RE: Re[5]: [asterisk-users] Latest Debian and latest zaptel

2006-11-10 Thread brandon kruz

svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
cd zaptel
make clean ; make distclean ; sh configure ; make ; make install

modprobe ztdummy

see if those give you any errors



From: Christian [EMAIL PROTECTED]
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Hi,
Well, I managed to find out what svn was and I also downloaded the latest 
1.4 version of zaptel and Asterisk.

svn checkout http://svn.digium.com/svn/zaptel/branches/1.4
But the problem is still there. Cannot install Zaptel. Asterisk and libpri 
installs just fine. Want to use ztdummy.

Soon giving up on this one!
Many thanks,
Christian


On 2006-11-10 at 23:51 Michiel van Baak wrote:

On Nov 10, 2006, at 11:12 PM, Christian wrote:

 Hi,
 But what is the problem, why doesnt it install?
 I am a little new to this so still learning.
 Many thanks,
 Christian

Use the latest 1.4 svn version instead of beta2
That will probably fix your problem.
I installed latest 1.4 svn checkout on debian etch 2 times today
while playing with xen so I can confirm it's working great now.

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Why is it drug addicts and computer afficionados are both called
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RE: [asterisk-users] Queues and multiple lines

2006-11-07 Thread brandon kruz

Using SIP:

Just create another user account
say the softphones user's name is bob:

create [bob] (bob's main line on his softphone)
create [bob1] (same configuration options, then you can do
all your other configurations for this user )

hope this helps

anyone is open to correcting me :]

my 2 cents
`KruZ~



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Subject: [asterisk-users] Queues and multiple lines
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Say I have agents using a softphone like eyebeam that has 6 lines. They log 
in to the queue. Say there are 3 agents in my queue. 3 calls come in and 
all three agents are on a call. Now a fourth call comes in. Is it possible 
to have it setup so that the 4 call rings on line 2 of one of my agents, if 
they don't get it within the time limit it rings on line 2 of another agent 
and so on. An agent can then put their current call on hold and go to the 
new call, say something like thanks for calling please hold, then go back 
to their first call, finish it up and then go back to the second call. I 
hope that made sense. I'm sure there is a way to get it done, but how 
flexible is the current queue system in Asterisk with stuff like this?


--
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000

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RE: [asterisk-users] MOH distorted on Pound Key Linux on asterisk 1.2.8

2006-09-19 Thread brandon kruz

is this open source poundkey?

and can i see your moh conf??

im guessing its open source pk since you mentioned the asterisk 1.2.8 part.

and also is it the default MOH or your own cooked up version??
also i recommend, if not necessary the EXACT mpg version described in the 
conf
(321 0.9?) something similar, please try this, but first of all before we go 
that extreme.


lets see your conf's and if you made your own music, or default MOH

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1.2.8

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Running Asterisk 1.2.8 on Pound Key linux which I downloaded from Digium
site.

Uname output:

Linux localhost 2.6.13.4-1.x86.i686.cmov #1 Wed Nov 23 11:31:48 EST 2005
i686 athlon i386 GNU/Linux





It didn't come with mpg123 so I downloaded it from the internet.  MOH
works, but it is terribly  loud and mistorted. Tried running under
quitemp3 profile but it didn't help.



I feel like there is something I may be missing here. Any ideas???

Thanks in advance.



Jeronimo.





==

Jeronimo Romero

EUS Networks

Email: [EMAIL PROTECTED]

Cell: 917-332-7238

Office: 212-624-5943

Web: www.euscorp.com

==







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Re: [asterisk-users] Pri Event 6 and 8

2006-09-19 Thread brandon kruz


jra i think you are correct, because the timing source
being the hardware or channel psuedo can get bogged down
i believe, or at least from my experience i have seen this on more
than one occasion dealing with the timing source being either 
locked(terminology?)

or under use, or just bogged down

also if its recieving timing from another card not on the local system
( for example, normal T1 setups.)


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On Tue, Sep 19, 2006 at 07:02:28PM -0400, Doug Lytle wrote:
 Dave Wise wrote:
 I was looking through my log files and keep seeing:
 
 PRI got event: 8 on Primary D-channel of span 3
 on different spans.  sometimes it is event 6.

 A quick Google on this showed a post by Mark on bugs.digium.com:

  Generally 6 and 8 errors are not a problem, but can be indicitive of
 either missed interrupts or a timing mismatch. In any case this is a
 technical support issue and not a bug in software and should be pursued
 through digium tech support.

I wonder if they mean a clock slip on the line, by that, or something
else?

Cheers,
-- jra
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Re: [asterisk-users] Digium GUI?

2006-09-19 Thread brandon kruz

just wait until 1.4 is released
and we can all find out if there really is
or is not one



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No it's not, it's supposed to just be a framework for developers and
resellers to create GUIs that can go on the appliance.

On 9/19/06, shadowym [EMAIL PROTECTED] wrote:



I am talking about the GUI that was announced as part of the new Asterisk
Appliance.

Sounds like it is going to be a full featured GUI like FreePBX.


-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Monday, September 18, 2006 8:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digium GUI?

 So the press announcement said that the new Digium GUI will be
 available in
 v1.4 sometime in Oct.  Is the GUI already there in Trunk or is there
 some other branch of development that the general public cannot access?

Do you mean this?

http://svn.digium.com/view/asterisk/trunk/static-http/



On 9/18/06, Don Fanning [EMAIL PROTECTED] wrote:
 You mean the menuselect ncurses screen?  If yes, then yes... it's a
 gui. :)

 -Original Message-
 From: shadowym [mailto:[EMAIL PROTECTED]
 Sent: Monday, September 18, 2006 4:43 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Digium GUI?


 So the press announcement said that the new Digium GUI will be
 available in
 v1.4 sometime in Oct.  Is the GUI already there in Trunk or is there
 some other branch of development that the general public cannot access?

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--

Mitchel Constantin
Snap - A desktop user interface for Asterisk
www.snapanumber.com





Re: [asterisk-users] Digum g729 and g723

2006-09-04 Thread brandon kruz

haha
im reporting this
but i think your a nub
and what happend was you just took the actual files
so it wasnt hard, you just dont have a physical liscense
there are things all over the web on how to enable g729
its not hard
its the liscensing that can get you sued for a very much amount of money...


you can get any software you want and illegally install it
using that in an enterprise envirnment, thats just stupid, and you will get 
caught.



of course if this is in fact legit, im very very sorry, im a nub
but from the sound of it, it sounds illegal, you will get caught

fess up
get a job
a pay for business edition

my 2 cents
`KruZ~

please someone let me know if this is in fact legit, else i am now 
researching this, and it will be stopped


please dont do anything illegal and stupid
it doesnt make you any more 1337 than you can find ways to
enable these codecs without paying for them

its your chance to get sued.
go for it.




From: Kannaiyan Natesan [EMAIL PROTECTED]
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To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com, Commercial and 
Business-Oriented Asterisk Discussionasterisk-biz@lists.digium.com, 
Asterisk Developers Mailing List asterisk-dev@lists.digium.com

Subject: Re: [asterisk-users] Digum g729 and g723
Date: Tue, 5 Sep 2006 10:47:37 +0800
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Hey,

Is this code released by Digium?
Looks like directly from digium. Is it GPL with License and Royalty?

Unlimited channels and no restriction !

Author mentioned as Mark Spencer.
If we want to pay the license fees, should we have to Pay to VoiceAge 
directly?


Hats off. Thanks once again to Mark.

Kannaiyan

On 9/5/06, Joe shmoe [EMAIL PROTECTED] wrote:

Would you like to have the codecs written by Mark
Spencer for Asterisk?  The same binary codecs
available when you purchase a licence?  You're in
luck!  The following link will allow you to have
Digiums codecs.


http://www.savefile.com/files/20972

Come one come all!


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Re: [asterisk-users] Error in ubuntu dapper

2006-07-22 Thread brandon kruz

look for another, more configurable softphone if you can

but asterisk needs that for incoming sip(or maybe its iax, not sure)

you should be able to configure the differnet port
you want to use a softphone on your asterisk system
and btw netstat -a | grep 5060 worked for me

but who knows

let me know how it goes, im going to do some more research



From: don Paolo Benvenuto [EMAIL PROTECTED]
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Subject: Re: [asterisk-users] Error in ubuntu dapper
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Yeah, ekiga was using port 5060, althoug netstat -a didn't say it.

I might issue netstat -alpn

The weird thing is that I had configured ekiga so that it used port
5061, but unfortunatly if ekiga is run before asterisk it catchs port
5060 too.

El vie, 21-07-2006 a las 18:13 -0500, brandon kruz escribió:
 in addition to russel
 use
 (in ubuntu)
 sudo netstat
 or man netstat for further, more precise methods
 look for your specific port
 eg
 sudo netstat -a | grep 5060
 and it shoudl tell you the process name, and what directory it is 
comming

 from
 shut it off
 and do that
 sudo netstat -a | grep 5060 again
 it should be clear
 then start asterisk :]

 keep us updated.

 the fact that it is from module chan_sip, as russel said
 i believe its a sip phone indeed(softphone, X-lite, SJ-phone, etc)

 
 On Fri, 2006-07-21 at 12:37 -0400, don Paolo Benvenuto wrote:
   Jul 21 12:31:51 WARNING[6333]: chan_sip.c:12637 reload_config: 
Failed to

 bind to 10.152.58.9:5060: Address already in use
 
 It looks like another application on your system is using port 5060.
 Did you install any new software such as a soft phone?
 
 If you are now using another application that wants to use port 5060,
 you will need to configure one of them to use a different port.
 

--
Buon Cammino!

don Paolo Benvenuto

Vuoi sapere di più su quello che succede qui?
leggi il mio diario a http://www.chiesamissionaria.it/diario

Visita l'enciclopedia libera, dove puoi contribuire anche tu:
http://it.wikipedia.org/

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RE: [asterisk-users] How to connect XLite with public IP?

2006-07-22 Thread brandon kruz


unfortunatly
its not a firewall problem(or at least thats not the main one)
the main one is forwarding the asterisk related ports to your IP(internal 
IP)

go to http://whatismyip.com
make sure u keep that information

then go to your asterisk box and type ifconfig(and get your itnernal IP)
then type your default gateway(your router) ip (usually 192.168.x.x, 
linksys=192.168.1.1, dlink=192.168.0.1, u can figure that out probably)


and type that into your itnernet browser
and forward ports to your asterisk box(the IP adress that asterisk is on)

hope this helps
`KruZ~


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Subject: [asterisk-users] How to connect XLite with public IP?
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   Hi Friends,

I have installed Asterisk in my house with public ip. Now, I want to 
connect to my Asterisk server from my office system with XLite softphone. I 
was given my Asterisk server public IP in my XLite Domain field. But, it is 
not connecting. It is giving an error i.e.,  Registration error: 408 - 
Request timedout. I tried using firewall and without using firewall. 
Please tell me how to configure my XLite softphone to connect with my 
Asterisk server?


Looking forward to your kind response. Thank you.

Regards,
Chandra.

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Re: [asterisk-users] NAT and externip problem or bug

2006-07-22 Thread brandon kruz


specially if you only have 5060 but not the sip pass through ports as he 
mentioned
(1-2) then you can establish a call but no voice data is sent(or so 
that is in theory)


so make sure you get this right with your asterisk box and everything should 
work



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Subject: Re: [asterisk-users] NAT and externip problem or bug
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On Jul 22, 2006, at 11:13 AM, Robert Jenkins wrote:


Oh well..

I already had localnet set:

localnet = 192.168.0.0  ; Internal NETWORK address
localmask = 255.255.255.0   ; Internal netmask

All the involved PCs  Sipura boxes are using 192.168.0.x addresses.

The Sipura boxes work, but the fact that asterisk is sending the  external 
IP

to any device on the local network seems to me to be a bug..



You didn't mention whether you were also forwarding ports 1-2  to 
the SIP Proxy (ie asterisk).  Thats where the actual RTP (voice  data) is 
passing.  Also you need to be sure that there aren't  multiple clients on 
your lan all trying to use the same ports for  signaling (ie 5060), as this 
will fail.


Hope this helps.
Marty

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Re: [asterisk-users] Ztdummy

2006-07-22 Thread brandon kruz

im not sure of this error
but it has to be common if it is from to quick of a startup
(but the module zaptel is very quick to probe)
if worst comes to worst
this modprobeing should be in a startup script i imagine
you can just throw down a wait command(depending on what version of linux, 
and

what you are using as a language to startup the modprobe(bash? perl? etc.)

let me know if a little waiting helps

`KruZ~



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Subject: Re: [asterisk-users] Ztdummy
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At 09:27 PM 7/18/2006, you wrote:

Using Zaptel-1.2.7
Asterisk 1.2.10
OS: CentOS 3.4

I am having a problem trying to get ztdummy and it wont work. Here is what 
I did the following and got:

[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

FATAL: Error running install command for ztdummy
[EMAIL PROTECTED] ~]#

What I find interesting is that timing will work. However I don't feel 
comfortable letting the client use the system if this can affect him in 
anyway. Thanks.


Dovid


I get this if I'm too quick to execute the commands - zaptel takes some 
time to come up.



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Re: [asterisk-users] cannot received calls in pstn line

2006-07-22 Thread brandon kruz

sounds like you have a TDM card
with an fxo port(FOR YOUR PSTN, i think)
and 1-3 fxs ports for your phone lines
make sure you have your zaptel.conf
and zapata.conf and extensions.conf right
with the context that your ports are in
and the signalling that it uses(oppisite of the port)
be careful because u can burn out your card and maybe phone line if u do 
this wrong



itll look like this
[incoming]
exten = 100,1,Answer()
exten = 100,2,Dial(Zap/1,20,rR)
exten = 101,1Answer()
exten = 101,2,Dial(Zap/2,20,rR)

etc...


let me know if this helps for i think thats what you are asking

the redmodule is the module you will plug your pstn line into
then you have to set it up in zapata.conf and zaptel.conf to show your 
signalling and context
obviously you will want to name it something like [incoming] and make sure 
you have an

[incoming] context in extensions.conf

`KruZ~



From: Alex Robar [EMAIL PROTECTED]
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Subject: Re: [asterisk-users] cannot received calls in pstn line
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Agreed... How are your extensions connected? What hardware are you using to
connect your PSTN lines? Rephrase your question and we can probably help 
you

out.

Alex

On 7/22/06, Martin Joseph [EMAIL PROTECTED] wrote:



On Jul 22, 2006, at 3:00 AM, Lito Lampitoc wrote:

 Hi All,

 I'm having problems receiving calls in my direct lines, but it's
 working fine in local lines (extensions). When a direct line is
 connected to my fxo it can't handle the call, but when an extension
 is connected it's ok.

 Any suggestion?

I think you might try rephrasing your question as it currently
doesn't seem to make any sense?
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--
Alex Robar
[EMAIL PROTECTED]




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RE: [asterisk-users] X100P clone not working

2006-07-22 Thread brandon kruz

dmesg | grep x100p

also
im almost positive(correct me if im wrong anyone) you do NOT
need ztdummy
i repeat, do NOT need ztdummy with x100p

then rmmod zaptel  rmmod wctdm  modprobe zaptel  modprobe wctdm
or wctfxo or w/e

let me know how this works out
(post the section for your x100p from zaptel.conf and zapata.conf if you 
could m8)





Hi,

I have problem to set up an X100P clone card.
Installation of zaptel was successful.
Also modprobe of zaptel, ztdummy and wcfxo without problems.

kernel: wcfxo: module not supported by Novell, setting U taint flag.
kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - IRQ 
193

kernel: wcfxo: DAA mode is 'FCC'
kernel: Found a Wildcard FXO: Generic Clone
kernel: Registered tone zone 0 (United States / North America)


But if I do #ztcfg -vvv no channels will be found.

the log file reports me: kernel: Registered tone zone 0 (United States / 
North

America)

Iam quite sure that zaptel.conf is correct.
fxsks=1
loadzone=us
defaultzone=us

So Iam not sure what happened here, I have found that this clone as used by
many others, so it should work.

thanks

best regards

Frank
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Re: [asterisk-users] Ztdummy

2006-07-22 Thread brandon kruz

i thought so because ztdummy isnt even necessary
unistall and comment out :D

(rmmod ztdummy and look at your modprobe.conf as stated by cohen, ty cohen)

good luck, keep us updated.



From: Tzafrir Cohen [EMAIL PROTECTED]
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Subject: Re: [asterisk-users] Ztdummy
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On Wed, Jul 19, 2006 at 03:27:03AM -0400, Dovid Bender wrote:
 Using Zaptel-1.2.7
 Asterisk 1.2.10
 OS: CentOS 3.4

 I am having a problem trying to get ztdummy and it wont work. Here is 
what I did the following and got:

 [EMAIL PROTECTED] ~]# modprobe zaptel
 [EMAIL PROTECTED] ~]# modprobe ztdummy
 Notice: Configuration file is /etc/zaptel.conf
 line 0: Unable to open master device '/dev/zap/ctl'

The error is because of an unnecessary post-load action of the module
ztdummy.

just rem-out the 'ztdummy' line in the modprobe.conf that zaptel
installs. (or better: don't let it install anything there)

There is no need to configure ztdummy's span.

--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] X100P clone not working

2006-07-22 Thread brandon kruz


configs look fine to me
it has to be a module problem im guessing


try rmmod for zaptel and wctfxo or w/e then modprobing again(must be root)

if that does not work(which is does on my system, so i added it into a 
startup conf)



kernel: wcfxo: module not supported by Novell

are you sure that wcfxo is the correct module for that device??

hoping to help
`KruZ~






From: Frank Darner [EMAIL PROTECTED]
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Subject: Re: [asterisk-users] X100P clone not working
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Hi,
i tried w/o ztdummy but nothing changed.

any ideas?


zaptel.conf
--
fxsks=1
loadzone=us
defaultzone=us

zapata.conf

[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
echocancle=yes
transfer=yes
echocanclewhenbridged=yes
channel = 1

# modprobe zaptel  modprobe wcfxo

kernel: zaptel: module not supported by Novell, setting U taint flag.
kernel: Zapata Telephony Interface Registered on major 196
kernel: Zaptel Version: SVN-tag-1.2.6-r1096 Echo Canceller: KB1
kernel: wcfxo: module not supported by Novell, setting U taint flag.
kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - IRQ 
193

kernel: wcfxo: DAA mode is 'FCC'
kernel: Found a Wildcard FXO: Generic Clone
kernel: Registered tone zone 0 (United States / North America)

# lsmod | grep zaptel
zaptel201732  1 wcfxo
crc_ccitt   2176  1 zaptel


# ztcfg -v
Zaptel Configuration
==
0 channels configured.



 dmesg | grep x100p

 also
 im almost positive(correct me if im wrong anyone) you do NOT
 need ztdummy
 i repeat, do NOT need ztdummy with x100p

 then rmmod zaptel  rmmod wctdm  modprobe zaptel  modprobe wctdm
 or wctfxo or w/e

 let me know how this works out
 (post the section for your x100p from zaptel.conf and zapata.conf if you
 could m8)

 Hi,
 
 I have problem to set up an X100P clone card.
 Installation of zaptel was successful.
 Also modprobe of zaptel, ztdummy and wcfxo without problems.
 
 kernel: wcfxo: module not supported by Novell, setting U taint flag.
 kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - 
IRQ

 193
 kernel: wcfxo: DAA mode is 'FCC'
 kernel: Found a Wildcard FXO: Generic Clone
 

Re: [asterisk-users] X100P clone not working

2006-07-22 Thread brandon kruz

here is some more information
http://www.voip-info.org/wiki/view/X100P+clone



From: Frank Darner [EMAIL PROTECTED]
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Subject: Re: [asterisk-users] X100P clone not working
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Hi,
i tried w/o ztdummy but nothing changed.

any ideas?


zaptel.conf
--
fxsks=1
loadzone=us
defaultzone=us

zapata.conf

[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
echocancle=yes
transfer=yes
echocanclewhenbridged=yes
channel = 1

# modprobe zaptel  modprobe wcfxo

kernel: zaptel: module not supported by Novell, setting U taint flag.
kernel: Zapata Telephony Interface Registered on major 196
kernel: Zaptel Version: SVN-tag-1.2.6-r1096 Echo Canceller: KB1
kernel: wcfxo: module not supported by Novell, setting U taint flag.
kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - IRQ 
193

kernel: wcfxo: DAA mode is 'FCC'
kernel: Found a Wildcard FXO: Generic Clone
kernel: Registered tone zone 0 (United States / North America)

# lsmod | grep zaptel
zaptel201732  1 wcfxo
crc_ccitt   2176  1 zaptel


# ztcfg -v
Zaptel Configuration
==
0 channels configured.



 dmesg | grep x100p

 also
 im almost positive(correct me if im wrong anyone) you do NOT
 need ztdummy
 i repeat, do NOT need ztdummy with x100p

 then rmmod zaptel  rmmod wctdm  modprobe zaptel  modprobe wctdm
 or wctfxo or w/e

 let me know how this works out
 (post the section for your x100p from zaptel.conf and zapata.conf if you
 could m8)

 Hi,
 
 I have problem to set up an X100P clone card.
 Installation of zaptel was successful.
 Also modprobe of zaptel, ztdummy and wcfxo without problems.
 
 kernel: wcfxo: module not supported by Novell, setting U taint flag.
 kernel: ACPI: PCI Interrupt :05:04.0[A] - GSI 16 (level, low) - 
IRQ

 193
 kernel: wcfxo: DAA mode is 'FCC'
 kernel: Found a Wildcard FXO: Generic Clone
 kernel: Registered tone zone 0 (United States / North America)
 
 
 But if I do #ztcfg -vvv no channels will be found.
 
 the log file reports me: kernel: Registered tone zone 0 (United States 
/

 North
 America)
 
 Iam quite sure that zaptel.conf is correct.
 fxsks=1
 loadzone=us
 

RE: [asterisk-users] newbbie question

2006-07-22 Thread brandon kruz


hey pablo
i havent messed aorund with stun that much except as a repeator
but maybe this is your problem ;]


exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])
should be
exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])
notice the $ sign in front of {EXTEN} that declares it a variable(haha like 
my oxymoron?)

tells asteirsk to lookup the extensions dialed by the user




Hello, I am playing with my new * install, and there are a couple of things
that I don't understand, if someone could point me in the right direction 
it

will be appreciated.

I am trying to configure a voipstunt.com account to place outgoing calls,
and this is my config.

sip.conf:

[voipstunt]
type=friend ; (or peer if we don't need incoming calls, or if there is a
separate section with type=user)
host=sip.voipstunt.com
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
username=voipstuntuser_replacement
fromuser=voipstuntuser_replacement
secret=hiddenpassword
qualify=1000 ; optional
canreinvite=no ; new SIP servers don't like reINVITEs
dtmfmode=inband ; only inband currently works, and not that well

extensions.conf:

[internal]
exten = 787793,1,Dial(SIP/john)
exten = 700099,1,Dial(SIP/maribel)
exten = 10,1,Dial(SIP/dieguez)
exten = 11,1,Dial(SIP/chparson)
exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])


As stated in asteriskTFOT _NXXX will match 541152184829 which is 
the

phone number of my place, which I am trying to place a call.

I am asuming that the sign + in (SIP/[EMAIL PROTECTED]) will be appended 
to

what I press in my softphone.

All I get when I call to 541152184829 is:

-- Executing Dial(SIP/john-0819a010, SIP/[EMAIL PROTECTED]) in 
new

stack
-- Called [EMAIL PROTECTED]
Jul 22 20:15:45 NOTICE[1396]: chan_sip.c:1997 auto_congest: Auto-congesting
SIP/voipstunt-0819f520
-- SIP/voipstunt-0819f520 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/john-0819a010' status is 'CONGESTION'


Any idea? suggestion?

Thanks in advance for any comment/help.

Pablo

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RE: [asterisk-users] Asterisk Dial Plan to Play Message

2006-07-22 Thread brandon kruz

youll have to decide what context this goes in
either
[internal]
or [incoming]
but i hope you can figure this out yourself
here is an idea

[internal]
exten = s,1,Answer()
exten = s,n,Playback(pbx-invalid)
exten = s,n,Hangup()

so now when a user from [internal] picks up the phone the pbx-invalid.gsm is 
played from

the asterisk sounds directory
so now you are saying to yourself i want my own voice, to record
check this out

[internal]
exten = s,1,Answer()
exten = s,n,Playback(custom)
exten = s,n,Hangup()

(dont tell anyone this part, or just add it temporarily)

exten = 999,1,Answer()
exten = 999,n,Record(custom.gsm)
exten = 999,n,Wait(1)
exten = 999,n,Playback(custom)

you will hear a beep after u dial 999, start recording
then hangup or i think u can type # to stop and move to the next option



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Subject: [asterisk-users] Asterisk Dial Plan to Play Message
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Anyone know how to use dial plan to play messages as soon as a phone is 
picked up. Like when a user picks up a phone, get a message to contact 
administrator instead of a dial tone?




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Re: [asterisk-users] newbbie question

2006-07-22 Thread brandon kruz

no problem
most of the time
it is these annoying little problems

stay in touch
glad i could be of assistance :D



From: Pablo L. Arturi [EMAIL PROTECTED]
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 exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])
 should be
 exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED])
 notice the $ sign in front of {EXTEN} that declares it a variable(haha
like
 my oxymoron?)
 tells asteirsk to lookup the extensions dialed by the user

Brandon, that whas exactly the problem.

Thank you very much!!! now I am calling :)


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Re: [asterisk-users] Asterisk Dial Plan to Play Message

2006-07-22 Thread brandon kruz

it will either way work will it not
just because there is more way to do something doesnt mean it wrongs sir

bette ot have answer
then to not have answer on applications that will no
its a good practice




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Subject: Re: [asterisk-users] Asterisk Dial Plan to Play Message
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brandon kruz wrote:

youll have to decide what context this goes in
either
[internal]
or [incoming]
but i hope you can figure this out yourself
here is an idea

[internal]
exten = s,1,Answer()
exten = s,n,Playback(pbx-invalid)
exten = s,n,Hangup()


*sigh*

Playback will BY DEFAULT answer the line.  The only time you need an 
Answer() before a Playback() is if you want a Wait() between them.


Doesn't anyone read the docs for the applications they use?  For a good 
time type: show application playback


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RE: [asterisk-users] Asterisk Dial Plan to Play Message

2006-07-22 Thread brandon kruz

thank you russel
forgot to mention this.



From: Russell Bryant [EMAIL PROTECTED]
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Subject: RE: [asterisk-users] Asterisk Dial Plan to Play Message
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On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote:
 [internal]
 exten = s,1,Answer()
 exten = s,n,Playback(custom)
 exten = s,n,Hangup()

This, by itself, does not solve the problem where you want the message
to be played back when the phone is picked up without any user
intervention.  If you're using zap phones, you can simply set this
option:

immediate=yes

Then, as soon as the phone goes off hook, the call will begin at the 's'
extension in the configured context instead of providing dialtone.

--
Russell Bryant
Software Developer
Digium, Inc.

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RE: [asterisk-users] Invalid module format (ztdummy)

2006-07-21 Thread brandon kruz

are you using an actual card that needs zaptel?
because if so you wont need ztdummy for the timing, the timing is included 
in the hardware



thats what i first think when i see this
the second thing is the fact that it said you GCC version is to new
that makes no sence
but you can remove that particular version, and apt-get(if in debian)
the exact version
or dl from gcc themselves
hope this helps
keep me updated



From: Pablo L. Arturi [EMAIL PROTECTED]
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Hello listo.

I am trying to install asterisk in a VPS (OpenVZ) and I am following steps
from http://www.telephreak.org/papers/vpa/

As stated there, I need to install zaptel in the hardware node, and then
load the module ztdummy in Hardware node to be available in VPS.

compilation and install went ok, but now I try to load the module and get
some errors:

[EMAIL PROTECTED] zaptel-1.2.7]# modprobe ztdummy
WARNING: Error inserting zaptel
(/lib/modules/2.6.8-022stab078.14/extra/zaptel.ko): Invalid module format
WARNING: Error inserting zaptel
(/lib/modules/2.6.8-022stab078.14/extra/zaptel.ko): Invalid module format
FATAL: Error inserting ztdummy
(/lib/modules/2.6.8-022stab078.14/extra/ztdummy.ko): Invalid module format
FATAL: Error running install command for ztdummy

I tried a dmesg to see if there was something there and I found:

[EMAIL PROTECTED] zaptel-1.2.7]# dmesg
...
zaptel: version magic '2.6.8-022stab078.14 686 gcc-3.4' should be
'2.6.8-022stab078.14 686 gcc-3.3'
zaptel: version magic '2.6.8-022stab078.14 686 gcc-3.4' should be
'2.6.8-022stab078.14 686 gcc-3.3'
ztdummy: version magic '2.6.8-022stab078.14 686 gcc-3.4' should be
'2.6.8-022stab078.14 686 gcc-3.3'

Any idea how to fix this?

Thanks in advance,
Pablo

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Re: [asterisk-users] Error in ubuntu dapper

2006-07-21 Thread brandon kruz

in addition to russel
use
(in ubuntu)
sudo netstat
or man netstat for further, more precise methods
look for your specific port
eg
sudo netstat -a | grep 5060
and it shoudl tell you the process name, and what directory it is comming 
from

shut it off
and do that
sudo netstat -a | grep 5060 again
it should be clear
then start asterisk :]

keep us updated.

the fact that it is from module chan_sip, as russel said
i believe its a sip phone indeed(softphone, X-lite, SJ-phone, etc)



On Fri, 2006-07-21 at 12:37 -0400, don Paolo Benvenuto wrote:
 Jul 21 12:31:51 WARNING[6333]: chan_sip.c:12637 reload_config: Failed to 
bind to 10.152.58.9:5060: Address already in use


It looks like another application on your system is using port 5060.
Did you install any new software such as a soft phone?

If you are now using another application that wants to use port 5060,
you will need to configure one of them to use a different port.

--
Russell Bryant
Software Developer
Digium, Inc.

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RE: [asterisk-users] If you prefer to read this mail list as a forum ...

2006-07-20 Thread brandon kruz

this is very cool and saves my inbox from all these messages
and i really like being able to view all them them
thank you
props
2 kudos



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Hi,
I got realy tired of looking at Asterisk lists in Outlook so I moved it 
into the phpBB2 type forum. It seems to be working well for me and I think 
some of you may find it usefull too.

So here it is at:
http://forum.globalvoicenet.com/
Take it for a spin and let me know if you like the idea.
Thanks,
robert






Read this topic online here:
http://forum.globalvoicenet.com/viewtopic.php?p=3738#3738








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