatibility issues between asterisk and the c-client 
> using SSL?
> When I enable SSL I get the error:
> *pbx.c:1832 pbx_extension_helper: No application 'VoiceMailMain'
> *whenever I am trying to access voicemail.
> 
> But when SSL is disabled everything works great, just like its supposed 
> to with imap.
> 
> Any ideas?

It looks like app_voicemail is failing to load when you build c-client 
with SSL support.  Try running "module load app_voicemail.so" from the 
Asterisk CLI to see what the error message is.

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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Re: [asterisk-users] AstDB/Berkely DB - Hash function? Balanced-Tree? b-Tree? Linked List?

[Russell Bryant]

Karl Fife wrote:
> Does anyone know enough about the implementation of AstDB to know
> whether the data structure is a Hash function, a Balanced-Tree, a
> b-Tree, or a Linked List? 

I've never looked at the internals of db1.  However, by simply looking 
at what code is included, it looks like it is based on a b-tree.

You may have to add some debugging within db1 to see how nodes actually 
get laid out when you add your 160k entries.  The code is in main/db1-ast/.

If you're doing something this large, I would encourage you to consider 
just using a different database, and using func_odbc to access it as 
opposed to the astdb functions.

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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[asterisk-users] noise cancelling headset vs handset

[Vikas]

Observation: When the agents talk using the polycom 650 handset the
voice quality on the other end is much better compared to if they talk
using the plantronics noise cancelling headset. If you would like a
recorded phone call when a agent is taking using the headset vs when a
agent is talking using the polycom 650 handset let me know and I can
post a URL to the file under this thread.

Is this happening because they are using a noise cancelling headset.
It would seem reasonable to expect that the noise cancelling headset
is cancelling out some of the tones in the voice.

I have two questions:
A. How to get the same valice quality on the other end from a headset
as compared to a polycom 650 handset ?
B. When I went shopping from a headset I found three different kinds
1. flexboom. 2. Microboom and 3. soundtube .What is the difference
between these ?

Thanks for your time,

sysadmin
http://www.grmtech.com

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Re: [asterisk-users] asterisk realtime and creating "new" contexts

[Mike Clark]

Todd Fulton wrote:
> Hi,
>
> I'm trying to create a multi-tennant asterisk installation  where 
> each of my customers has its own context.  Well, I've got asterisk 
> realtime working, and I can add/update extensions to existing contexts 
> in extensions.conf without a problem.  However, when I attempt to 
> create database entries with a context that is NOT in extensions.conf, 
> I get an error "invalid extension".
>
> I've found several posts around the net asking this question, but no 
> answers.  Has anyone out there dealt with this problem?
>
> Any help would be great!
>
>
> Todd
>
Todd:

Unfortunately, new contexts don't seem to show up in "real time". I 
solved this in RAGUI by putting #exec statements in the extensions.cong 
file that scan the extensions table and generate the proper contexts. 
However, you still have to do a reload to get the contexts to be 
available in Asterisk.

Here is an example:

in extensions.conf

#exec /opt/pointcall/asterisk/scripts/load_extensions.rb

I used Ruby, but it could be Perl , PHP or whatever

load_extensions.rb

#!/usr/local/bin/ruby
#

require 'mysql'

hostname = "host"
username = "user"
password = "pass"
database = "rtdb"

my = Mysql.new(hostname, username, password, database)

res = my.query("SELECT DISTINCT context FROM extensions ORDER by context")
#
res.each do |row|
context = row[0]
print "\n"
print '[' + context + "]\n"
print "Switch => Realtime/" + context + "\n"
end


Thanks,

Mike

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Re: [asterisk-users] Asterisk stress call test

[Atis Lezdins]

On Fri, Aug 15, 2008 at 8:31 AM, aby azid <[EMAIL PROTECTED]> wrote:
> Hi everyone,
>
> I'm required to make  a stress call on Asterisk server ( > 2000 calls per
> seconds). Are there tools for me to do this sort of test. I was thinking of
> sending loads of Asterisk call files simultaneously (starting with 100 call
> files). Really appreciate if anyone can come up with ideas or tools for me
> to achieve this.
>

Hi,

I've written test framework, you'll need another machine with Asterisk
(+php) on it to generate calls. It allows to write scripts in PHP to
emulate random customer actions, etc.. You can download it here
http://ftp.iq-labs.net/pbx-test/
If you find it useful, or get into some problems, don't hesitate to write me.

If you need just bunch of identical calls, you may also try out SIPp.

Regards,
Atis




-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] Multi-homed Asterisk

[Grygoriy Dobrovolskyy]

2008/8/14 Jon Weisman <[EMAIL PROTECTED]>

> Hey guys its me again! So I need to setup our Asterisk server with multiple
> IP providers. The server has two NICs, we have two providers, now we want
> redundancy! Any guides on how to set this up? We're running Fedora Core 5
> w/
> Asterisk 1.4.
>
> What I would like is that incase one link goes down, or even if the link is
> showing up, but for some reason its not routing traffic the other link
> should take over. This is for both inbound and outbound SIP. Now the hard
> part... I need this to be reliable.
>
> Your help is greatly appreciated,
>
> Jon
>
>
>
> Use pfsense or zeroshell they have a failover with multiple wans.
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Re: [asterisk-users] asterisk realtime and creating "new" contexts

[Todd Fulton]

hi saul,could you give me more info on the "VMX-CONTEXT" concept?  i tried to google it, but could find nothing.i am trying to do exactly what you state in terms of creating a virtual slice of the box for each user.  thanks!todd

 Original Message 
Subject: Re: [asterisk-users] asterisk realtime and creating "new"
contexts
From: Saul Bejarano <[EMAIL PROTECTED]>
Date: Thu, August 14, 2008 8:01 pm
To: asterisk-users@lists.digium.com


The VMX-CONTEXT can be managed from database

mysql> select * from globals;
But you will have to specify the same on each extension under sip
once the extension under table sip is the one that calls the CONTEXT to 
route the call.

| PARKNOTIFY   | SIP/200 |
| RECORDEXTEN  | ""  |
| RINGTIMER| 15  |
| DIRECTORY| last|
| AFTER_INCOMING   | |
| IN_OVERRIDE  | forcereghours   |
| REGTIME  | 7:55-17:05  |
| REGDAYS  | mon-fri |
| DIRECTORY_OPTS   | |
| DIALOUTIDS   | 1/2/3/4/5/6/|
| OUT_1| ZAP/g0  |
| VM_PREFIX| *   |
| VM_OPTS  | |
| VM_GAIN  | |
| VM_DDTYPE| u   |
| TIMEFORMAT   | kM  |
| TONEZONE | us  |
| ALLOW_SIP_ANON   | yes |
| VMX_CONTEXT  | from-internal

It will make it sort of complicated thought because it was build to be 
the generic element routing the calls out of the SIP Registrar, by 
having individual Context what you are trying to do is partition one 
Asterisk box to function as multiple offering invidual termination to 
each one of your users or complete management of a virtual slice of the 
box :)

Todd Fulton wrote:
> Hi,
> 
> I'm trying to create a multi-tennant asterisk installation  where 
> each of my customers has its own context.  Well, I've got asterisk 
> realtime working, and I can add/update extensions to existing contexts 
> in extensions.conf without a problem.  However, when I attempt to create 
> database entries with a context that is NOT in extensions.conf, I get an 
> error "invalid extension".
> 
> I've found several posts around the net asking this question, but no 
> answers.  Has anyone out there dealt with this problem?
> 
> Any help would be great!
> 
> 
> Todd
> 
> 
> 
> 
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Re: [asterisk-users] New GUI for Realtime Asterisk - RAGUI

[Mindaugas Kezys]

Congratulations with the release!

I'm curious also about the statement on your page: 

"Realtime Asterisk uses the MySQL relational database to access dialplan,
extension and configuration data. 
This allows for dynamic additions and changes to users, extensions and
dialplans without having to restart or reload the system."

What version of Asterisk are you using?

>From my experience starting from 1.4.19 Asterisk Realtime is completely
broken:

1. http://bugs.digium.com/view.php?id=12362
2. http://bugs.digium.com/view.php?id=12925
3. http://bugs.digium.com/view.php?id=12921

Also how do you go about changing details for device in DB and not using
"sip realtime prune PEER" + 'sip reload'?

Without that your changes to devices are not active.

Good luck!

Regards,
Mindaugas Kezys
http://www.kolmisoft.com

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Mike Clark
> Sent: Thursday, August 14, 2008 12:07 AM
> To: asterisk-users@lists.digium.com; Commercial and Business-Oriented
> Asterisk Discussion
> Subject: [asterisk-users] New GUI for Realtime Asterisk - RAGUI
> 
> Our company, WebPoint IT Solutions has just released an open source
> (GPL
> V2 license), Ruby on Rails based gui manager for Realtime Asterisk
> called RAGUI.
> 
> RAGUI is definitely a work in progress and has rough edges, but we
> expect to polish it up in the upcoming weeks and months. All comments,
> contributions, and criticisms are welcomed!
> 
> Here are the links:
> Sourceforge: http://sourceforge.net/projects/ragui/
> Website: http://www.ragui.net
> 
> Enjoy!
> 
> Mike Clark
> WebPoint IT Solutions
> 
> 
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Re: [asterisk-users] Asterisk stress call test

[Al Baker]

OK - but again - more specifics are needed.
If you are going TDM over T1 that is a Totally Different Animal
than cranking up all these using IAX or .
Also, you still have to identify how many simultaneous calls you will have.
Again 1000 calls done "essentially" all at once is a different animal 
than if if stagger them , even a little.
So your traffic Profile will make a huge difference

aby azid wrote:
> Hi,
>
> Thanks for the reply mates, to Al Baker, It's a stress test for 
> Asterisk outgoing calls, this is to see how Asterisk cope when 
> thousands(1000 - 2000) of calls made simultaneously from the server.
>
> To Mik, where do I find the pbx_spool.c ?, really appreciate if u can 
> explain more details on the method you used.
>
> Cheers,
> Aby Azid
> Vyke Asia
>
> On Fri, Aug 15, 2008 at 1:45 PM, Saul Bejarano <[EMAIL PROTECTED] 
> > wrote:
>
> Remember the rule of 30Mhz per call when you kill the machine and also
> the bandwidth usage on connected calls.
>
> Kind regards,
>
> Saul Bejarano
>
> aby azid wrote:
> > Hi everyone,
> >
> > I'm required to make  a stress call on Asterisk server ( > 2000
> calls
> > per seconds). Are there tools for me to do this sort of test. I was
> > thinking of sending loads of Asterisk call files simultaneously
> > (starting with 100 call files). Really appreciate if anyone can
> come up
> > with ideas or tools for me to achieve this.
> >
> > Cheers,
> > Aby Azid
> > Vyke Asia
> >
> >
> >
> 
> >
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Re: [asterisk-users] Asterisk stress call test

[aby azid]

Hi,

Thanks for the reply mates, to Al Baker, It's a stress test for Asterisk
outgoing calls, this is to see how Asterisk cope when thousands(1000 - 2000)
of calls made simultaneously from the server.

To Mik, where do I find the pbx_spool.c ?, really appreciate if u can
explain more details on the method you used.

Cheers,
Aby Azid
Vyke Asia

On Fri, Aug 15, 2008 at 1:45 PM, Saul Bejarano <[EMAIL PROTECTED]> wrote:

> Remember the rule of 30Mhz per call when you kill the machine and also
> the bandwidth usage on connected calls.
>
> Kind regards,
>
> Saul Bejarano
>
> aby azid wrote:
> > Hi everyone,
> >
> > I'm required to make  a stress call on Asterisk server ( > 2000 calls
> > per seconds). Are there tools for me to do this sort of test. I was
> > thinking of sending loads of Asterisk call files simultaneously
> > (starting with 100 call files). Really appreciate if anyone can come up
> > with ideas or tools for me to achieve this.
> >
> > Cheers,
> > Aby Azid
> > Vyke Asia
> >
> >
> > 
> >
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