Re: [Asterisk-Users] Billsec on CDR

[Tan Aks]

Isn't there any way to make callprogress work for people in Europe? What is
it that is needed to make it work?

T



- Original Message - 
From: "Martin Pycko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, June 19, 2003 11:36 PM
Subject: Re: [Asterisk-Users] Billsec on CDR


It has to do with the fact that with analog channels like FXO
we don't have a way to tell whether the call has been answered or not.
So after the interfaces sends the called number we assume that the
call got answered. This happens unless you have callprogress=yes
in zapata.conf. But it's designed to be working only in US.

Martin

On Thu, 19 Jun 2003, Dan Fernandez wrote:

> I have an X100P and when I place calls to the PSTN which are not answered,
the Billsec field of the CDR still logs the seconds that the phone rang.
>
> Can someone please confirm that this has to do with the ringcadance of the
indications.conf file? Is there anything else I need to check ?
>
> Thanks in advance
>

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[Asterisk-Users] databases for billing

[carlos del mayor]

hi
I want to do a database to save the cdr with a billing finality. I've created the database in mysql (thanks for the table and all that!) but I'm not sure of how to 'connect' asterisk to that database in order to save there the cdr. Is the cdr_mysql.conf what I have to config? Or must I do a script, with the 'database' AGI commands? 
Any help would be so apreciated!Thanks a lot
carlosYahoo! SorteosJuega a la Lotería Primitiva sin salir de casa

Re: [Asterisk-Users] Chan_oh323 problem

[Michael Manousos]

Andrey Tyushkin wrote:
Hello
 
I have the following problem using chan_oh323
I have DialGate 2160 for SystemBas (www.sysbas.com 
) connected to PSTN
(H.323 to FXO/FXS gateway)
when i try to make call form one pstn phone to other trough asterisk or 
when i make call from software h.323 client trough asterisk and this 
gateway to pstn i have the problem with voice quality.
The side that initiated call can be heared clearly, but the side that 
answered the call can be heared aproximatly at half.
E.g. on this side everything is heared with some cuts.
I runed asterisk -vvvc
and during the call it constantly printning:
WARNIGN .. File chan_oh323.c, Line 761 (oh323_read) H323:20016 Requested 
read buffer size is too long (2560) !
and next line message about truncating buffer.
 
I've changed OH323_MAX_BUF to 2560 - warning message disappeared but it 
didn't improove the quality.
 
These messages are just warnings, voice data don't get lost.
I think that the problem is on the direction from Asterisk
to the H.323 endpoint. There is a problem on this direction.
Some RTP packets have wrong timestamps and are dropped on the other
end, causing some voice gaps.
I think I have fixed it (some more test), and soon I will make
a new release.

I configures Milliwatt aplication and when i ring from PSTN i hear 
beeps, not continuees tone
and its printning warning message in line 1460 : Only doing 640 bytes 
(1280 bytes requested)
(sometimes prints 2560 bytes requested)
(from soft client almost everything is ok, only some seldom clicks)
 
I thought that this is the problem of dialgate, but when i use it w/o 
asterisk it works w/o this problems.
 
Thanks in advance,
Andrey


Michael.

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[Asterisk-Users] [HS] results testing asterisk with ISDN BRI & look for help tounderstand configuring SIP with asterisk

[Hervé Thibaud]

configuration
ISDN BRI
card : ISDN Olitec PCI 128 (hisax gazel)
internet connection by ISDN 64kb/s
dynamic IP 
nom de domaine registered to : dyndns.org avec ddclient to register IP
par ddclient
asterisk (on internet gateway)
configuration pour ISDN BRI par virtual modems /dev/ttyI* (modem.conf)
logical telephone SIP "SJPHONE" on 2 local stations "windows"
(i don't succeed to use telephon SIP X-lite with asterisk)

testing résults with "asterisk"
SJPHONE local -> IVR asterisk   : OK
extern telephon (analogic) -> SJPhone : OK
SJphone -> extern telephon  : OK
extern telephon -> SJPHONE : OK
local network SJPhone ->local network SJPhone (with asterisk) OK
configuration sjphone : 
Use Local OuntBound Proxy (selected)
Proxy IP address 192,168,0,1 port 5060
caller ID : SIP @domain.dyndns.org (stations défined dans
/etc/asterisk/sip.conf)

I don't understand what i have to make and set to communicate with external telephons 
SIP (Sjphone, X-lite, MS messenger ...)
Must i have a SIP provider subscription, how to integrate this subscription with 
asterisk 

The purpose i have is to keep control with asterisk to tape, redirect, establish 
conference ... with communicates

I am "swimming" with (english) documentation anglaise
and i understand very badly asterisk system, my knowledge in system software an linux 
is too low

But with your patient help, i am sure i'll reach

thanks to help me


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[Asterisk-Users] Asterisk VS. Bayonne

[K a z]

Could someone familiar with both break down the most memorable pro's & con's 
and why you have decided to use Asterisk?

Thanks

_
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Re: [Asterisk-Users] [HS] results testing asterisk with ISDN BRI & look for help to understand configuring SIP with asterisk

[Andy Powell]

Hi,

>I don't understand what i have to make and set to communicate with
>external telephons SIP (Sjphone, X-lite, MS messenger ...)
>Must i have a SIP provider subscription, how to integrate this
>subscription with asterisk 

Do you mean internally i.e. Sjphone, X-lite, MS messenger phones
on your pc's or other people - out there on the net?

You could take a look at my guide - it may help explain things
(then again it may not)

http://www.automated.it/guidetoasterisk.htm

I recently had to move hosting co's, just noticed the one I moved
was old!! I've updated it...


>I am "swimming" with (english) documentation anglaise

You're lucky, I'm English and I have trouble speaking it!

HTH

Andy


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[Asterisk-Users] Active ISDN PCMCIA card

[Michael Manousos]

Are there any suggestions for active ISDN CAPI PCMCIA cards
that are known to work with Asterisk?
Thanks,
Michael.


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[Asterisk-Users] Firewalling, Ports and rtp.conf..

[WipeOut .]

Hi,

Am I correct in this..

I want to setup IPTABLES to protect my * box..

The default rtp.conf defines that * will use ports 1 to 2..

IAX listens on 5036..

SIP listens on 5060..

I am assuming all ports used by * are UDP..

So I am planning on setting my server to block all inbound traffic except UDP ports 
5060, 5036 and 1-2..

Am I leaving anything out??

Thanks..
-- 
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[Asterisk-Users] Asterisk hogging CPU resources

[Derek Beaumont]

Here's the problem:
I start asterisk, and it takes up around 3-4% of my CPU
resources.
However, this number continues to climb over the hours until it
is close to 100%.
Usually it takes around a day to climb up to approximately 95 or
96%

Has anybody experienced the following problem before?



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Re: [Asterisk-Users] Asterisk VS. Bayonne

[James Sizemore]

Asterisk, kind of has support for SIP, Bayonne has none at all. (Last 
time I checked.)

K a z wrote:

Could someone familiar with both break down the most memorable pro's & 
con's and why you have decided to use Asterisk?

Thanks

_
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RE: [Asterisk-Users] Asterisk VS. Bayonne

[Uriel Carrasquilla]

I certainly hope that there is more to their difference.  I have not
compared both of them recently but I did back during summer 99.  Then in my
mind I decided that the one that I could get working first would stay.  I am
still with asterisk.  I found that Mark was more than willing to help to
help me get going.  Bayonne wanted consulting fees that were expensive on my
zero budget.
Asterisk back in 99 had its problem but it was promissing:
1) inexpensive Zapata cards
2) very effective, simple communication protocol (iax) that would work over
firewalls and nat.
3) very straight forward dial plan (extensions.conf)
4) gno-phone was on the works.

Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James
Sizemore
Sent: Friday, June 20, 2003 9:28 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk VS. Bayonne


Asterisk, kind of has support for SIP, Bayonne has none at all. (Last
time I checked.)

K a z wrote:

>
> Could someone familiar with both break down the most memorable pro's &
> con's and why you have decided to use Asterisk?
>
> Thanks
>
> _
> Add photos to your e-mail with MSN 8. Get 2 months FREE*.
> http://join.msn.com/?page=features/featuredemail
>
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[Asterisk-Users] Free World Dialup change which may be relevant to *

[James H. Cloos Jr.]

Jeff just posted on [EMAIL PROTECTED] about some changes to
fwd, and mentioned that at least some clients may need to register
@fwd.pulver.com rather than @192.246.69.223.

Given the recent thread either here or on -dev about the syntax of
the register command for sip peers, I suspect the register command
for fwd peering may need to change from eg:

register => 11596:[EMAIL PROTECTED]

to something like:

register => [EMAIL PROTECTED]:[EMAIL PROTECTED]

to be able to continue to register w/ fwd.

(At least if your checkout is new enough to grok that syntax)

-JimC

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Re: [Asterisk-Users] Asterisk hogging CPU resources

[Steven Critchfield]

On Fri, 2003-06-20 at 08:32, Derek Beaumont wrote:
> Here's the problem:
>   I start asterisk, and it takes up around 3-4% of my CPU
> resources.
>   However, this number continues to climb over the hours until it
> is close to 100%.
>   Usually it takes around a day to climb up to approximately 95 or
> 96%
> 
> Has anybody experienced the following problem before?

What kind of interfaces are you using?

I'm using zap and IAX on my main asterisk server that deals in about
300-400 calls a day without the cpu load you are seeing.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Billsec on CDR

[Martin Pycko]

You need to change the FREQs for the events. I don't know exactly how the
code works. There was someone on the list that claimed to have the UK
callprogress working.

regards
Martin

On Fri, 20 Jun 2003, Tan Aks wrote:

> Isn't there any way to make callprogress work for people in Europe? What is
> it that is needed to make it work?
>
> T
>
>
>
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, June 19, 2003 11:36 PM
> Subject: Re: [Asterisk-Users] Billsec on CDR
>
>
> It has to do with the fact that with analog channels like FXO
> we don't have a way to tell whether the call has been answered or not.
> So after the interfaces sends the called number we assume that the
> call got answered. This happens unless you have callprogress=yes
> in zapata.conf. But it's designed to be working only in US.
>
> Martin
>
> On Thu, 19 Jun 2003, Dan Fernandez wrote:
>
> > I have an X100P and when I place calls to the PSTN which are not answered,
> the Billsec field of the CDR still logs the seconds that the phone rang.
> >
> > Can someone please confirm that this has to do with the ringcadance of the
> indications.conf file? Is there anything else I need to check ?
> >
> > Thanks in advance
> >
>
> ___
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>
>
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Re: [Asterisk-Users] Firewalling, Ports and rtp.conf..

[Angelo Sampietro]

Friday, June 20, 2003, 3:22:20 PM, you wrote:

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W> Subject: [Asterisk-Users] Firewalling, Ports and rtp.conf..
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W> Date: Fri, 20 Jun 2003 13:22:20 +

W> Hi,

W> Am I correct in this..

W> I want to setup IPTABLES to protect my * box..

W> The default rtp.conf defines that * will use ports 1 to 2..

W> IAX listens on 5036..

W> SIP listens on 5060..

W> I am assuming all ports used by * are UDP..

W> So I am planning on setting my server to block all inbound traffic except UDP ports 
5060, 5036 and 1-2..

W> Am I leaving anything out??

W> Thanks..


but why you wanna protect your * box in this wey?
it should be with a provate IP address afther a firewall so from
outside nobody should attack your box...
and if you need connectivity with other networks you should use
VPN's...

BTW, the port that you wrote are what you need :)

regards

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Re: [Asterisk-Users] databases for billing

[Martin Pycko]

cdr_mysql.conf

On Fri, 20 Jun 2003, carlos del mayor wrote:

> hi
> I want to do a database to save the cdr with a billing finality. I've created the 
> database in mysql (thanks for the table and all that!) but I'm not sure of how to 
> 'connect' asterisk to that database in order to save there the cdr. Is the 
> cdr_mysql.conf what I have to config? Or must I do a script, with the 'database' AGI 
> commands?
> Any help would be so apreciated!
> Thanks a lot
> carlos
>
>
> -
> Yahoo! Sorteos
> Juega a la Lotería Primitiva sin salir de casa



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Re: [Asterisk-Users] Asterisk hogging CPU resources

[Lubomir Christov]

Yes, I have such a problem .
On my asterisk box (P4 2.4MhZ) I'm doing SIP/SIP and H323/SIP 
(chan_h323) bridging up to 20-30 channels.

Derek Beaumont wrote:
Here's the problem:
I start asterisk, and it takes up around 3-4% of my CPU
resources.
However, this number continues to climb over the hours until it
is close to 100%.
Usually it takes around a day to climb up to approximately 95 or
96%
Has anybody experienced the following problem before?



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[Asterisk-Users] Asterisk hogging CPU resources

[Derek Beaumont]

The interfaces I'm using are 2 X100Ps and a TDM400P




What kind of interfaces are you using?

I'm using zap and IAX on my main asterisk server that deals in about
300-400 calls a day without the cpu load you are seeing.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] Error compiling, is it only mee?

[Christopher Arnold]

Hi all,

since a couple of days i have been wanting to upgrade Asterisk, but my
system complains. (Se errors below) Since im ripping the cvs version i
guessed that it was something temporary. But it seems persistant.

Is this only me or is it happening to other people?

Anyone who has a workaround?
Or naturally even better, a solution...

Oh, i almost forgot im running:
# uname -a
Linux sip.rabbta.com 2.4.20-18.8 #1 Thu May 29 07:20:39 EDT 2003 i686
athlon i386 GNU/Linux
# gcc -v
Reading specs from /usr/lib/gcc-lib/i386-redhat-linux/3.2/specs
Configured with: ../configure --prefix=/usr --mandir=/usr/share/man
--infodir=/usr/share/info --enable-shared --enable-threads=posix
--disable-checking --host=i386-redhat-linux --with-system-zlib
--enable-__cxa_atexit
Thread model: posix
gcc version 3.2 20020903 (Red Hat Linux 8.0 3.2-7)


/Chris





make[1]: Entering directory `/usr/local/src/asterisk/res'
gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE  -O6 -march=i686  -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\"CVS-06/18/03-11:58:58\" -DINSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"
-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\"
-DASTMODDIR=\"/usr/lib/asterisk/modules\"
-DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DBUSYDETECT
-DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC   -c -o res_indications.o
res_indications.c
res_indications.c: In function `handle_playtones':
res_indications.c:195: too few arguments to function `ast_playtones_start'
res_indications.c:197: too few arguments to function `ast_playtones_start'
res_indications.c: At top level:
res_indications.c:216: warning: function declaration isn't a prototype
make[1]: [res_indications.o] Error 1 (ignored)



make[1]: Entering directory `/usr/local/src/asterisk/apps'
gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE  -O6 -march=i686  -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\"CVS-06/18/03-11:58:58\" -DINSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"
-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\"
-DASTMODDIR=\"/usr/lib/asterisk/modules\"
-DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DBUSYDETECT -fPIC   -c
-o app_db.o app_db.c
In file included from app_db.c:16:
../include/asterisk/options.h:30: parse error before "ast_startuptime"
../include/asterisk/options.h:30: warning: type defaults to `int' in
declaration of `ast_startuptime'
../include/asterisk/options.h:30: warning: data definition has no type or
storage class
../include/asterisk/options.h:31: parse error before "ast_lastreloadtime"
../include/asterisk/options.h:31: warning: type defaults to `int' in
declaration of `ast_lastreloadtime'
../include/asterisk/options.h:31: warning: data definition has no type or
storage class
app_db.c: In function `deltree_exec':
app_db.c:98: warning: suggest explicit braces to avoid ambiguous `else'
make[1]: [app_db.o] Error 1 (ignored)

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[Asterisk-Users] SIP registration without password (secret)

[Cerrajetto]

Hello,

I'm trying to registrate a Nuance Server in Asterisk (using SIP) with no
success.

It seems that Nuance does not send any secret/password (there is no way to
define it!), this is the list of parameters that Nuance provides for
registration:

audio.sip.UserAgentURI=sip:[EMAIL PROTECTED]
audio.sip.UserAgentPort=
audio.sip.ProxyServerURI=sip::
audio.sip.LocationServerURI=sip::

Our values:

audio.sip.UserAgentURI=sip:[EMAIL PROTECTED]
audio.sip.UserAgentPort=5060
audio.sip.ProxyServerURI=sip:192.168.200.200 (ASTERISK SERVER)
audio.sip.LocationServerURI=sip:192.168.200.200 (ASTERISK SERVER)

In sip.conf:

[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls of not registered phones

;Voz
[541160]
type=friend
secret=
insecure=no
host=192.168.200.160
dtmfmode=inband


¿Is there a way to indicate to DO NOT USE any password in the registration
process?

Thank you,
Cerrajetto

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Re: [Asterisk-Users] databases for billing

[carlos del mayor]

can you be more explicit, please? or give me some examples? please, i'm little lost!
thanks a lot
carlosMartin Pycko <[EMAIL PROTECTED]> wrote:
cdr_mysql.confOn Fri, 20 Jun 2003, carlos del mayor wrote:> hi> I want to do a database to save the cdr with a billing finality. I've created the database in mysql (thanks for the table and all that!) but I'm not sure of how to 'connect' asterisk to that database in order to save there the cdr. Is the cdr_mysql.conf what I have to config? Or must I do a script, with the 'database' AGI commands?> Any help would be so apreciated!> Thanks a lot> carlos>>> -> Yahoo! Sorteos> Juega a la Lotería Primitiva sin salir de casa___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersYahoo!
 SorteosJuega a la Lotería Primitiva sin salir de casa

Re: [Asterisk-Users] Active ISDN PCMCIA card

[Tan Aks]

We use and sell the AVM B1 PCI V4.0 card. It seems to work well with
asterisk apart from slight echo that I noticed when receiving an isdn -->
* --> remote sip phone call.

Tan



- Original Message - 
From: "Michael Manousos" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, June 20, 2003 12:28 PM
Subject: [Asterisk-Users] Active ISDN PCMCIA card



Are there any suggestions for active ISDN CAPI PCMCIA cards
that are known to work with Asterisk?

Thanks,
Michael.



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[Asterisk-Users] where to get adsi phones in europe ?

[Thomas Haeger]

Hi all,

have anybody an idea where to get adsi phones in europe ?



Thanks,

Thomas.

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Re: [Asterisk-Users] Asterisk hogging CPU resources

[Steven Critchfield]

This blows my main idea of not having a timing source to keep asterisk
from entering a busy loop. 

Are you running the most current CVS? I know there had been a bug some
time back that caused every asterisk thread to open handles on
/dev/zap/timer repeatedly and at some point my system had run out of
file handles to give out and performance started sucking. A CVS upgrade
fixed that. Oddly enough too was that it only happened on 1 of my 3
asterisk machines.

On Fri, 2003-06-20 at 10:06, Derek Beaumont wrote:
> The interfaces I'm using are 2 X100Ps and a TDM400P
> 
> 
> 
> 
> What kind of interfaces are you using?
> 
> I'm using zap and IAX on my main asterisk server that deals in about
> 300-400 calls a day without the cpu load you are seeing.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Unable to find a path

[Tilghman Lesher]

On Thursday 19 June 2003 12:49 pm, Gerardo wrote:
> I just installed Asterisk 0.4.0 with all the default options, and
> the configuration samples it has. When I try to dial from an h323
> client (gnomemeeting) I get this message on the messages file:
>
> Jun 19 11:48:45 WARNING[15375]: File file.c, Line 410
> (ast_openstream): File demo-congrats does not exist in any format
> Jun 19 11:48:45 WARNING[15375]: File file.c, Line 553
> (ast_streamfile): Unable to open demo-congrats (format 8): Success
>
> Actually, demo-congrats.gsm is installed on
> /var/lib/asterisk/sounds.

You need to load format_gsm.so and codec_gsm.so

-Tilghman

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[Asterisk-Users] Manager interface, again

[Andy Powell]

Ok, 

is it me or do some of the commands just not work properly? I asked for mailboxstatus
and got:

Response: Success
Message: Mailbox Status
Mailbox: 1000
Waiting: 0

which is all well and good, except of course I have 2 messages waiting... which kinda 
means
it only works, if you have 0 messages... (using voicemail not voicemail2)

Andy



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[Asterisk-Users] Newbie questions.....

[Chris Bshaw]

Hi.

I have just successfully setup Asterisk with 2 Cisco 7940 phones (converted 
for SIP) and a SIP softphone on a W2K box.and it all seems to work very 
well.to those who wrote this software, it is really cool.

Anyway, I am new to this software, and I have a lot of questions which I am 
hoping someone on the mailing list might be able to answer for me.I am 
basically trying to get an idea of  how/what I can do with Asterisk that I 
am already doing with our existing phone system

Sorry about the length of the mailthe docs don't seem to cover some of 
the topics below.

Thanx in advance for any help.

Chris.



* We currently have a Cisco IP telephony system (using their 
CallManager).am I right in saying that Cisco phones using Skinny will 
not work with asterisk? Is it ever likely too?

* When we connect and power on a Cisco 79X0 phone for the first time, it 
automatically registers with the CallManager and is assigned a temporary 
number. We then do into the CallManager admin interface and assign it to its 
owner, give it its permanent number etc. Among the things which happen are 
the TFTP files for the phone (eg: SEP.cnf) get created as part 
of the automatic registration. When I converted the phones to SIP, I had to 
manually create config files for each phone (SIP.cnf 
etc.).is there any way I can have this happen automatically?

* We have an 8 port E1 card in a Cisco 6509 which takes our main phone trunk 
to the public network. Can we connect an Asterisk PBX server with an E1 card 
to this? If so, could we then connect the Asterisk PBX to the callmanager? 
(Perhaps with another extension range).and if so, how?

* .or could we connect Asterisk to the 6509 over IP and so make it part 
of the main phone system?

* We have a Nortel Meridian PBX on our other campus which is connected to 
our IP telephony system via an E1 link to a Cisco Vg200 voice H.323 
gateway...would there be any way to point asterisk at this gateway and 
make it part of our main phone system that way? again if so how?

_
Add photos to your messages with MSN 8. Get 2 months FREE*. 
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Re: [Asterisk-Users] Dual T400P, SMP, performance issues

[Jayson Vantuyl]

On Tue, Jun 17, 2003 at 10:09:53PM +0200, The Traveller wrote:
> this time.  My guess is that you only don't see it if the console
> screensaver has already come on while it happens.
To disable the console screen saver (since my units are all on a KVM,
this is a Good Thing (TM) ), run:

setvesablank off

This is part of the kbd package (for Debian and RedHat at least).

Jayson

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RE: [Asterisk-Users] SIP registration without password (secret)

[Richard Alexander]

You could try placing the password after the username in the URI:

sip:username:[EMAIL PROTECTED]



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cerrajetto
Sent: Friday, June 20, 2003 11:17 AM
To: Asterisk Users
Subject: [Asterisk-Users] SIP registration without password (secret)

Hello,

I'm trying to registrate a Nuance Server in Asterisk (using SIP) with no
success.

It seems that Nuance does not send any secret/password (there is no way
to
define it!), this is the list of parameters that Nuance provides for
registration:

audio.sip.UserAgentURI=sip:[EMAIL PROTECTED]
audio.sip.UserAgentPort=
audio.sip.ProxyServerURI=sip::
audio.sip.LocationServerURI=sip::

Our values:

audio.sip.UserAgentURI=sip:[EMAIL PROTECTED]
audio.sip.UserAgentPort=5060
audio.sip.ProxyServerURI=sip:192.168.200.200 (ASTERISK SERVER)
audio.sip.LocationServerURI=sip:192.168.200.200 (ASTERISK SERVER)

In sip.conf:

[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls of not registered phones

;Voz
[541160]
type=friend
secret=
insecure=no
host=192.168.200.160
dtmfmode=inband


¿Is there a way to indicate to DO NOT USE any password in the
registration
process?

Thank you,
Cerrajetto

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[Asterisk-Users] Asterisk hogging CPU resources

[Derek Beaumont]

The version of asterisk I am running is:
Asterisk CVS-06/05/03-09:41:32

I am not 100% sure of how to update Asterisk.
Do I just download using CVS, then run
make clean
make upgrade
?


---
This blows my main idea of not having a timing source to keep asterisk
from entering a busy loop. 

Are you running the most current CVS? I know there had been a bug some
time back that caused every asterisk thread to open handles on
/dev/zap/timer repeatedly and at some point my system had run out of
file handles to give out and performance started sucking. A CVS upgrade
fixed that. Oddly enough too was that it only happened on 1 of my 3
asterisk machines.

On Fri, 2003-06-20 at 10:06, Derek Beaumont wrote:
> The interfaces I'm using are 2 X100Ps and a TDM400P
> 
> 
> 
> 
> What kind of interfaces are you using?
> 
> I'm using zap and IAX on my main asterisk server that deals in about
> 300-400 calls a day without the cpu load you are seeing.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>



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Re: [Asterisk-Users] where to get adsi phones in europe ?

[Tan Aks]

We sell the CE approved versions of the PT390. Contact me offline and I'll
give you details.

Tan

- Original Message - 
From: "Thomas Haeger" <[EMAIL PROTECTED]>
To: "Asterisk User" <[EMAIL PROTECTED]>
Sent: Friday, June 20, 2003 4:33 PM
Subject: [Asterisk-Users] where to get adsi phones in europe ?


Hi all,

have anybody an idea where to get adsi phones in europe ?



Thanks,

Thomas.

***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow

FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email:  [EMAIL PROTECTED]
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Re: [Asterisk-Users] Dual T400P, SMP, performance issues

[Jayson Vantuyl]

On Fri, Jun 13, 2003 at 12:21:48PM -0300, Alberto Bertogli wrote:
> Even better, you might also want to try out splitting the interrupts among
> each processor, according to the card. Like binding interrupts from card0
> to cpu0 and from card1 to cpu1; but really don't remember if you can do
> such things in 2.4.

This can be done.  Look at /proc/irq/{number}/smp_affinity

This file contains a hexadecimal map of which processors to use.  To
lock down to the first processor, I believe you can:

echo 0001 > /proc/...

And the second:

echo 0002 > /proc/...

And the third:

echo 0004 > /proc/...

If you've got a handle on bitmasks, you get the idea.

Jayson

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Re: [Asterisk-Users] Asterisk hogging CPU resources

[Steven Critchfield]

>From your asterisk source directory issue these commands

make clean
cvs update -d
make upgrade


The clean makes sure there is less stuff in the way of CVS to try and
find newer copies. The -d of the cvs command will also download files
that you may not have in your current tree. and the make upgrade is like
make install except it doesn't overwrite your configs or sound files.

On Fri, 2003-06-20 at 11:30, Derek Beaumont wrote:
> The version of asterisk I am running is:
> Asterisk CVS-06/05/03-09:41:32
> 
> I am not 100% sure of how to update Asterisk.
> Do I just download using CVS, then run
> make clean
> make upgrade
> ?
> 
> 
> ---
> This blows my main idea of not having a timing source to keep asterisk
> from entering a busy loop. 
> 
> Are you running the most current CVS? I know there had been a bug some
> time back that caused every asterisk thread to open handles on
> /dev/zap/timer repeatedly and at some point my system had run out of
> file handles to give out and performance started sucking. A CVS upgrade
> fixed that. Oddly enough too was that it only happened on 1 of my 3
> asterisk machines.
> 
> On Fri, 2003-06-20 at 10:06, Derek Beaumont wrote:
> > The interfaces I'm using are 2 X100Ps and a TDM400P
> > 
> > 
> > 
> > 
> > What kind of interfaces are you using?
> > 
> > I'm using zap and IAX on my main asterisk server that deals in about
> > 300-400 calls a day without the cpu load you are seeing.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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AW: [Asterisk-Users] where to get adsi phones in europe ?

[Thomas Haeger]

Hello Tan,

thanks for your info, but how can i contact you ?

I don't see any contact information.

Regards,

Thomas.

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Tan Aks
Gesendet: Freitag, 20. Juni 2003 18:31
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] where to get adsi phones in europe ?


We sell the CE approved versions of the PT390. Contact me offline and I'll
give you details.

Tan

- Original Message -
From: "Thomas Haeger" <[EMAIL PROTECTED]>
To: "Asterisk User" <[EMAIL PROTECTED]>
Sent: Friday, June 20, 2003 4:33 PM
Subject: [Asterisk-Users] where to get adsi phones in europe ?


Hi all,

have anybody an idea where to get adsi phones in europe ?



Thanks,

Thomas.

***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow

FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email:  [EMAIL PROTECTED]
***

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[Asterisk-Users] Poor quality with FWD - codec selection issue?

[Iain Stevenson]

A colleague called me through my * system via FWD using SJPhone and the 
quality was distinctly poor - a lot of hum and delay.  Looking at the debug 
log the codec used was miscellaneously numbered 0, 4 and 8.  I thought I'd 
disabled 4 (g.723) but it appears not.  My sip.conf has this:

general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0  ; Address to bind to
context = voip-sip
defaultexpiry = 3600
register => 12345:[EMAIL PROTECTED]/39
disallow=all
allow=alaw
allow=ulaw
I was expecting this would stop g.723 from being even tried - am I missing 
something?

Is there any config option for SJphone that clobbers g.723?

 Iain
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Re: [Asterisk-Users] Question :: groundstart and loopstart :: Update

[Surfer Dude]

Thanks for the responses.
 
Here is an update on my zap groundstart, loopstart, 
"disconnect supervision" woes.  Apparently, the groundstart mode in 
the CAC FXO module works on loopstart lines.  I still don't understand how 
and why.
 
I have loopstart lines.
 
The only way "disconnect supervision" works on my system is if 
I set the FXO module to groundstart.  However!  There is a big 
caveat.  In groundstart mode, incoming caller ID stops working!  This 
is something I am going to have to live with for now as disconnecting the line 
on hang-up is far more important than caller ID.
 
I have no idea if the caller ID problem is a problem with 
CAC FXO modules or the zap channel software or the fact that we are using 
groundstart signaling on a loopstart line.
 
I can confirm that this is the behavior with both CAC I and 
CAC II units as I have a CAC II unit and I spoke with Michael on this list who 
has a CAC I unit.  Both of us are in the same predicament.
 
Anyway, I hope others find this info useful.  If anyone 
has any other ideas on how I can get my caller ID to work I would be really 
excited about that.
 
 
Thanks,
Jason
 
PS: I have seen people mention that, when you have many lines, 
PRI is cheaper.  This is not our case at all.  San Francisco.  We 
have eight lines at the cost of $0.01 per line.  All we pay for really is 
the call time.
 

  - Original Message - 
  From: 
  Surfer Dude 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Tuesday, June 17, 2003 4:05 
PM
  Subject: [Asterisk-Users] Question :: 
  groundstart and loopstart
  
  Hello Astrites,
   
  I was just about to send out a long email about not being 
  able to detect hang-up with my CAC II FXO module on my PSTN POTS lines.  
  I had tried many configurations.  I had attached a toner to the line to 
  see if I was getting this "disconnect supervision" signal after hang-up.  
  (toner has a line powered light in the off position)  It seemed that I 
  was getting the signal, as:
   
      The light was steady on 
  onhook
      When the line rang the light 
  dimmed.
      When I picked up the line stayed dim (If 
  I remember correctly)
      When the other end hung up the light 
  stayed on
      Then turned off momentarily a few seconds 
  later.  (I suspected this was "disconnect supervision")
   
  Anyway asterisk never detected hang-up.  Everything 
  else worked.  I was frustrated as hell.
   
  After a few days of trying everything, I tried to set my CAC 
  II FXO module to Groundstart and configured zaptel.conf and zapata.conf to 
  groundstart signaling.
   
  Voila!  Everything work!
   
  The question is how does groundstart and loopstart 
  work?  How was I able to have a somewhat working system using loopstart 
  when it seems that the phone company signaling was groundstart?  Can I 
  assume that my lines are indeed groundstart?
   
  I am really happy that it works.  I can now begin the 
  process of deployment with a much greater feeling of security.
   
  Jason
   
   

Re: [Asterisk-Users] Asterisk hogging CPU resources

[John Congdon]

I just had this problem and Marin found it to be the fact that I was
not running the recommended version of mpg123
Try starting there

John

On Friday, June 20, 2003, at 12:03  PM, Steven Critchfield wrote:

This blows my main idea of not having a timing source to keep asterisk
from entering a busy loop.
Are you running the most current CVS? I know there had been a bug some
time back that caused every asterisk thread to open handles on
/dev/zap/timer repeatedly and at some point my system had run out of
file handles to give out and performance started sucking. A CVS upgrade
fixed that. Oddly enough too was that it only happened on 1 of my 3
asterisk machines.
On Fri, 2003-06-20 at 10:06, Derek Beaumont wrote:
The interfaces I'm using are 2 X100Ps and a TDM400P




What kind of interfaces are you using?
I'm using zap and IAX on my main asterisk server that deals in about
300-400 calls a day without the cpu load you are seeing.
--
Steven Critchfield  <[EMAIL PROTECTED]>
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Re: [Asterisk-Users] Asterisk hogging CPU resources

[Surfer Dude]

I will just put my few cense in here...  I would keep an eye on what you are
updating however.  For instance, in sh, or bash:

cvs -n update -dA 2>&1 |grep -v "^cvs"

will show you, in a nice format which files would be updated were you to
update the files.  After you get that list you may want to check on each
file.

cvs st 

show you which version of the file you have locally and what the version on
the server is.  This way you can see how many changes have taken place on
the file since your last update.

cvs log 

Will give you the comments that the developer added to each version of the
file.  These are sometimes very important to know.  It could help you decide
whether or not you want to actually update at this time.

cvs diff -rHEAD 

will show you the actually difference in the file between the version you
have and the very latest version in the repository.  You can use this
command to watch the code that is being checked in.  You can see how bugs
are fixed.  And you may spot a problem with the code that was checked in.

All these commands can help you get a feel of what is actually going on.  If
you do an update and then problems occur you may be more equipped to find
out why the problems are occurring and what you can do to get around them.

This has been one of my methods copping on the tip of any codeline.

Jason




- Original Message - 
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, June 20, 2003 9:45 AM
Subject: Re: [Asterisk-Users] Asterisk hogging CPU resources


> >From your asterisk source directory issue these commands
>
> make clean
> cvs update -d
> make upgrade
>
>
> The clean makes sure there is less stuff in the way of CVS to try and
> find newer copies. The -d of the cvs command will also download files
> that you may not have in your current tree. and the make upgrade is like
> make install except it doesn't overwrite your configs or sound files.
>
> On Fri, 2003-06-20 at 11:30, Derek Beaumont wrote:
> > The version of asterisk I am running is:
> > Asterisk CVS-06/05/03-09:41:32
> >
> > I am not 100% sure of how to update Asterisk.
> > Do I just download using CVS, then run
> > make clean
> > make upgrade
> > ?
> >
> >
> > ---
> > This blows my main idea of not having a timing source to keep asterisk
> > from entering a busy loop.
> >
> > Are you running the most current CVS? I know there had been a bug some
> > time back that caused every asterisk thread to open handles on
> > /dev/zap/timer repeatedly and at some point my system had run out of
> > file handles to give out and performance started sucking. A CVS upgrade
> > fixed that. Oddly enough too was that it only happened on 1 of my 3
> > asterisk machines.
> >
> > On Fri, 2003-06-20 at 10:06, Derek Beaumont wrote:
> > > The interfaces I'm using are 2 X100Ps and a TDM400P
> > >
> > >
> > >
> > > 
> > > What kind of interfaces are you using?
> > >
> > > I'm using zap and IAX on my main asterisk server that deals in about
> > > 300-400 calls a day without the cpu load you are seeing.
> -- 
> Steven Critchfield  <[EMAIL PROTECTED]>
>
> ___
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>
>

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Re: [Asterisk-Users] Manager interface, again

[Andy Powell]

Ok, 

like a  I'll answer my own message:

If in your voicemail.conf you have * configured to the send message in an email you 
will NOT get a stutter dialtone or any MWI light you may have. I've just removed my 
email address from voicemail.conf.. much better like that... 

HTHSITF

Andy



*** REPLY SEPARATOR  ***

On 20/06/2003 at 18:17 Andy Powell wrote:

>Ok, 
>
>is it me or do some of the commands just not work properly? I asked for
>mailboxstatus
>and got:
>
>Response: Success
>Message: Mailbox Status
>Mailbox: 1000
>Waiting: 0
>
>which is all well and good, except of course I have 2 messages waiting...
>which kinda means
>it only works, if you have 0 messages... (using voicemail not voicemail2)
>
>Andy
>
>
>
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Re: [Asterisk-Users] Question :: groundstart and loopstart :: Update

[TC]

Callerid issue
1) if you run ztmonitor on 
the fxo line & call in do you hear the fsk tone
 if yes then we beleive the CAC is passing 
fsk
    2) in chan_zap->ss_thread around line 
4154 (current cvs)
    if you get to the 
callerid_feed at least once then
  if you 
get to chan_zap->ss_thread->callerid_get  around line 4163 (current 
cvs)
    does 
this parse fail
  
else
    do you 
get check sumfails or sum fin else
    else
  zaptel 
is not detecting the ring/fsk correct
  might 
need to tunimng on ZT_RINGTRAILER
 else
    CAC is hooped

-Original Message-From: 
Surfer Dude <[EMAIL PROTECTED]>To: [EMAIL PROTECTED] 
<[EMAIL PROTECTED]>Date: 
June 20, 2003 10:24 AMSubject: Re: [Asterisk-Users] Question 
:: groundstart and loopstart :: Update
Thanks for the responses.
 
Here is an update on my zap groundstart, loopstart, 
"disconnect supervision" woes.  Apparently, the 
groundstart mode in the CAC FXO module works on loopstart lines.  I 
still don't understand how and why.
 
I have loopstart lines.
 
The only way "disconnect supervision" works on 
my system is if I set the FXO module to groundstart.  However!  
There is a big caveat.  In groundstart mode, incoming caller ID stops 
working!  This is something I am going to have to live with for now as 
disconnecting the line on hang-up is far more important than caller 
ID.
 
I have no idea if the caller ID problem is a problem 
with CAC FXO modules or the zap channel software or the fact that we are 
using groundstart signaling on a loopstart line.
 
I can confirm that this is the behavior with both CAC I 
and CAC II units as I have a CAC II unit and I spoke with Michael on this 
list who has a CAC I unit.  Both of us are in the same 
predicament.
 
Anyway, I hope others find this info useful.  If 
anyone has any other ideas on how I can get my caller ID to work I would be 
really excited about that.
 
 
Thanks,
Jason
 
PS: I have seen people mention that, when you have many 
lines, PRI is cheaper.  This is not our case at all.  San 
Francisco.  We have eight lines at the cost of $0.01 per line.  
All we pay for really is the call time.
 

- Original Message - 
From: 
Surfer 
Dude 
To: [EMAIL PROTECTED] 

Sent: Tuesday, June 17, 2003 4:05 
PM
Subject: [Asterisk-Users] Question 
:: groundstart and loopstart

Hello Astrites,
 
I was just about to send out a long email about not 
being able to detect hang-up with my CAC II FXO module on my PSTN POTS 
lines.  I had tried many configurations.  I had attached a 
toner to the line to see if I was getting this "disconnect 
supervision" signal after hang-up.  (toner has a line powered 
light in the off position)  It seemed that I was getting the 
signal, as:
 
    The light was steady on 
onhook
    When the line rang the light 
dimmed.
    When I picked up the line stayed 
dim (If I remember correctly)
    When the other end hung up the 
light stayed on
    Then turned off momentarily a few 
seconds later.  (I suspected this was "disconnect 
supervision")
 
Anyway asterisk never detected hang-up.  
Everything else worked.  I was frustrated as hell.
 
After a few days of trying everything, I tried to set 
my CAC II FXO module to Groundstart and configured zaptel.conf and 
zapata.conf to groundstart signaling.
 
Voila!  Everything work!
 
The question is how does groundstart and loopstart 
work?  How was I able to have a somewhat working system using 
loopstart when it seems that the phone company signaling was 
groundstart?  Can I assume that my lines are indeed 
groundstart?
 
I am really happy that it works.  I can now begin 
the process of deployment with a much greater feeling of 
security.
 
Jason
 
 

Re: [Asterisk-Users] Active ISDN PCMCIA card

[Olaf Menzel]

On Friday 20 June 2003 13:28, Michael Manousos wrote:
> Are there any suggestions for active ISDN CAPI PCMCIA cards
> that are known to work with Asterisk?
>

You can try AVM B1 PCMCIA. This card is fully I4L compliant but AVM has 
developed a LINUX capi 2.0 stack. 
http://www.avm.de/en/products/hardware/active/B1_PCMCIA/index.html
The Linux Capi driver you find here:
ftp://ftp.avm.de/cardware/b1_pcm/linux/
Be aware that the Capi4Linux driver is distributed only as binary and 
especially prepared for Suse distributions. WIth some adaptations it should 
work with other distributions as well. Otherwise you should use I4L for this 
card. BTW. The Capi4Linux driver works also for the AVM Fritz which is much 
cheaper than the B1 device and supports full CAPI functionality such as G3 
Fax.

regards

Olaf
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[Asterisk-Users] Asterisk hogging CPU resources

[Derek Beaumont]

What is the recommended version of mpg123?
I am running 0.59r


I just had this problem and Marin found it to be the fact that I was
not running the recommended version of mpg123

Try starting there

John

On Friday, June 20, 2003, at 12:03  PM, Steven Critchfield wrote:

> This blows my main idea of not having a timing source to keep asterisk
> from entering a busy loop.
>
> Are you running the most current CVS? I know there had been a bug some
> time back that caused every asterisk thread to open handles on
> /dev/zap/timer repeatedly and at some point my system had run out of
> file handles to give out and performance started sucking. A CVS
upgrade
> fixed that. Oddly enough too was that it only happened on 1 of my 3
> asterisk machines.
>
> On Fri, 2003-06-20 at 10:06, Derek Beaumont wrote:
>> The interfaces I'm using are 2 X100Ps and a TDM400P
>>
>>
>>
>> 
>> What kind of interfaces are you using?
>>
>> I'm using zap and IAX on my main asterisk server that deals in about
>> 300-400 calls a day without the cpu load you are seeing.
> -- 
> Steven Critchfield  <[EMAIL PROTECTED]>
>
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Re: [Asterisk-Users] Manager interface, again

[Steven Critchfield]

On Fri, 2003-06-20 at 12:36, Andy Powell wrote:
> Ok, 
> 
> like a  I'll answer my own message:
> 
> If in your voicemail.conf you have * configured to the send message in
> an email you will NOT get a stutter dialtone or any MWI light you may
> have. I've just removed my email address from voicemail.conf.. much
> better like that... 

This is odd because I email all my users voicemail out and the ones that
don't clear the voicemail on the phone still get stutter tones. I had to
inform them of what to do, and then mass delete their voicemail to get
the stutter tone to stop. One user had almost 50 messages waiting.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Manager interface, again

[Andy Powell]

>This is odd because I email all my users voicemail out and the ones that
>don't clear the voicemail on the phone still get stutter tones. I had to
>inform them of what to do, and then mass delete their voicemail to get
>the stutter tone to stop. One user had almost 50 messages waiting.

I had this initially, but it was due to a 'zombie' message (as pointed out to me
by citats... an easy way to double check is to connect to the manager interface 
and look at the status of a mailbox.Iif it reports something like

mailbox: 1000
Response: Success
Message: Mailbox Status
Mailbox: 1000
Waiting: 1 <--- that is NOT a msg count btw


then the user would get a stutter tone.. if Waiting: 0 then they wont...

I'd be interested to hear if this is/isn;t the case on your setup

HTH

Andy



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RE: [Asterisk-Users] Manager interface, again

[Wade Weppler]

Same here.  E-mail and MWI/Stutter tone work fine together.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steven Critchfield
> Sent: Friday, June 20, 2003 2:14 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Manager interface, again
> 
> On Fri, 2003-06-20 at 12:36, Andy Powell wrote:
> > Ok,
> >
> > like a  I'll answer my own message:
> >
> > If in your voicemail.conf you have * configured to the send message in
> > an email you will NOT get a stutter dialtone or any MWI light you may
> > have. I've just removed my email address from voicemail.conf.. much
> > better like that...
> 
> This is odd because I email all my users voicemail out and the ones that
> don't clear the voicemail on the phone still get stutter tones. I had to
> inform them of what to do, and then mass delete their voicemail to get
> the stutter tone to stop. One user had almost 50 messages waiting.
> --
> Steven Critchfield  <[EMAIL PROTECTED]>
> 
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Re: [Asterisk-Users] SIP registration without password (secret)

[John Todd]

Remove the "secret=" lines for SIP peers that do not have passwords.

Here is an example of a host that sends us calls but no password:

[foo1]
host=192.168.200.160
type=friend
dtmfmode=inband
That's it; very simple.  If you discover that SIP messages seem to be 
"ignored" in one direction, se the "insecure=yes" feature.  This will 
allow SIP inbound messages with return ports other than 5060 (which 
some SIP severs brokenly use.)

JT



Hello,

I'm trying to registrate a Nuance Server in Asterisk (using SIP) with no
success.
It seems that Nuance does not send any secret/password (there is no way to
define it!), this is the list of parameters that Nuance provides for
registration:
audio.sip.UserAgentURI=sip:[EMAIL PROTECTED]
audio.sip.UserAgentPort=
audio.sip.ProxyServerURI=sip::
audio.sip.LocationServerURI=sip::
Our values:

audio.sip.UserAgentURI=sip:[EMAIL PROTECTED]
audio.sip.UserAgentPort=5060
audio.sip.ProxyServerURI=sip:192.168.200.200 (ASTERISK SERVER)
audio.sip.LocationServerURI=sip:192.168.200.200 (ASTERISK SERVER)
In sip.conf:

[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls of not registered phones
;Voz
[541160]
type=friend
secret=
insecure=no
host=192.168.200.160
dtmfmode=inband
¿Is there a way to indicate to DO NOT USE any password in the registration
process?
Thank you,
Cerrajetto
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Re: [Asterisk-Users] Newbie questions.....

[John Todd]

Hi.

I have just successfully setup Asterisk with 2 Cisco 7940 phones 
(converted for SIP) and a SIP softphone on a W2K box.and it all 
seems to work very well.to those who wrote this software, it is 
really cool.

Anyway, I am new to this software, and I have a lot of questions 
which I am hoping someone on the mailing list might be able to 
answer for me.I am basically trying to get an idea of  how/what 
I can do with Asterisk that I am already doing with our existing 
phone system

Sorry about the length of the mailthe docs don't seem to cover 
some of the topics below.

Thanx in advance for any help.

Chris.

* We currently have a Cisco IP telephony system (using their 
CallManager).am I right in saying that Cisco phones using Skinny 
will not work with asterisk? Is it ever likely too?
Skinny is not included as a channel in Asterisk at this time.

There are reports of a Skinny channel well into development - see 
http://www.sf.net/projects/sccp  and we await further testing.

* When we connect and power on a Cisco 79X0 phone for the first 
time, it automatically registers with the CallManager and is 
assigned a temporary number. We then do into the CallManager admin 
interface and assign it to its owner, give it its permanent number 
etc. Among the things which happen are the TFTP files for the phone 
(eg: SEP.cnf) get created as part of the automatic 
registration. When I converted the phones to SIP, I had to manually 
create config files for each phone (SIP.cnf 
etc.).is there any way I can have this happen automatically?
Yes and no.  You still will have to create a file called SIP.cnf which contains the "extensions" that you expect the 
phone to use.  However, if you have an RFC compliant DHCP server, you 
should be able to make everything happen automatically except for the 
generation of that extension.  There are almost no hooks between any 
of the very sophisticated Cisco configuration files and Asterisk; 
they are _separate_ systems.  Asterisk simply deals with SIP devices 
and their SIP transactions - Asterisk does _not_ configure SIP 
devices, and Asterisk is not Cisco-specific in any treatment of SIP 
transactions.

I seem to recall that there is a Cisco 79xx administration tool in 
the http://www.vovida.org/ pages somewhere.

* We have an 8 port E1 card in a Cisco 6509 which takes our main 
phone trunk to the public network. Can we connect an Asterisk PBX 
server with an E1 card to this? If so, could we then connect the 
Asterisk PBX to the callmanager? (Perhaps with another extension 
range).and if so, how?
Yes.  The manual should explain further details.


* .or could we connect Asterisk to the 6509 over IP and so make 
it part of the main phone system?
I don't know.  Does the 6509 talk SIP?

* We have a Nortel Meridian PBX on our other campus which is 
connected to our IP telephony system via an E1 link to a Cisco Vg200 
voice H.323 gateway...would there be any way to point asterisk 
at this gateway and make it part of our main phone system that way? 
again if so how?
Yes.  That's too complex to explain adequately here, but you should 
try setting it up to answer the question yourself.

JT

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thanks!, was Re: [Asterisk-Users] newbie needs SIP config examples -- especially soft phones

[Reed Wade]

thanks to everyone for your gracious assistance; it stills wants
plenty of minor adjustments but I now have the core of a nicely working
system
-reed

At 11:56 PM 6/17/2003 -0500, John Laur wrote:
> So far, I've only been able to get the XTEN Lite phone working
> and I really don't understand how I set it up. I used "xten"
> for every option everywhere (display name, username, password,
> and Domain/Realm) and the corresponding section in sip.conf.
> I've had no luck getting the SJ Labs soft phone to connect using
> a similar blunderbuss method.
[youruser] ;username here and also below...
type=friend;dial both to and from
username=youruser  ;same thing as in brackets above
password=password  ;password obviously
context=default;or put whatever you want - this is the sip realm too
mailbox=1234   ;for message waiting
host=dynamic   ;might be coming from different ip's
callerid="Soft Phone" <1234>
nat=yes;might be behind a nat
> I'm wondering if someone could point me to SIP configuration
> examples or education so I can understand what I'm doing. I'm
> finding the client configuration more confusing that the *
> configs.
Your client will want an auth name or two (use the username for these), a
secret or password (the password), a port number (5060 is the default and
you can change it in the [general] section of sip.conf), maybe a realm
(the context though it is not important for authentication), a sip proxy
address - your asterisk server's ip address, and that should be it. Most
have an option you have to turn on to tell the client to actually register
with the proxy. turn that on and check to see that your client is
connected with 'show sip peers' on the asterisk console. It might also be
helpful to turn on 'sip debug' to see if your client is trying to
register. If you got the x-lite working the others should be easy too..
You'll see..
> An example of password protected SIP phone access would also be
> very helpful.
see above.

> I need to be able to support folks working from home connecting
> through the net as well inside the office. I expect NAT to be
> a pain.
NAT is not so hard once you get it going. First: make sure your asterisk
server has a public IP address and the ONLY default gateway on the machine
is set to the router for the public ip. Make sure you have set nat=yes in
the corresponding sip.conf entry for the device you're setting up, then
start poking at your client for the settings that say "I'm behind a NAT"
-- they are designed to make sure the packets source at the same UDP ports
they need to come back to so that the NAT's will open up a pathway back to
the internal device. Some clients do this by default anyway -- On the
X-Lite phone you don't really have to do much of anything -- maybe uncheck
the box that says "Send Internal IP" though I have found that it doesnt
really matter if nat=yes on the asterisk box. On the cisco 7960 phones,
the following settings work:
nat_enable: 1
nat_address; ""
voip_control_port: 5060
start_media_port: 16384 ; You can reduce this port range if you
end_media_port: 32766   ; have a picky firewall
nat_received_processing: 1  ; Makes phone re-register if your ip changes
Hope this helps you some...

John

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RE: [Asterisk-Users] Newbie questions.....

[tim.mcqueen]

>>* .or could we connect Asterisk to the 6509 over IP and so make 
>>it part of the main phone system?

>I don't know.  Does the 6509 talk SIP?

It doesn't appear to.  I would love to be wrong.  It does support MGCP,
though.

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RE: [Asterisk-Users] Manager interface, again

[Andy Powell]

On 20/06/2003 at 14:45 Wade Weppler wrote:

>Same here.  E-mail and MWI/Stutter tone work fine together.
>

if that attaching the file or just sending a messages without a file attached..?

Andy



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RE: [Asterisk-Users] Manager interface, again

[Steven Critchfield]

On Fri, 2003-06-20 at 15:20, Andy Powell wrote:
> On 20/06/2003 at 14:45 Wade Weppler wrote:
> 
> >Same here.  E-mail and MWI/Stutter tone work fine together.
> >
> 
> if that attaching the file or just sending a messages without a file attached..?

We attach the voicemail, thats how 2 of my users eneded up with 50 or so
messages waiting. They didn't feel the need to listen to or delete the
messages via the phone when they had listened to them via the email
interface.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] Manager interface, again

[Andy Powell]

>> if that attaching the file or just sending a messages without a file
>attached..?
>
>We attach the voicemail, thats how 2 of my users eneded up with 50 or so
>messages waiting. They didn't feel the need to listen to or delete the
>messages via the phone when they had listened to them via the email
>interface.
>-- 
>Steven Critchfield  <[EMAIL PROTECTED]>
>


Mmmm, that's odd... as soon as I add the email address I stop getting
the stutter tone and MWI...

Andy


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Re: [Asterisk-Users] Asterisk hogging CPU resources

[Martin Pycko]

Well if you have lots of /dev/timer opened than you have to edit your
asterisk/Makefile and comment out ZAPTEL_OPTIMIZATIONS or something like
that.

Martin

On Fri, 20 Jun 2003, Derek Beaumont wrote:

> What is the recommended version of mpg123?
> I am running 0.59r
>
> 
> I just had this problem and Marin found it to be the fact that I was
> not running the recommended version of mpg123
>
> Try starting there
>
> John
>
> On Friday, June 20, 2003, at 12:03  PM, Steven Critchfield wrote:
>
> > This blows my main idea of not having a timing source to keep asterisk
> > from entering a busy loop.
> >
> > Are you running the most current CVS? I know there had been a bug some
> > time back that caused every asterisk thread to open handles on
> > /dev/zap/timer repeatedly and at some point my system had run out of
> > file handles to give out and performance started sucking. A CVS
> upgrade
> > fixed that. Oddly enough too was that it only happened on 1 of my 3
> > asterisk machines.
> >
> > On Fri, 2003-06-20 at 10:06, Derek Beaumont wrote:
> >> The interfaces I'm using are 2 X100Ps and a TDM400P
> >>
> >>
> >>
> >> 
> >> What kind of interfaces are you using?
> >>
> >> I'm using zap and IAX on my main asterisk server that deals in about
> >> 300-400 calls a day without the cpu load you are seeing.
> > --
> > Steven Critchfield  <[EMAIL PROTECTED]>
> >
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[Asterisk-Users] Can Zaptel cards be used with other Linux apps?

[Dylan VanHerpen]

Can Zaptel T1/E1 cards also be used with other Linux apps (for instance 
as the WAN interface on a Linux based router, or with Bayonne, Vocal)?

Thks, Dylan.

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Re: [Asterisk-Users] Can Zaptel cards be used with other Linuxapps?

[Steven Critchfield]

On Fri, 2003-06-20 at 16:10, Dylan VanHerpen wrote:
> Can Zaptel T1/E1 cards also be used with other Linux apps (for instance 
> as the WAN interface on a Linux based router, or with Bayonne, Vocal)?

As a router, absolutely. As a interface for other voice products, well
you will probably have to do some programming there, but there is now
technical reason stopping it from happening. Just beware the licenses.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] 4 channel TDM40B only has 1 channel now

[Anthony Minessale]

Did I miss a new way configure this card cos after 
I made the newest snapshot only 1 of the 4 interfaces 
shows up to asterisk.
 
 
 
Do you Yahoo!?
SBC Yahoo! DSL - Now only $29.95 per month!

RE: [Asterisk-Users] Can Zaptel cards be used with other Linux apps?

[Wade Weppler]

Linux based router?  Yes.

Bayonne and Vocal?  I don't think so, at least not for voice.

-wade


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dylan VanHerpen
> Sent: Friday, June 20, 2003 5:10 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Can Zaptel cards be used with other Linux apps?
> 
> Can Zaptel T1/E1 cards also be used with other Linux apps (for instance
> as the WAN interface on a Linux based router, or with Bayonne, Vocal)?
> 
> Thks, Dylan.
> 
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RE: [Asterisk-Users] Newbie questions.....

[Chris Bshaw]

Thanx for the infounfortunately, I think we would need an Communications 
Media Modulewhich we don't have

Chris.



From: <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Newbie questions.
Date: Fri, 20 Jun 2003 15:21:53 -0500
>>* .or could we connect Asterisk to the 6509 over IP and so make
>>it part of the main phone system?
>I don't know.  Does the 6509 talk SIP?

It doesn't appear to.  I would love to be wrong.  It does support MGCP,
though.
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Re: [Asterisk-Users] Newbie questions.....

[Chris Bshaw]

Hi

Thanx for the info.sorry to hassle you, but I have follow on questions 
below.

I seem to recall that there is a Cisco 79xx administration tool in the 
http://www.vovida.org/ pages somewhere.
Had a look at this.this tool will certainly make managing Cisco SIP 
phones easierthanx

Yes.  The manual should explain further details.
I am not a voice comms expert and our Cisco IP Telephony system was 
installed by an outside company

Are you referring to the Asterisk manual here?

Just looking at the manual, it seems to me that I could use a Zaptel E1 card 
and configure the zapata.conf file appropriately..am I correct?

If so, I would just need to learn how to configure an E1 port on the 
6509.and configure the CallManager to know where Asterisk is in the 
number range

* We have a Nortel Meridian PBX on our other campus which is connected to 
our IP telephony system via an E1 link to a Cisco Vg200 voice H.323 
gateway...would there be any way to point asterisk at this gateway and 
make it part of our main phone system that way? again if so how?
Yes.  That's too complex to explain adequately here, but you should try 
setting it up to answer the question yourself.
Again, I am at a bit of a loss since I am not a voice comms expertwhere 
would I begin in Asterisk...is there a H.323 channel (as there is for SIP) 
in Asterisk?...or is this a silly question.?

I don't see any mention of H.323 in the conf files

and lastly, one further question.I got voicemail working, but the 
red light on the Cisco 79X0 phone doesn't light when a voice mail is 
waiting.is there a way to enable this?

Thanx  very much for the info, and thanx in advance for any further info

Chris.

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Re: [Asterisk-Users] Newbie questions.....

[John Todd]

[snip]

Yes.  The manual should explain further details.
I am not a voice comms expert and our Cisco IP Telephony system was 
installed by an outside company

Are you referring to the Asterisk manual here?
Yes.

Just looking at the manual, it seems to me that I could use a Zaptel 
E1 card and configure the zapata.conf file appropriately..am I 
correct?
Yes, you could connect your Cisco to an Asterisk server via an E1 PRI.

If so, I would just need to learn how to configure an E1 port on the 
6509.and configure the CallManager to know where Asterisk is in 
the number range
That is correct.

* We have a Nortel Meridian PBX on our other campus which is 
connected to our IP telephony system via an E1 link to a Cisco 
Vg200 voice H.323 gateway...would there be any way to point 
asterisk at this gateway and make it part of our main phone system 
that way? again if so how?
Yes.  That's too complex to explain adequately here, but you should 
try setting it up to answer the question yourself.
Again, I am at a bit of a loss since I am not a voice comms 
expertwhere would I begin in Asterisk...is there a H.323 channel 
(as there is for SIP) in Asterisk?...or is this a silly 
question.?
Yes, there is an H.323 channel for Asterisk.  Two, in fact.  See the 
list archives for details.

I don't see any mention of H.323 in the conf files

and lastly, one further question.I got voicemail working, 
but the red light on the Cisco 79X0 phone doesn't light when a voice 
mail is waiting.is there a way to enable this?
Yes.  See "mailbox=" in sip.conf.  You're going to need to be much 
more specific if you want more assistance.

Thanx  very much for the info, and thanx in advance for any further info

Chris.

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[Asterisk-Users] Specifying Allowed Codecs in iax.conf

[Eric Wieling]

What's the proper way to specify the allowed codecs in iax.conf?  It
doens't like allow=ilbc,gsm but if I put two allow= lines, one for ilbc
and one for gsm it seems to always to want to use gsm.

--Eric
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[Asterisk-Users] Moving up... T1 with 50 extensions

[Mark Street]

OK, I think I'm ready for the next step from the X100P.  Splitting a T1 with a 
T100P and a channel bank/s ~ 50 extensions. 

I think the existing system uses NEC phones for ~34 extensions, I don't know 
the specific model yet of the phones or the channel bank.

If the channel bank is not compatible with Asterisk/digium what brand should I 
be looking at to interface best with Asterisk and digium cards?

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Re: [Asterisk-Users] Moving up... T1 with 50 extensions

[Steven Critchfield]

On Fri, 2003-06-20 at 20:02, Mark Street wrote:
> OK, I think I'm ready for the next step from the X100P.  Splitting a T1 with a 
> T100P and a channel bank/s ~ 50 extensions. 

T1 = 24 channels therefore to get 48 extensions you will need 2 T1s and
2 channel banks. You might want to look into the T400P so you will have
extra ports left over for your incoming lines. Remember 3 T100Ps are
equal in cost as 1 T400P, and the fewer cards you have in the machine
means fewer interupts and less trouble configuring.

> I think the existing system uses NEC phones for ~34 extensions, I don't know 
> the specific model yet of the phones or the channel bank.
> 
> If the channel bank is not compatible with Asterisk/digium what brand should I 
> be looking at to interface best with Asterisk and digium cards?

The first thing to see, are these actually analog phones? can you put
them on pots lines? If not, they aren't compatible. 

-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] best ISDN BRI solution for DID

[Chad Sawyer]

I want to test asterisk with its DID functions on a 
smaller scale without the expense of a PRI.  I am in the US, and want the 
most reliable/stable, and feature rich solution that I can have.
 
What ISDN card would be reccommended?  

 
My clec switch operator says he can/will send DID 
info to my BRI's if I wish.  Can Asterisk, or one of its addons support 
this?
 
What BRI solution would be reccommended?  
ISDN4Linux, chan_capi?  Any info is appreciated, I hope to get this right 
the first time without having to buy more hardware after the initial 
purchase.
 
Any advice from you guys with experience would be 
appreciated.
 
Chad

[Asterisk-Users] More than one param to AGI

[Eric Wieling]

I'm starting to write an AGI script.  I want to pass more than one
parameter to the script, but seem to be unable to.

extensions.conf:
   exten => 85,1,AGI(/etc/asterisk/agi/args.agi,myarg1,myarg2)

args.agi:
   #!/usr/bin/perl
   print STDERR "FNORD prog = $0\n";
   print STDERR "FNORD arg 1 = $ARGV[0]\n";
   print STDERR "FNORD arg 2 = $ARGV[1]\n";
   print STDERR "FNORD arg 3 = $ARGV[2]\n";
   print STDERR "FNORD arg 4 = $ARGV[3]\n";

What I get at the console:
   FNORD prog = /etc/asterisk/agi/args.agi
   FNORD arg 1 = myarg1
   FNORD arg 2 =
   FNORD arg 3 =
   FNORD arg 4 =

Is this just a limitation of AGI or am I horribly confuzzled?

--Eric

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Re: [Asterisk-Users] More than one param to AGI

[Steven Critchfield]

Just a timtowtdi comment, use setvar and get the args from that. You
will get the added benifit of having the args named and not be position
sensitive.

As of late I have been very happy with the idea of passing hashes in
perl to get away from postitional variables. 

I'm not sure if the call to agi will split the variables for you, or if
you have to do it once it is inside the app. Either way, the option
above should do you fine. 

On Sat, 2003-06-21 at 00:04, Eric Wieling wrote:
> I'm starting to write an AGI script.  I want to pass more than one
> parameter to the script, but seem to be unable to.
> 
> extensions.conf:
>exten => 85,1,AGI(/etc/asterisk/agi/args.agi,myarg1,myarg2)
> 
> args.agi:
>#!/usr/bin/perl
>print STDERR "FNORD prog = $0\n";
>print STDERR "FNORD arg 1 = $ARGV[0]\n";
>print STDERR "FNORD arg 2 = $ARGV[1]\n";
>print STDERR "FNORD arg 3 = $ARGV[2]\n";
>print STDERR "FNORD arg 4 = $ARGV[3]\n";
> 
> What I get at the console:
>FNORD prog = /etc/asterisk/agi/args.agi
>FNORD arg 1 = myarg1
>FNORD arg 2 =
>FNORD arg 3 =
>FNORD arg 4 =
> 
> Is this just a limitation of AGI or am I horribly confuzzled?
> 
> --Eric
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Re: [Asterisk-Users] best ISDN BRI solution for DID

[WipeOut .]

Just as an example I have just put an ISDN system up a few days ago.. I am still in 
the process of getting my head around it but it terms of hardware and software I have 
used chan_capi and a passive AVM PCI card that I picked up off Ebay here in the UK for 
£4.20..

So far I have just got it to the point where I am able to make calls and have not had 
the serious echo problems that everyone warns about when using a passive card.. This 
is a very cheap way to get a system up and running and to get your head around it.. 
Then after that you can move on to the more expensive PRI/T1/E1 type systems..

The AVM passive card was suggested to me by Kapjod (The creator of chan_capi.) and it 
was really good advice.. I have a system now that has cost me very little to setup but 
is fully functional.. Also I don't know what the availablity of these cards is in the 
US..

I'm sorry but can't answer you questions about DID.. I am still learning.. :)

Later..

> I want to test asterisk with its DID functions on a smaller scale without the 
> expense of a PRI.  I am in the US, and want the most reliable/stable, and feature 
> rich solution that I can have.
> 
> What ISDN card would be reccommended?  
> 
> My clec switch operator says he can/will send DID info to my BRI's if I wish.  Can 
> Asterisk, or one of its addons support this?
> 
> What BRI solution would be reccommended?  ISDN4Linux, chan_capi?  Any info is 
> appreciated, I hope to get this right the first time without having to buy more 
> hardware after the initial purchase.
> 
> Any advice from you guys with experience would be appreciated.
> 
Chad
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