date:20041228

[Asterisk-Users] asterisk CVS-HEAD is nutzo!

2004-12-28 Thread Gabriel Afana

I probably dont know what I am doingthats all

But from a clean install I installed zaptel, libpri, asterisk CVS-HEAD and
asterisk-addons.  All went ok...but asterisk is acting crazy.  It wont let
me register any SIP channels, otherwise it hangs when I start it at the SIP
part.  When I start asterisk without anything registered, it starts but will
only process a few commands then it stops responding:

[EMAIL PROTECTED] asterisk]# /usr/sbin/asterisk -r
Asterisk CVS-HEAD-12/29/04-12:27:52, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
Connected to Asterisk CVS-HEAD-12/29/04-12:27:52 currently running on g0
(pid = 5343)
Usage: set verbose 
   Sets level of verbose messages to be displayed.  0 means
   no messages should be displayed. Equivalent to -v[v[v...]]
   on startup
Usage: set debug 
   Sets level of core debug messages to be displayed.  0 means
   no messages should be displayed. Equivalent to -d[d[d...]]
   on startup.
g0*CLI> show version
Asterisk CVS-HEAD-12/29/04-12:27:52 built by [EMAIL PROTECTED] on a x86_64
running Linux
g0*CLI> stop now
g0*CLI> stop now
g0*CLI> alsdkfj
g0*CLI> alsdj;fa
g0*CLI> bla bla
g0*CLI> hello

I started with my own config files, but have gone all the way back to the
default sample filesit still does the same thing.

One quick note, on the SIP channel registration, when I register as a peer
or friend, it locks up when it loads.  If I register as a user, it asterisk
will load, however it wont work.  I have Realtime enabled through MySQL, and
its up and running and can connect and everything, but I dont have any
settings getting defined through it (via the extconfig.conf file)

I had everything working perfect for like a week using the stable
version...so not sure if this CVS-HEAD version has a lot of changes that are
making my stable config files not work or if its just not working right on
my server.

I am running RedHat Enterprise Server ES Update 4 on AMD64 platform.

I am going to downgrade back to the stable version for nowbut I would
love to get this CVS-HEAD version working because I really need the Realtime
Extensions feature!!!

Gabe

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Re: [Asterisk-Users] how to debug frame slips?

2004-12-28 Thread Adam Goryachev

On Tue, 2004-12-28 at 06:10, Michael Welter wrote:
> Try 'lspci -v' and look at the latency timer for your Digium card(s). 
> You can set it higher with 'setpci -v -s xx:yy.0 LATENCY=TIMER=ff' (xx 
> is the bus number and yy is the slot).
> 

Shouldn't you decrease the latency? ie, to something lower like 16 or 8
or 0 ?? or is a higher value better??

PS, some of my devices have a latency of 0, is that normal?
00:00.0 Host bridge: Advanced Micro Devices [AMD] AMD-760 MP [IGD4-2P]
System Controller (rev 20)
Flags: bus master, 66Mhz, medium devsel, latency 32
Memory at fa00 (32-bit, prefetchable) [size=32M]
Memory at f830 (32-bit, prefetchable) [size=4K]
I/O ports at 1010 [disabled] [size=4]
Capabilities: [a0] AGP version 2.0

00:01.0 PCI bridge: Advanced Micro Devices [AMD] AMD-760 MP [IGD4-2P]
AGP Bridge (prog-if 00 [Normal decode])
Flags: bus master, 66Mhz, medium devsel, latency 64
Bus: primary=00, secondary=01, subordinate=01, sec-latency=64

00:07.0 ISA bridge: Advanced Micro Devices [AMD] AMD-768 [Opus] ISA (rev
05)
Flags: bus master, 66Mhz, medium devsel, latency 0

00:07.1 IDE interface: Advanced Micro Devices [AMD] AMD-768 [Opus] IDE
(rev 04) (prog-if 8a [Master SecP PriP])Subsystem: Advanced
Micro Devices [AMD] AMD-768 [Opus] IDE
Flags: bus master, medium devsel, latency 0
I/O ports at f000 [size=16]



Thanks,
Adam

-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 8304 [EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

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Re: [Asterisk-Users] PRI & CPU Usage

2004-12-28 Thread Adam Goryachev

On Wed, 2004-12-29 at 11:22, Derek Conniffe wrote:
> Hi everyone,  
> 
> I'm a bit unclear about how PRI voice channels use CPU usage in an asterisk
> box - it is like a codec conversion or does a PRI channel have a generic
> codec itself?

AFAIK, a PRI will use either ALAW or ULAW, depending on your telco's
default. ie, in Australia we use ALAW, in the US and possible other
places, they might use ULAW

>   I am wondering what the CPU requirements are to, for example,
> handle all 120 channels on a TE405p and then trunk all the conversations
> elsewhere over an IAX2 link.

Well, that is hard to say, assuming you are not going to transcode to
some other codec, then I *assume* it would be minimal, but... given the
trunking, there may be some additional conversion needed.

Regards,
Adam

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Re: [Asterisk-Users] Sending call to analog then to Vmailaftertimeout?

2004-12-28 Thread Me

Nevermind, it looks like "Asterisk cmd Read" is my lucky command :)

Thanks!

Start Your Own Internet Service!
http://www.YourOwnISP.com

- Original Message - 
From: "Me" <[EMAIL PROTECTED]>
To: "C F" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial
Discussion" 
Sent: Wednesday, December 29, 2004 12:19 AM
Subject: Re: [Asterisk-Users] Sending call to analog then to
Vmailaftertimeout?


> I was trying this logic before, I got as far as going into the Macro,
> playing a message and then.. Well... I got lost, I am not sure how to go
> about require them to press a button. Normally I can make someone press an
> extension but from what I read about Macros in * you have to stay within
the
> "s" extension.
>
> Any idea where I can find an example of this sort of thing?
>
> Thanks!
>
> Start Your Own Internet Service!
> http://www.YourOwnISP.com
> - Original Message - 
> From: "C F" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Tuesday, December 28, 2004 11:34 PM
> Subject: Re: [Asterisk-Users] Sending call to analog then to
> Vmailaftertimeout?
>
>
> > -- Forwarded message --
> > From: C F <[EMAIL PROTECTED]>
> > Date: Wed, 29 Dec 2004 00:34:28 -0500
> > Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
> aftertimeout?
> > To: Me <[EMAIL PROTECTED]>
> >
> >
> > try the M option which will do a macro and will not connect the caller
> > unless s/he presses some button. and if no button is pressed then it
> > goes to VM. now remember to replay the message (to press the button) a
> > few times b4 going to VM otherwise they will never hear it, since *
> > considers it answered .
> > http://www.voip-info.org/wiki-Asterisk+cmd+dial
> >
> >
> > On Tue, 28 Dec 2004 23:29:54 -0600, Me <[EMAIL PROTECTED]> wrote:
> > > I was aware of the "c" option but it's a pain for people to have to
> press
> > > the # sign plus they have to know they are suppose to do that. In
> addition,
> > > I tried to use the "A" option to play a sound to them when they answer
> > > reminding them to press pound at the end of the message but the sound
> > > doesn't play until they press pound :)
> > >
> > > So.. It appears I am still stuck with * considering the call answered
> when
> > > the Zap channels grabs it and connects the other leg of the call.
> Hopefully
> > > there is some other way to make this happen.
> > >
> > > Thanks for the feedback though.
> > >
> > > Start Your Own Internet Service!
> > > http://www.YourOwnISP.com
> > >
> > > - Original Message -
> > > From: "C F" <[EMAIL PROTECTED]>
> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > 
> > > Sent: Tuesday, December 28, 2004 6:26 PM
> > > Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
> > > aftertimeout?
> > >
> > > > Follow these:
> > > > http://www.voip-info.org/wiki-Asterisk+zap+channels
> > > > looks like this would work:
> > > >  exten => 1200,1,playback(pls-wait-connect-call)
> > > >  exten => 1200,2,Dial(Zap/1c/551212,20,rTt) ;note the c after
the
> > > > channel number
> > > >  exten => 1200,3,VoiceMail([EMAIL PROTECTED])
> > > >  exten => 1200,4,Goto,t|1
> > > >
> > > >
> > > > On Tue, 28 Dec 2004 14:20:02 -0600, Me <[EMAIL PROTECTED]>
> wrote:
> > > > > Sorry about the HTML emails, on my laptop and forgot to change the
> > > sending
> > > > > format from the default.
> > > > >
> > > > >
> > > > > - Original Message -
> > > > > From: Me
> > > > > To: asterisk-users@lists.digium.com
> > > > > Sent: Tuesday, December 28, 2004 2:01 PM
> > > > > Subject: [Asterisk-Users] Sending call to analog then to Vmail
after
> > > > > timeout?
> > > > >
> > > > > I have one analog line hooked in my Asterisk box using an x100p (I
> think
> > > > > that's the model number).
> > > > >
> > > > > When I do this in my extensions.conf:
> > > > >
> > > > > exten => 1200,1,playback(pls-wait-connect-call)
> > > > > exten => 1200,2,Dial(Zap/1/551212,20,rTt)
> > > > > exten => 1200,3,VoiceMail([EMAIL PROTECTED])
> > > > > exten => 1200,4,Goto,t|1
> > > > >
> > > > > The phone rings beyond the 20 second timeout and never really goes
> to
> > > the *
> > > > > voicemail. I can't seem to get it to timeout regardless of how
many
> > > seconds
> > > > > I set it to.
> > > > >
> > > > > I assume this has something to do with the fact that * considers
the
> > > call
> > > > > answered as soon as the zap channel picks it up, right?
> > > > >
> > > > > Anyhow, is there a way to make the above config work and go to the
*
> > > > > voicemail after 20 seconds if the called party does not answer
after
> 20
> > > > > seconds? Also, what happens if the called party's line is busy,
have
> not
> > > run
> > > > > into this yet so I am curious.
> > > > >
> > > > > Thanks!
> > > > >
> > > > > --
> > > > > Start Your Own Internet Service!
> > > > > http://www.YourOwnISP.com
> > > > >
> > > > >
> > > > > __

Re: [Asterisk-Users] Sending call to analog then to Vmailaftertimeout?

2004-12-28 Thread Me

I was trying this logic before, I got as far as going into the Macro,
playing a message and then.. Well... I got lost, I am not sure how to go
about require them to press a button. Normally I can make someone press an
extension but from what I read about Macros in * you have to stay within the
"s" extension.

Any idea where I can find an example of this sort of thing?

Thanks!

Start Your Own Internet Service!
http://www.YourOwnISP.com
- Original Message - 
From: "C F" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, December 28, 2004 11:34 PM
Subject: Re: [Asterisk-Users] Sending call to analog then to
Vmailaftertimeout?


> -- Forwarded message --
> From: C F <[EMAIL PROTECTED]>
> Date: Wed, 29 Dec 2004 00:34:28 -0500
> Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
aftertimeout?
> To: Me <[EMAIL PROTECTED]>
>
>
> try the M option which will do a macro and will not connect the caller
> unless s/he presses some button. and if no button is pressed then it
> goes to VM. now remember to replay the message (to press the button) a
> few times b4 going to VM otherwise they will never hear it, since *
> considers it answered .
> http://www.voip-info.org/wiki-Asterisk+cmd+dial
>
>
> On Tue, 28 Dec 2004 23:29:54 -0600, Me <[EMAIL PROTECTED]> wrote:
> > I was aware of the "c" option but it's a pain for people to have to
press
> > the # sign plus they have to know they are suppose to do that. In
addition,
> > I tried to use the "A" option to play a sound to them when they answer
> > reminding them to press pound at the end of the message but the sound
> > doesn't play until they press pound :)
> >
> > So.. It appears I am still stuck with * considering the call answered
when
> > the Zap channels grabs it and connects the other leg of the call.
Hopefully
> > there is some other way to make this happen.
> >
> > Thanks for the feedback though.
> >
> > Start Your Own Internet Service!
> > http://www.YourOwnISP.com
> >
> > - Original Message -
> > From: "C F" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > Sent: Tuesday, December 28, 2004 6:26 PM
> > Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
> > aftertimeout?
> >
> > > Follow these:
> > > http://www.voip-info.org/wiki-Asterisk+zap+channels
> > > looks like this would work:
> > >  exten => 1200,1,playback(pls-wait-connect-call)
> > >  exten => 1200,2,Dial(Zap/1c/551212,20,rTt) ;note the c after the
> > > channel number
> > >  exten => 1200,3,VoiceMail([EMAIL PROTECTED])
> > >  exten => 1200,4,Goto,t|1
> > >
> > >
> > > On Tue, 28 Dec 2004 14:20:02 -0600, Me <[EMAIL PROTECTED]>
wrote:
> > > > Sorry about the HTML emails, on my laptop and forgot to change the
> > sending
> > > > format from the default.
> > > >
> > > >
> > > > - Original Message -
> > > > From: Me
> > > > To: asterisk-users@lists.digium.com
> > > > Sent: Tuesday, December 28, 2004 2:01 PM
> > > > Subject: [Asterisk-Users] Sending call to analog then to Vmail after
> > > > timeout?
> > > >
> > > > I have one analog line hooked in my Asterisk box using an x100p (I
think
> > > > that's the model number).
> > > >
> > > > When I do this in my extensions.conf:
> > > >
> > > > exten => 1200,1,playback(pls-wait-connect-call)
> > > > exten => 1200,2,Dial(Zap/1/551212,20,rTt)
> > > > exten => 1200,3,VoiceMail([EMAIL PROTECTED])
> > > > exten => 1200,4,Goto,t|1
> > > >
> > > > The phone rings beyond the 20 second timeout and never really goes
to
> > the *
> > > > voicemail. I can't seem to get it to timeout regardless of how many
> > seconds
> > > > I set it to.
> > > >
> > > > I assume this has something to do with the fact that * considers the
> > call
> > > > answered as soon as the zap channel picks it up, right?
> > > >
> > > > Anyhow, is there a way to make the above config work and go to the *
> > > > voicemail after 20 seconds if the called party does not answer after
20
> > > > seconds? Also, what happens if the called party's line is busy, have
not
> > run
> > > > into this yet so I am curious.
> > > >
> > > > Thanks!
> > > >
> > > > --
> > > > Start Your Own Internet Service!
> > > > http://www.YourOwnISP.com
> > > >
> > > >
> > > > ___
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users@lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > > ___
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users@lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > ___
> > > Asterisk-Use

[Asterisk-Users] external Radius Server integration with asterisk

2004-12-28 Thread Inam




Hi every body
 
I am bit new with asterisk .
i have configured it to get it working for SIP 
calls through xlite dialer from xten.
for this i used sip.cof and the user were defined 
there ,i want to put the user in db and want the raduis to authenticate the 
user through it .
can any body let me kow how can i confugure the 
external radius server with asterisk so that asterisk send radius request for 
athentication to some radius server.
thanks in advance .
 
INAMULLAH
 
 
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Re: [Asterisk-Users] Invalid Extension

2004-12-28 Thread Norman Zhang

I can dial-in and here the prompt, but whenever I select 101, I get 
invalid extension. May I ask, is this the right way of answering 
incoming calls?

[inbound-sip]
exten => 533990,1,Answer
exten => s,2,ResponseTimeout(5)
exten => s,3,Background(mymenu)
Bearing in mind that the extensions are => extension, priority, 
something to do, you seem to be missing s,1...
I need to replace all s with 533990, so * would answer to incoming 
calls. If I use s, I get error 404. Am I missing something?

Regards,
Norman Zhang
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Re: [Asterisk-Users] Callmanager 4.1 and Asterisk

2004-12-28 Thread Gonzalo Gasca Meza

You need to create a SIP trunk in CCM and in Asterisk a peer in sip.conf with the IP address of the CCM (trunk)
In the trunk configuration change the transport to UDP.
Enter the IP of Asterisk.
And create a route pattern with gateway the SIP trunk
 
In Asterisk in extensions.conf create the route to CCM phones.
I have this setup in my lab with CCM 4.02sr1 and works so fine.
If you need the sip.conf / extensions.conf and an screenshot of the route pattern and SIP trunk config just let me know!
Happy holidays!
 
Keith O'Brien <[EMAIL PROTECTED]> wrote:



I have a similar setup.   To make it easy and get the best of both worlds, have the Linux softphones (SIP or IAX) register to Asterisk.   Keep the physical phones registered to CM.   From there setup a dialplan on both Call Manager and Asterisk to relay calls between the two systems.   For example, assign all physical phones extension 2XXX and softphones 3XXX.   Have asterisk route 2XXX calls to CM via SIP and vice versa on Call Manager.
Also, just so that you are aware you can register a SIP Linux softclient to Cisco Call Manager if you are running Version 4.1
---
Hello everybody,
im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager.
We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux machines, i read about callmanager uses skinny a propetary protocol and there are no softphones from linux to talk with it, so we need to install vmware to use ipcommunicator or the other solutions as i read is get the asterisk server using sip phones in the linux and windows machines and configure the call manager to talk with the asterisk server thru sip protocol, is this the real way to do that?? is there a easy way to do this?? i found this link
http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration
but i need to know what things to do to transfer all the extensions from de callmanager to the asterisk sw, or if only made the changes in the sip.conf as said in the link above the callmanager gets all the control??
or if i need to declare all the extensions in the asterisk?? can anybody help me??
TIA
Edgar

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[Asterisk-Users] Mediatrix 1204 DialPlan and Delay

2004-12-28 Thread Gonzalo Gasca Meza

Hi everybody,
I have setup a Mediatrix 1204, the calls worked fine, both incoming and outgoing.
The problem here is the delay.
When I do a call to the PSTN or receive a call from the PSTN exists a delay of 4 seconds after answer or sending the call.
For OUTGOING
My Dialplan for the Mediatrix box is the following, here at Mexico we use 8 digits for local calls.
([1-9]xxx|01xx||060|0xx)
I have verified that inmediatly after I dial from my IP phone, the in-use light turns on in Mediatrix but the call is not pass until the 4 seconds timer expires.
I have tried disabling the Dial plan but it didnt help
Form Mediatrix documentation
The Timer is set to 4 seconds. It can be used to indicate that if users have not dialed a digit for 4 seconds, it is likely that they have finished dialing and the gateway can make the call. A Dial Map for this could be:
[2-9]xxT
FOR INCOMING
The same 4 seconds delay after the call is sent to Asterisk.
The problem here, is that despite we answer or not the call, once the call is sent to Mediatrix, the calling party hear 2 ring-back tones generated by Mediatrix, then the ringback for Asterisk
Once the call is passed to Asterisk and starts ringing, if we call from a cell phone,home or office the call is marked as answered and the call timer starts no matter if is answered or not.
Any ideas?
I have tried sending the # at the end with no success.
Thanks!
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