Re: [Asterisk-Users] Monitor

[Asterisk]

Comments below:
Anton Krall wrote:
Guys.
 
How does monitor work? Ive enabled the feature to start monitoring when *!
is pressed but I see that my calls are left with some IN and OUT file... how
can I merge those into one? 
Check out the "m" option
 
Also, when does asterisk records a call? I know I configured it to record
queue calls... but what else?
 
Ah! which brings me to another question, when using queues, agents signin
and they get MOH until a user calls but... on other call center apps, the
agents signin and can actually hangup the phone, which rings when a call
comes thru can asterisk behave in this manner or do the agents have to
be offhook for this?
 
Check out the AgentCallBackLogin cmd
Thx!
 
__
Anton Krall
 

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Julian
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[Asterisk-Users] SIP echo on LAN

[Nic le Roux]

Good 
Morning,
 
I have a weird 
situation,
I'm testing with 
Xlite as SIP phone (is it any good ) and dialing an extension (also Xlite on 
same LAN) and I'm getting a real bad echo on the dialer's side and a not so bad 
one on the receivers side.
 
Has anyone had 
something like this ?
Aparently one should 
only get echo when you break out onto a telco network ?
 
 
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[Asterisk-Users] Monitor

[Anton Krall]

Guys.
 
How does monitor work? Ive enabled the feature to start monitoring when *!
is pressed but I see that my calls are left with some IN and OUT file... how
can I merge those into one? 
 
Also, when does asterisk records a call? I know I configured it to record
queue calls... but what else?
 
Ah! which brings me to another question, when using queues, agents signin
and they get MOH until a user calls but... on other call center apps, the
agents signin and can actually hangup the phone, which rings when a call
comes thru can asterisk behave in this manner or do the agents have to
be offhook for this?
 
Thx!
 
__
Anton Krall
 

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[Asterisk-Users] Thank You Note

[Anton Krall]

Guys.. in the past couple of days I have been asking A LOT of questions and
I just want to take a minute to thank everybody that has helped me and will
continue to help newbies on their understand of Asterisk.
 
Also, I hope Asterisk's makers are reading this.. I want to congratulate you
on an excellent piece of software!! I LOVE ASTERISK!
 
Hope you will have the support and stamina to continue making this great
software.
 
Thank you!
 
__
Anton Krall
 

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[Asterisk-Users] CallerID

[Anton Krall]

Guys... I see there is a callerid parameter on zapata.conf... what does that
cid modify? the callerid people see when you call them using any PSTN line? 
 
Is there a way to send the SIP phone the incoming callerid frpm PSTN lines
asrecevied and append some string depending on the line it is coming from?
 
__
Anton Krall
 

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[Asterisk-Users] Conference between 2 lines

[Anton Krall]

Is there a way to make a join conference between 2 lines? like when you have
2 incoming calls and you merge them together with you? how can you do this
on * if its possible?
 
__
Anton Krall
 

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Re: SV: [Asterisk-Users] Newbie question

[Tim De Lange]

Hi!

Thanks! Now I'll just have to figure out what to put in extensions.conf

Thanks for the help, I will ask on this list again if I can't figure it
out.

Regards

TIm


On Sat, 2005-02-19 at 09:53, Thorben Jensen wrote:
> Hi Tim,
> 
> You could put he call back into the queue when the dial times out. Check for
> the length of the CALLERID, if it's equal to the length of your internal
> numbers then goto voicemail otherwise goto the queue.
> 
> thorben
> 
> -Oprindelig meddelelse-
> Fra: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] PÃ vegne af Tim De Lange
> Sendt: 18. februar 2005 14:25
> Til: asterisk-users@lists.digium.com
> Emne: [Asterisk-Users] Newbie question
> 
> Hello!
> 
> When the oprator transfers calls to internal extensions to unavailable
> or busy extensions, how can I prevent these calls from going to
> voicemail, and route them back to the oprator?  But other calls, ie
> internal between extensions, and calls coming in via DID should get
> voicemail if extensions are busy / unavailable?
> 
> Any help be appreciated.
> 
> TIA!
> 
> Tim
> 
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Re: [Asterisk-Users] Re: Ring/Off-hook in strange state 6 on channel...

[Eric Wieling]

I no not have call progress or busydetect enabled.
[EMAIL PROTECTED] wrote:
Hello Eric,
 

call progress detection is the problem. Asterisk mistakenly recognizes 
the call to be answered and then still "hears" the ringing that should 
not be there if the line was really up.

 

To solve the problem you would have to either implement a progress 
detection matching your country's indication tones or at least adjust 
the existing one for US or Costa Rica in dsp.c.

 

By the way on the "Read the damn docs." issue:
You could choose not to answer these questions. Anyone not getting an 
answer here would possibly take a second look at the docs and find what 
he might have overlooked.

 

Regards
Tobias
 

 

 > I have a Channelized Voice T-1 (FXO channels) configured as fxs_ks
 > signalling and am getting the following messages on the console:
 >
 > Feb 18 15:22:33 WARNING[1862]: chan_zap.c:3428 zt_handle_event:
 > Ring/Off-hook in strange state 6 on channel 2
 >
 > Does anyone have any idea what is causing this and how to fix it?  The
 > mailing list archives had lots of questions, but no answers regarding
 > this error message.

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This message has been 'sanitized'.  This means that potentially
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[Asterisk-Users] Fwd: res_config_mysql & chan_iax2 socket_read error

[Christ Nguyen]

Dear all,

I just update my asterisk from v 1.0 to lasted CVS, I also used module
res_config_mysql in asterisk-addons. Every working fine, but i got
problem with IAX user can't make a call, when IAX user make a call, i
got message in console like this
Feb 21 06:20:07 NOTICE[365]: chan_iax2.c:6090 socket_read: Rejected
connect attempt from
xxx.xxx.xxx.xxx, who was trying to reach '411@'
411 is the one of my exten number. sip user can call this number
without any problem.
Could any one help me to fixed this ?

Thank you very much,

Christ
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[Asterisk-Users] Re: Ring/Off-hook in strange state 6 on channel...

[Tobias . Cermann]

Hello Eric,
 
call progress detection is the problem. Asterisk mistakenly recognizes the 
call to be answered and then still "hears" the ringing that should not be there 
if the line was really up.
 
To solve the problem you would have to either implement a progress 
detection matching your country's indication tones or at least adjust the 
existing one for US or Costa Rica in dsp.c.
 
By the way on the "Read the damn docs." issue:
You could choose not to answer these questions. Anyone not getting an 
answer here would possibly take a second look at the docs and find what he might 
have overlooked.
 
Regards
Tobias
 
 
> I have a Channelized Voice T-1 (FXO channels) configured as fxs_ks 
> signalling and am getting the following messages on the console:
> 
> Feb 18 15:22:33 WARNING[1862]: chan_zap.c:3428 zt_handle_event: 
> Ring/Off-hook in strange state 6 on channel 2
> 
> Does anyone have any idea what is causing this and how to fix 
it?  The > mailing list archives had lots of questions, but no 
answers regarding > this error message.
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[Asterisk-Users] Sangoma A101

[Altus Snyman]

Good day all
Is there any difference in the sangoma zaptel.conf and zapata.conf then
other cards
Thanks
altus

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Re: [Asterisk-Users] How to announce the DNID to the called party

[Howard Lowndes]

On Mon, 2005-02-21 at 15:50, dkwok wrote:
> How to announce the DNID to the called party who picks up the phone and 
> say the correct greeting?
> 
> I suppose it has to say to the called party before the call is bridged. 
> So it has to do something before the dial command transfer the call.
> 
> Any ideas?

Check out the "A" option to the Dial command.

> 
> David Kwok
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-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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Re: [Asterisk-Users] Using Hylafax and Digium T100P

[Lee Howard]

On 2005.02.16 18:57 Darren Nickerson wrote:
Granted the Patton 2997 is limited to 14,400 maximum
This is off-topic for this thread, but the Patton 2977 does support 
V.34-Fax (speeds 2400 through 33600 baud), at least in recent firmware 
revisions.  However, you are correct in that the card traditionally has 
not, and still it only supports V.34-Fax in Class 2.1 mode (in other 
words, not in a Class 1.0 yet).

Lee.
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[Asterisk-Users] SIP to SIP calls have no audio until put on hold and taken back off - SOLVED

[David Ludlow]

Thanks to Pau (the original person to pose the question on this list), it's fixed.  The firewall was getting in the way.  I needed to open up UDP ports 1 to 2 for RTP traffic.

See the following for more info:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20rtp.conf
http://www.voip-info.org/wiki-Asterisk+firewall+rules


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Re: [Asterisk-Users] Asterisk H323 support

[Андрей Кочетков]

Здравствуйте Asterisk Users Mailing List - Non-Commercial Discussion,

Monday, February 21, 2005, 2:14:20 PM, Вы писали:

===8<==Original message text===
kolo sos> Hi,

kolo sos> anybody knows what's missing or problem why i cant
kolo sos> compile asterisk-oh323 in my machine?

kolo sos> i got this compiled successfully

kolo sos> Openh323 - v1.12.2
kolo sos> pwlib - v1.5.2

kolo sos> except 

kolo sos> asterisk-oh323 - v0.6.5

kolo sos> here's the output as i run make...

kolo sos> [EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$ make
kolo sos> for x in wrapper asterisk-driver; do make -C $x build
kolo sos> || exit 1 ; done
kolo sos> make[1]: Entering directory
kolo sos> `/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper'
kolo sos> ../check_ver /home/mkoy/pwlib pwlib
kolo sos> ../check_ver /home/mkoy/openh323 openh323
kolo sos> g++ -DP_LINUX=2.4.26 -ffunction-sections
kolo sos> -fdata-sections -D_REENTRANT -Wall -fPIC
kolo sos> -DP_USE_PRAGMA -DPHAS_TEMPLATES
kolo sos> -I/home/mkoy/pwlib/include/ptlib/unix
kolo sos> -I/usr/include/pwlib -I/home/mkoy/pwlib/include
kolo sos> -DPTRACING -I/home/mkoy/openh323/include -DHAS_IXJ
kolo sos> -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING
kolo sos> -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\"1.5.2\"
kolo sos> -DOPENH323VERSION=\"1.12.2\" 
kolo sos> -I/home/mkoy/pwlib/include/ptlib/unix
kolo sos> -I/home/mkoy/pwlib/include
kolo sos> -I/home/mkoy/openh323/include
kolo sos> -I/home/mkoy/openh323/include/openh323
kolo sos> -I../asterisk-driver -c asteriskaudio.cxx -o
kolo sos> asteriskaudio.o
kolo sos> asteriskaudio.cxx: In destructor `virtual
kolo sos>PAsteriskSoundChannel::~PAsteriskSoundChannel()':
kolo sos> asteriskaudio.cxx:167: error: `baseChannel' undeclared
kolo sos> (first use this
kolo sos>function)
kolo sos> asteriskaudio.cxx:167: error: (Each undeclared
kolo sos> identifier is reported only once
kolo sos>for each function it appears in.)
kolo sos> make[1]: *** [asteriskaudio.o] Error 1
kolo sos> make[1]: Leaving directory
kolo sos> `/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper'
kolo sos> make: *** [subdirs_build] Error 1
kolo sos> [EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$



kolo sos> Kolosos
kolo sos> Philippines



kolo sos> __ 
kolo sos> Do you Yahoo!? 
kolo sos> Meet the all-new My Yahoo! - Try it today! 
kolo sos> http://my.yahoo.com 
 

kolo sos> ___
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kolo sos> http://lists.digium.com/mailman/listinfo/asterisk-users
kolo sos> To UNSUBSCRIBE or update options visit:
kolo sos>http://lists.digium.com/mailman/listinfo/asterisk-users

===8<===End of original message text===

  For channel asterisk-oh323-v0.6.5
  need
  openh323-Janus_patch4-src-tar.gz
  pwlib-Janus_patch4-src-tar.gz


-- 
С уважением:
Андрей Кочетков
ООО "Современные Системы Связи", "Мобил-Телеком"
Чита, ул. Заб. Рабочего, 94
тел.: 8 (3022) 23-33-33 
mailto:[EMAIL PROTECTED]

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[Asterisk-Users] HELP NEEDED! - Asterisk GUI

[Julius Kidubuka]

Hello,

I am trying to setup an Asterisk GUI with the help of astman(please visit
http://astman.sourceforge.net/am-user-guide.html).

I have installed astman and currently assessing my GUI using;
http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl

I am trying to get the menu options in my GUI to work but to no avail.
Currently my parameters are set to;

Asterisk Install Directory: /usr/ports/net/asterisk/work/asterisk-1.0.3/
Asterisk Config Directory: /usr/local/etc/asterisk
Profile Editor Working Directory: /usr/local/etc/asterisk

Any ideas on how I can go about this?

Thanks in advance.

-- 
Rgds,
Julius Kidubuka.
"My advice to you is get married: if you find a good wife you'll be happy;
if not, you'll become a philosopher."




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[Asterisk-Users] Asterisk H323 support

[kolo sos]

Hi,

anybody knows what's missing or problem why i cant
compile asterisk-oh323 in my machine?

i got this compiled successfully

...Openh323 - v1.12.2
...pwlib - v1.5.2

except 

...asterisk-oh323 - v0.6.5

here's the output as i run make...

[EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$ make
for x in wrapper asterisk-driver; do make -C $x build
|| exit 1 ; done
make[1]: Entering directory
`/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper'
./check_ver /home/mkoy/pwlib pwlib
./check_ver /home/mkoy/openh323 openh323
g++ -DP_LINUX=2.4.26 -ffunction-sections
-fdata-sections -D_REENTRANT -Wall -fPIC
-DP_USE_PRAGMA -DPHAS_TEMPLATES
-I/home/mkoy/pwlib/include/ptlib/unix
-I/usr/include/pwlib -I/home/mkoy/pwlib/include
-DPTRACING -I/home/mkoy/openh323/include -DHAS_IXJ
-DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING
-DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\"1.5.2\"
-DOPENH323VERSION=\"1.12.2\" 
-I/home/mkoy/pwlib/include/ptlib/unix
-I/home/mkoy/pwlib/include
-I/home/mkoy/openh323/include
-I/home/mkoy/openh323/include/openh323
-I../asterisk-driver -c asteriskaudio.cxx -o
asteriskaudio.o
asteriskaudio.cxx: In destructor `virtual
   PAsteriskSoundChannel::~PAsteriskSoundChannel()':
asteriskaudio.cxx:167: error: `baseChannel' undeclared
(first use this
   function)
asteriskaudio.cxx:167: error: (Each undeclared
identifier is reported only once
   for each function it appears in.)
make[1]: *** [asteriskaudio.o] Error 1
make[1]: Leaving directory
`/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper'
make: *** [subdirs_build] Error 1
[EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$



Kolosos
Philippines



__ 
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Meet the all-new My Yahoo! - Try it today! 
http://my.yahoo.com 
 

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Re: [Asterisk-Users] * > Mobile Phone > Mobile Network

[ian sison (mailing list)]

Another option is something like cellsocket http://www.cellsocket.com
I haven't tried these, but some positive experiences posted in 
some sites ive been googling seem encouraging.

There are models for motorola and nokia phones.



On Mon, 21 Feb 2005 15:21:59 +1100, Mathew McKernan <[EMAIL PROTECTED]> wrote:
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of David
> Uzzell
> Sent: Monday, 21 February 2005 2:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] * > Mobile Phone > Mobile Network
> 
> Ok I have a question. Seen it come and go around the mailling list for a
> 
> while but never really seen an answer that seems to sort it out.
> 
> What is needed is some interface from * > Mobile Phone > Mobile Network
> Service.
> 
> At this point all the providers in AUS that I have found are charging a
> Premium Rate for Land Line > Mobile Network services.
> 
> What I would like to do is be able to purchase a low rate Mobile SIM
> that I can chuck into a Mobile Phone and have it setup so that I route
> the Mobile calls through it.
> 
> Rembering that most if not all mobile phones can be accessed via RS232
> interface.
> 
> Anyone done this or seen it done or know how to do it using * and
> whatever?
> 
> Cheers
> David
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> 
> Hi David,
> 
> Have a look at some second hand Ericson kits on Ebay. They had special
> units, that basically had a normal GSM Ericson phone in them. But on the
> side had a normal Australian 610 socket and rj11 socket.
> 
> You could simply interface this into your digium cards as a normal pstn
> line.
> 
> They were originally designed for the exact purpose you want for
> coupling with existing telephone systems. They are also used for
> connection to fire signalling units and alarm systems.
> 
> Thanks
> 
> Mathew McKernan
> Digital World Computers
> Maribyrnong VIC
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Re: [Asterisk-Users] NAT and FWD

[Duane]

Anton Krall wrote:
> I was also looking for a way to do SIP while asterisk is behind a firewall
> so SIP Phones on the outside can connect.

like2fone.com is our sip service for e164.org, we have SER+Asterisk
setup and SER has transparent proxying if detects your connect is natted...

-- 

Best regards,
 Duane

http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers

"In the long run the pessimist may be proved right,
but the optimist has a better time on the trip."
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[Asterisk-Users] How many line appearance can Snom 200 handle?

[dkwok]

Snom 200 has be set up with extended key pad. The product literature 
also mention multiple sip registration.

How many registration can it handle? It does not seem to appear in the 
user manual.

David Kwok
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[Asterisk-Users] How to announce the DNID to the called party

[dkwok]

How to announce the DNID to the called party who picks up the phone and 
say the correct greeting?

I suppose it has to say to the called party before the call is bridged. 
So it has to do something before the dial command transfer the call.

Any ideas?
David Kwok
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Re: [Asterisk-Users] Modem as PSTN interface?

[Steven Critchfield]

On Sun, 2005-02-20 at 22:14 -0600, Jon Gabrielson wrote:
> I believe that is basically what the generic t100p is.
> Also, several other voicemodems are already supported
> by asterisk.  To my knowledge they are all FXO ports.
> I don't believe there are any modems that can provide
> FXS ports.  If someone knows of one, I would be interested
> in knowing about it.

Nope, a t100P is a T1 span. Nothing generic there.

The voicemodems that are "supported" are winmodems that happen to behave
similarly enough to the X100P card that they can be dealt with the same
way. Since winmodems move all the DSP work into the main CPU, it means
they are mostly simple PSTN digitizer. Instead of hooking DSP code to
the digitized audio path, asterisk just uses the audio.

Trouble is, most winmodem companies keep the implementation details very
quite. If you can't get that data you can't get it to work.

Next trouble is that no one here gets any benefit from your cheapest way
to play route. In fact your cheapest way to play is a detriment to our
community as you are likely to have a very bad experience and either bad
mouth the software because of your experience, or you may never get past
the basic user position and are just an occasional support drain. All
this without having contributed back to the community. There is a chance
you might move beyond that point, but it isn't likely.

So try and make a convincing case for anyone to beat their head against
the wall to support a low port density device that is unlikely to be
used by a large portion of the community and is not going to benefit
either the writer or the main supporters of asterisk.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] * > Mobile Phone > Mobile Network

[Paul Hales]

We have a bank of the Telular units here - all will capped sims in them.

Works well. 

Later,

PaulH

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Uzzell
Sent: Monday, 21 February 2005 2:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] * > Mobile Phone > Mobile Network

Ok I have a question. Seen it come and go around the mailling list for a while 
but never really seen an answer that seems to sort it out.

What is needed is some interface from * > Mobile Phone > Mobile Network Service.

At this point all the providers in AUS that I have found are charging a Premium 
Rate for Land Line > Mobile Network services.

What I would like to do is be able to purchase a low rate Mobile SIM that I can 
chuck into a Mobile Phone and have it setup so that I route the Mobile calls 
through it.

Rembering that most if not all mobile phones can be accessed via RS232 
interface.

Anyone done this or seen it done or know how to do it using * and whatever?

Cheers
David
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Re: [Asterisk-Users] Modem as PSTN interface?

[Jon Gabrielson]

I believe that is basically what the generic t100p is.
Also, several other voicemodems are already supported
by asterisk.  To my knowledge they are all FXO ports.
I don't believe there are any modems that can provide
FXS ports.  If someone knows of one, I would be interested
in knowing about it.


Jon.
  

On Sunday 20 February 2005 09:35 pm, PHP Mechanic wrote:
> > Hi, all
> >
> > Can a normal PCI modem be used to provide PSTN interface? I have seen
> > modems that have answering machine capabilities, so there should not be a
> > problem sending voice through them.
> >
> > Certainly, modem will be cheaper option then dedicated cards. Am I
> > missing something?
>
> Most modems don't operate at full-duplex. A normal modem can be used to
> send voice, then switch to recording voice, but it can't send and receive
> simultaneously.
>
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RE: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4C.O. lines

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> Jim Van Meggelen wrote:
>> [EMAIL PROTECTED] wrote:
>> 
>>> Jim Van Meggelen wrote:
>>> 
>>> 
 Yep, that's a possibility, but it's rather more kludgy than I'd
 like. (heck, the channel bank and T1 is more kludgy than I'd like
 - an integrated card would be my preference).
>>> 
>>> I haven't followed this thread closely but have you looked into the
>>> Voicetronix OpenSwitch cards? 
>>> 
>>> http://www.voicetronix.com.au/hda.htm
>> 
>> 
>> I've looked at them, but never heard much about them. Is anyone using
>> them? Can anyone give a comparison vs. the TDM400?
> 
> I'm using a Voicetronix OpenLine4, and it works well under asterisk.
> Initially I had some echo problems, but Voicetronix support
> is excellent and
> solved them (I've just updated the wiki with the bal# values
> they gave me).
> 
> I can't compare it to the TDM400, not having used one, but
> you can use
> multiple Voicetronix OpenSwitch 6 and 12 cards in one system
> without the
> interrupt problem of the TDM400.

That sounds like the ticket, then.

Thanks.

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005
 

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RE: [Asterisk-Users] * > Mobile Phone > Mobile Network

[Mathew McKernan]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Uzzell
Sent: Monday, 21 February 2005 2:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] * > Mobile Phone > Mobile Network

Ok I have a question. Seen it come and go around the mailling list for a

while but never really seen an answer that seems to sort it out.

What is needed is some interface from * > Mobile Phone > Mobile Network 
Service.

At this point all the providers in AUS that I have found are charging a 
Premium Rate for Land Line > Mobile Network services.

What I would like to do is be able to purchase a low rate Mobile SIM 
that I can chuck into a Mobile Phone and have it setup so that I route 
the Mobile calls through it.

Rembering that most if not all mobile phones can be accessed via RS232 
interface.

Anyone done this or seen it done or know how to do it using * and
whatever?

Cheers
David
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Hi David,

Have a look at some second hand Ericson kits on Ebay. They had special
units, that basically had a normal GSM Ericson phone in them. But on the
side had a normal Australian 610 socket and rj11 socket. 

You could simply interface this into your digium cards as a normal pstn
line.

They were originally designed for the exact purpose you want for
coupling with existing telephone systems. They are also used for
connection to fire signalling units and alarm systems.

Thanks

Mathew McKernan
Digital World Computers
Maribyrnong VIC
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RE: [Asterisk-Users] Modem as PSTN interface?

[Rudolf Ladyzhenskii]

Thanks.

There goes a good idea ;=(

Rudolf

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of PHP
Mechanic
Sent: Monday, February 21, 2005 2:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Modem as PSTN interface?


> Hi, all
>
> Can a normal PCI modem be used to provide PSTN interface? I have seen 
> modems that have answering machine capabilities, so there should not be a 
> problem sending voice through them.
>
> Certainly, modem will be cheaper option then dedicated cards. Am I missing 
> something?

Most modems don't operate at full-duplex. A normal modem can be used to send 
voice, then switch to recording voice, but it can't send and receive 
simultaneously. 

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[Asterisk-Users] DevKit Lite Questions/issues

[Doug Jaworski]

I am having some issues with the configuration with the DevKit Lite
It is not clear as to whether the configuration files are empty to start 
with or appended to.

Files in question are:
/etc/asterisk/zapata.conf
/etc/asterisk/extensions.conf
Your assistance is appreciated.
Thanks,
Doug
Q.I just recieved my DevKit Lite. What should I do now, and can you 
please show me?
A.Here is step by step guide:
a) Go into /usr/src directory on your linux box
> cd /usr/src

b) Download the latest CVS release of zaptel,asterisk packages
> export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
> cvs login (password is anoncvs)
> cvs checkout zaptel asterisk
c) Compile all the packages in this order: zaptel/asterisk
> cd /usr/src/zaptel; make install;
> cd ../asterisk
> make install
> make samples
d) Edit configuration files and make sure they have at least these 
lines: /etc/zaptel.conf
fxsks=1 #X100P
fxoks=2 #S100U
defaultzone=us
loadzone=us

/etc/asterisk/zapata.conf
[channels]
signalling=fxs_ks
context=incoming
channel=>1 ;X100P
context=internal
signalling=fxo_ks
channel=>2 ;S100U
/etc/asterisk/extensions.conf
[incoming]
exten => s,1,Dial,Zap/2|20
exten => s,2,Voicemail,u1000
exten => s,102,Voicemail,b1000
[outgoing]
;this extension is for dialing out local numbers
;the number starts with the 9 digit and is 7 digits long
exten => _9NXX,1,Dial,Zap/1/${EXTEN:1}
;this extensions is for dialing out long distance numbers
;the number starts with the 91 digits and is 10 digits long
exten => _91NXXNXX,1,Dial,Zap/1/${EXTEN:1}
;this extension is for dialing out international numbers
;the number starts with 9011 and is at least 4 digits long
exten => _9011.,1,Dial,Zap/1/${EXTEN:1}
;this extension is for checking voicemaill
;don't ask for voicemail number and password
exten => 8500,1,VoiceMailMain,s1000
/etc/asterisk/voicemail.conf
[general]
format=gsm|wav49|wav
maxmessage=180
[incoming]
1000 => 1000,Example Mailbox
e) load the apropriate modules
> modprobe zaptel
> modprobe wcfxo *
> modprobe wcusb
> ztcfg -vv **
* Note that you might have errors after "modprobe wcfxo" because in the 
file /etc/modules.conf there are lines like that: post-install wcfxo 
/sbin/ztcfg These lines load zaptel configuration program right after 
every "modprobe driver" command. So if you configured more than one 
device (X100 and S100U) you're going to have error message after loading 
"modprobe wcfxo". That's normal.

** If you have errors after you executed all four lines then it's 
propable that your system loaded "audio" module for S100U. This module 
is a USB sound card module and you have to unload it by executing "rmmod 
audio". After that execute one more time "ztcfg -vv" to be sure that 
everything is OK.

**2 The problem may be that you don't have a USB controller support 
turned on on your motherboard or USB support is not properly configured. 
Check if you have any entry of USB controller in /proc/pci system file. 
Also check if the USB controller driver is loaded in the kernel (lsmod). 
You should have one of these drivers loaded: usb-uhci,uhci,ohci. If you 
don't have it loaded you may try to load it using modprobe command, eg:
>modprobe usb-uhci
After the proper module is loaded when you plug in/out the S100U you 
should see some messages poping up on the console. If no messages pop up 
then remove the driver and try another one, eg: >rmmod usb-uhci; 
modprobe uhci
If that doesn't work you propably need to compile your own kernel with 
USB support or download a diffrent linux distribution.

Note also that if you "make configs" in the zaptel sources than there 
will be installed boot-up script to handle modprobing drivers for you.

f) Now that everything is installed properly run
> asterisk -vvvc
You may dial in now to the analog phone line connected to X100P and you 
should be able to connect to the phone connect to S100U. You should also 
be able to dial out through the phone connected to S100U by dialing out 
9 + number (for US local calls) or 91 + area + number (for US long 
distance calls) or 9011 + country code + area code + number (for 
international calls) When you dial 8500 you'll be able to check your 
voce mail box.

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[Asterisk-Users] * > Mobile Phone > Mobile Network

[David Uzzell]

Ok I have a question. Seen it come and go around the mailling list for a 
while but never really seen an answer that seems to sort it out.

What is needed is some interface from * > Mobile Phone > Mobile Network 
Service.

At this point all the providers in AUS that I have found are charging a 
Premium Rate for Land Line > Mobile Network services.

What I would like to do is be able to purchase a low rate Mobile SIM 
that I can chuck into a Mobile Phone and have it setup so that I route 
the Mobile calls through it.

Rembering that most if not all mobile phones can be accessed via RS232 
interface.

Anyone done this or seen it done or know how to do it using * and whatever?
Cheers
David
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[Asterisk-Users] Where to contrib the sound files ?

[david]

Hello,every one,
 
    I have recorded the voice files 
with mandarin (China). Where should I contrib the files ?
 
    Regards.
 
    David at iaxtalk.com
 
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Re: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4 C.O. lines

[James Andrewartha]

Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
Jim Van Meggelen wrote:

Yep, that's a possibility, but it's rather more kludgy than I'd like.
(heck, the channel bank and T1 is more kludgy than I'd like - an
integrated card would be my preference).
I haven't followed this thread closely but have you looked into the
Voicetronix OpenSwitch cards? 

http://www.voicetronix.com.au/hda.htm

I've looked at them, but never heard much about them. Is anyone using
them? Can anyone give a comparison vs. the TDM400?
I'm using a Voicetronix OpenLine4, and it works well under asterisk. 
Initially I had some echo problems, but Voicetronix support is excellent and 
solved them (I've just updated the wiki with the bal# values they gave me).

I can't compare it to the TDM400, not having used one, but you can use 
multiple Voicetronix OpenSwitch 6 and 12 cards in one system without the 
interrupt problem of the TDM400.

James Andrewartha
DAA Sysadmin
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Re: [Asterisk-Users] Modem as PSTN interface?

[PHP Mechanic]

Hi, all
Can a normal PCI modem be used to provide PSTN interface? I have seen 
modems that have answering machine capabilities, so there should not be a 
problem sending voice through them.

Certainly, modem will be cheaper option then dedicated cards. Am I missing 
something?
Most modems don't operate at full-duplex. A normal modem can be used to send 
voice, then switch to recording voice, but it can't send and receive 
simultaneously. 

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[Asterisk-Users] zt hook failed: Device or resource busy

[Eric Wieling]

I'm getting the message "zt hook failed: Device or resource busy". 
Not all the time, just sometimes.

As you can see it seems to happen when a port is hung up and then 
picked back up within 1 second.  What /etc/asterisk/zaptel.conf 
options might I use to fix this?  debounce=?  flash=?  something else?

Feb 20 18:31:48 VERBOSE[5583]: -- Starting simple switch on 'Zap/65-1'
Feb 20 18:31:53 VERBOSE[5583]: -- Executing Dial("Zap/65-1", 
"Zap/G1/5044659580") in new stack
Feb 20 18:31:53 VERBOSE[5583]: -- Called G1/504XXX
Feb 20 18:31:57 VERBOSE[5583]: -- Zap/27-1 answered Zap/65-1
Feb 20 18:31:57 VERBOSE[5583]: -- Attempting native bridge of 
Zap/65-1 and Zap/27-1
Feb 20 18:31:59 VERBOSE[5583]: -- Hungup 'Zap/27-1'
Feb 20 18:31:59 VERBOSE[5583]:   == Spawn extension (channel-bank, 
5044659580, 1) exited non-zero on 'Zap/65-1'
Feb 20 18:31:59 VERBOSE[5583]: -- Hungup 'Zap/65-1'
Feb 20 18:36:32 VERBOSE[5584]: -- Starting simple switch on 'Zap/65-1'
Feb 20 18:36:44 VERBOSE[5584]: -- Hungup 'Zap/65-1'
Feb 20 18:36:44 WARNING[5070]: zt hook failed: Device or resource busy
Feb 20 18:36:44 VERBOSE[5585]: -- Starting simple switch on 'Zap/65-1'
Feb 20 18:36:56 VERBOSE[5585]: -- Hungup 'Zap/65-1'
Feb 20 18:37:08 VERBOSE[5586]: -- Starting simple switch on 'Zap/65-1'
Feb 20 18:37:14 VERBOSE[5586]: -- Executing Dial("Zap/65-1", 
"Zap/G1/5043050452") in new stack
Feb 20 18:37:14 VERBOSE[5586]: -- Called G1/504XXX
Feb 20 18:37:19 VERBOSE[5586]: -- Zap/27-1 answered Zap/65-1
Feb 20 18:37:19 VERBOSE[5586]: -- Attempting native bridge of 
Zap/65-1 and Zap/27-1
Feb 20 18:38:35 VERBOSE[5586]: -- Hungup 'Zap/27-1'

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[Asterisk-Users] Modem as PSTN interface?

[Rudolf Ladyzhenskii]

Hi, all

Can a normal PCI modem be used to provide PSTN interface? I have seen modems 
that have answering machine capabilities, so there should not be a problem 
sending voice through them.

Certainly, modem will be cheaper option then dedicated cards. Am I missing 
something?

/***/
Rudolf Ladyzhenskii
Senior Design Engineer
Open Networks Pty. Ltd.
Level 26, 35 Collins Street,
Melbourne VIC 3000
e-mail: [EMAIL PROTECTED]
phone: +61 3 9656 5107
fax: +61 3 9656 5122
web: www.opennw.com
/***/


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Re: [Asterisk-Users] Phones for vitural office business

[Adam Goryachev]

On Mon, 2005-02-21 at 13:25 +1100, dkwok wrote:
> I am looking for IP phones that are suited to serviced office operation.
> 
> The business is to answer calls for customers. The incoming lines are E1
> and customers are allocated with DID. So the customers' phone can be
> answered with the customers' designed messages and instruction. This can
> be done easily with key phone system. It seems to be a problem for IP
> phone system. I read previous discussions on the pros and cons of key
> system and ip phone system. However, still need to offer an solution to
> the operation
> whereby when calls come in, the operators will be able to identify whose
> customer's call and answer for that customer accordingly.

IMHO, any phone with some lights/buttons on it will not be helpful,
unless you will only ever have less than X different customers/DID's.
IMHO, you need some sort of simple PC based app, and, if you think about
it, you could do a lot more than any other key system.

eg, When the call arrives, lookup the DID in the database, and retrieve
the call script/announcement, then send that data to your client
application along with the CND, and ring the phone.

You know now what to say, and who is calling before you even answer the
phone.

Extra points if you also include some work to see that this person has
called previously and left three messages in the last 20 minutes. (even
if the calls are being answered by different people, you could still let
them know that the previous messages have been sent, etc...

If you would like someone to write this solution, contact me off-list,
or see the wiki for asterisk consultants...

Regards,
Adam

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[Asterisk-Users] Adding zap channels under *@Home

[Robert Webb]

Hi all,

  With the ability of an easy install using [EMAIL PROTECTED], I have
decided to give it a try. It is my understanding though, that one cannot
add zap fxs ports as extensions using AMP. Is there anyone using
[EMAIL PROTECTED] and have added any extensions as zap fxs channels? Would
be interested in how you accomplished this.

Robert



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[Asterisk-Users] PLease help: Asterisk to Quintum interconnection

[Jessie V. Mabanglo]

My fellows,
 
We have [EMAIL PROTECTED] installed and we want to 
interconnect it with our existing quintum gateways.. any idea how to config 
that?
 
Your time is very much appreciated..
 
Cheers,
 
Jessie
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[Asterisk-Users] Phones for vitural office business

[dkwok]

I am looking for IP phones that are suited to serviced office operation.
The business is to answer calls for customers. The incoming lines are E1
and customers are allocated with DID. So the customers' phone can be
answered with the customers' designed messages and instruction. This can
be done easily with key phone system. It seems to be a problem for IP
phone system. I read previous discussions on the pros and cons of key
system and ip phone system. However, still need to offer an solution to
the operation
whereby when calls come in, the operators will be able to identify whose
customer's call and answer for that customer accordingly.
I have a look at Snom 220 with console. Does the line appearance
function solve this problem if DID is assigned to difference line of the
ip phone?
Any suggestion will be welcome.
David Kwok
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RE: [Asterisk-Users] NAT and FWD

[Anton Krall]

I was also looking for a way to do SIP while asterisk is behind a firewall
so SIP Phones on the outside can connect.
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Domingo, 20 de Febrero de 2005 07:08 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] NAT and FWD

I think you are trying to use SIP with FWD, isn't it?

Set FWD over IAX, follow the instructions:
http://www.fwd.pulver.com/advanced/iax

--
#Joseph

On Sun, 2005-02-20 at 18:12 -0600, Anton Krall wrote:
> Guys.
>  
> Im trying to figure out how to confgure FWD and NAT. I tried some 
> configs and tested thru the FWD webpage the incoming call.. I got the 
> incoming fine except it kept asking me for my name and seems it didnt 
> get it.. (nat problems, one way sound only)?
>  
> Also, I cant dialout.. 
>  
> Can somebody using NAT send me some example configs?
>  
> Thx!


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Re: [Asterisk-Users] Asterisk "no one is available to take your call"

[Lyle Giese]

It does not state it will dial forever.  Ring forever maybe.

You are posting portions of your extension.conf for outgoing calls from
Asterisk only.  I don't see anything here that is for incoming calls and
forwarding to 4607 when the call is not answered.

Lyle

- Original Message - 
From: "Greg Oliver" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, February 18, 2005 3:58 PM
Subject: Re: [Asterisk-Users] Asterisk "no one is available to take your
call"


> True, but it also states that with no timeout value that it will dial
> until the caller hangs up.
>
> I have included my dial pattern - can anyone see anything that would
> cause this, or something in my sip.conf or h323.conf files that would
> override these settings?
>
> Thanks,
>
> Greg Oliver
>
> [outbound]
> exten => _4XXX,1,NoOp("Route to CCM1" ${EXTEN} )
> exten => _4XXX,2,Dial(H323/${EXTEN})
> exten => _4XXX,3,Congestion
>
> exten => _5XXX,1,NoOp("Route to CCM1" ${EXTEN} ) exten =>
> _5XXX,2,Dial(H323/${EXTEN})
> exten => _5XXX,3,Congestion
>
> exten => _9NX,1,NoOp("Route to CCM1" ${EXTEN})
> exten => _9NX,2,Dial(H323/${EXTEN})
>
> exten => _91NX,1,NoOp("Route to CCM1" ${EXTEN})
> exten => _91NX,2,Dial(H323/${EXTEN})
>
>
> default context includes outbound, and contexts in sip.conf and
> h323.conf are using default.  Like I say, call answered before ~5
> seconds are fine, other than that it is transferred to 4607..
>
> Howard Lowndes wrote:
> > On Wed, 2005-02-16 at 11:05, Greg Oliver wrote:
> >
> >>OK - I can successfully make calls from SIp phone through an asterisk
> >>323 channel to a Cisco Call Manager and out a MGCP controlled gateway.
> >>
> >>The problem is that if the call is not answered within ~5 seconds, *
> >>gives the message "no one is available to take your call" and
> >>disconnects the call.  If I answer b4 the 5 seconds - everything is good
.
> >>
> >>Anywhere I need to set to get around this.
> >>
> >>I have tried the t,T settings (even though the docs say no entry is
> >>forever) with no luck.
> >
> >
> > Read the doco on the Dial command again.  It's noting to do with the Tt
> > option, it's the parameter before that that you need to set to the
> > timeout
> >
> >>Thanks,
> >>
> >>Greg Oliver
> >>___
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RE: [Asterisk-Users] help with @home

[Kurt Fankhauser]

Title: Message



i got 
it

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Kurt 
  FankhauserSent: Sunday, February 20, 2005 8:01 PMTo: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  RE: [Asterisk-Users] help with @home
  i 
  have got the softphone working, now i am trying to setup voicemail for my 
  @home box, under extension i have voicemail & directory enabled but when i 
  call that extension it just keeps rining and never goes to 
  voicemail
   
  kurt
  

-Original Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt 
FankhauserSent: Sunday, February 20, 2005 7:24 PMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
RE: [Asterisk-Users] help with @home
i 
got the xlite phone on my pc and i have no idea how to get it working, do i 
have to add it into the * box or just change some settings on the 
softphone?

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of dean 
  collinsSent: Sunday, February 20, 2005 3:36 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  RE: [Asterisk-Users] help with @home
  
  Just download a 
  free softphone and do it that way eg xten
   
   
   
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Kurt 
  FankhauserSent: Sunday, 
  February 20, 2005 9:18 PMTo: 'Asterisk Users Mailing List - Non-Commercial 
  Discussion'Subject: RE: [Asterisk-Users] help 
  with @home
   
  
  I'll buy a IP 
  phone tomarrow so i can do that
  
-Original 
Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of dean 
collinsSent: Sunday, 
February 20, 2005 2:40 PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: RE: [Asterisk-Users] help 
with @home
Can you work 
through a process of elimination if you record the file using an 
internal extension by dialing *77 and seeing if that 
works?
 
 
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt 
FankhauserSent: 
Sunday, February 20, 2005 7:42 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] help with 
@home
 

just reinstalled @home and i 
have a one of those 100 cards, anyways when i call from the pstn the box 
picks up but i hear nothing, then it clicks a couple times, then nothing 
again, i am trying to get the digital receptionist to work but it won't 
save my wav file to the @home box and all the radio buttons under 
incoming calls are greyed out. the greyed out thing seems to be my 
biggest problem right now, also do you have to use a ip phone to record 
your greeting because this wav file stuff isn't 
working.

 
Kurt 
Fankhauser
WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 


 
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Re: [Asterisk-Users] Amateur - Problema when installing

[Lyle Giese]

Hmmm, maybe you need to re-read the instructions?  You missed a major step.

Try doing a make before make install.

make;make install

Lyle

- Original Message - 
From: "Paulo - Ibest" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, February 18, 2005 6:28 AM
Subject: [Asterisk-Users] Amateur - Problema when installing


> Friends,
>
> I'm in trouble, I tried to install de Asterisk, based on the site manual,
> into a RedHat 9.0, I followed every step, and it doesn't work.
> When I does the libpri make install, the message is:
>
> 
> [EMAIL PROTECTED] zaptel]# cd ..
> [EMAIL PROTECTED] src]# cd libpri/
> [EMAIL PROTECTED] libpri]# make clean; make install

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Re: [Asterisk-Users] NAT and FWD

[Joseph]

I think you are trying to use SIP with FWD, isn't it?

Set FWD over IAX, follow the instructions:
http://www.fwd.pulver.com/advanced/iax

-- 
#Joseph

On Sun, 2005-02-20 at 18:12 -0600, Anton Krall wrote:
> Guys.
>  
> Im trying to figure out how to confgure FWD and NAT. I tried some configs
> and tested thru the FWD webpage the incoming call.. I got the incoming fine
> except it kept asking me for my name and seems it didnt get it.. (nat
> problems, one way sound only)?
>  
> Also, I cant dialout.. 
>  
> Can somebody using NAT send me some example configs?
>  
> Thx!


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Re: [Asterisk-Users] Help with config.

[Lyle Giese]

How many of what kind of cards do you have in your 
* server?  
 
Because of the way * was programed, it's normally 
recommended to have only two cards max and they have to have unique and seperate 
IRQ's.  The zaptel.conf seems to indicate you already had two cards and are 
adding a third.
 
Lyle
 

  - Original Message - 
  From: 
  Lucas 
  Wrenn 
  To: asterisk-users@lists.digium.com 
  
  Sent: Friday, February 18, 2005 9:09 
  AM
  Subject: [Asterisk-Users] Help with 
  config.
  
  
  Hello 
  all.
   
  I am trying to get my second x100p 
  card set up and am having some troubles.
   
  My zaptel.conf 
  reads:
   
  fxsks=1-2
  fxoks=3-4
  defaultzone=us
  loadzone=us
   
  before adding 
  this card my zaptel.conf read:
   
  fxsks=1
  fxoks=2-3
  defaultzone=us
  loadzone=us
   
  But now that I’ve made the change 
  I am getting the following error when running modprobe wcfxo and of course the same 
  error if I use 
  /sbin/ztcfg
   
  The error 
  reads:
   
  ZT_CHANCONFIG failed on channel 3: 
  No such device or address (6)
   
  Any ideas on how to clean this 
  up?
   
  Even though it goes against 
  everything I have found online I even tried “fxsks=1,2” because they are two 
  physically different cards.
   
  Thanks for your 
  help.
  Stumped.
  ([EMAIL PROTECTED])
  
  

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RE: [Asterisk-Users] help with @home

[Kurt Fankhauser]

Title: Message



i have 
got the softphone working, now i am trying to setup voicemail for my @home box, 
under extension i have voicemail & directory enabled but when i call that 
extension it just keeps rining and never goes to voicemail
 
kurt

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Kurt 
  FankhauserSent: Sunday, February 20, 2005 7:24 PMTo: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  RE: [Asterisk-Users] help with @home
  i 
  got the xlite phone on my pc and i have no idea how to get it working, do i 
  have to add it into the * box or just change some settings on the 
  softphone?
  

-Original Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of dean 
collinsSent: Sunday, February 20, 2005 3:36 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
RE: [Asterisk-Users] help with @home

Just download a 
free softphone and do it that way eg xten
 
 
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt 
FankhauserSent: Sunday, 
February 20, 2005 9:18 PMTo: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'Subject: RE: [Asterisk-Users] help with 
@home
 

I'll buy a IP phone 
tomarrow so i can do that
-Original 
  Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Sunday, February 20, 2005 2:40 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: RE: [Asterisk-Users] help 
  with @home
  Can you work 
  through a process of elimination if you record the file using an internal 
  extension by dialing *77 and seeing if that 
  works?
   
   
   
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Kurt 
  FankhauserSent: Sunday, 
  February 20, 2005 7:42 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] help with 
  @home
   
  
  just reinstalled @home and i 
  have a one of those 100 cards, anyways when i call from the pstn the box 
  picks up but i hear nothing, then it clicks a couple times, then nothing 
  again, i am trying to get the digital receptionist to work but it won't 
  save my wav file to the @home box and all the radio buttons under incoming 
  calls are greyed out. the greyed out thing seems to be my biggest problem 
  right now, also do you have to use a ip phone to record your greeting 
  because this wav file stuff isn't 
  working.
  
   
  Kurt 
  Fankhauser
  WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 
  
  
   
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RE: [Asterisk-Users] help with @home

[Kurt Fankhauser]

Title: Message



i got 
the xlite phone on my pc and i have no idea how to get it working, do i have to 
add it into the * box or just change some settings on the 
softphone?

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of dean 
  collinsSent: Sunday, February 20, 2005 3:36 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] help with @home
  
  Just download a free 
  softphone and do it that way eg xten
   
   
   
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 9:18 
  PMTo: 'Asterisk Users Mailing List - Non-Commercial 
  Discussion'Subject: RE: [Asterisk-Users] help with 
  @home
   
  
  I'll buy a IP phone 
  tomarrow so i can do that
  
-Original 
Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Sunday, February 20, 2005 2:40 
PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: RE: [Asterisk-Users] help with 
@home
Can you work 
through a process of elimination if you record the file using an internal 
extension by dialing *77 and seeing if that 
works?
 
 
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt 
FankhauserSent: Sunday, 
February 20, 2005 7:42 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] help with 
@home
 

just reinstalled @home and i 
have a one of those 100 cards, anyways when i call from the pstn the box 
picks up but i hear nothing, then it clicks a couple times, then nothing 
again, i am trying to get the digital receptionist to work but it won't save 
my wav file to the @home box and all the radio buttons under incoming calls 
are greyed out. the greyed out thing seems to be my biggest problem right 
now, also do you have to use a ip phone to record your greeting because this 
wav file stuff isn't working.

 
Kurt 
Fankhauser
WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 


 
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Re: [Asterisk-Users] wiki down?

[Ed Greenberg]

--On Sunday, February 20, 2005 10:38 PM + Peter Bowyer 
<[EMAIL PROTECTED]> wrote:

On Sun, 20 Feb 2005 22:45:42 +0100, Roy Sigurd Karlsbakk
<[EMAIL PROTECTED]> wrote:
It seems to me wiki downtime is somehow regular.
Is this the fact?
If so, should it be moved?
Just to add some balance to this threadJim and colleagues, thanks
for hosting the Wiki. You should take it as a compliment that when
it's down occasionally, so many people notice.
Peter
Indeed. We rely on it as a main reference material.
Thanks, Jim, etc.
 
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[Asterisk-Users] Traditional Ringback Tone

[Greg Blakely]

 
I am trying to get Asterisk to emulate the sounds of the earlier
telephone systems, and the settings in [us-old] are pretty helpful.  The
only thing lacking is ringback tone, which is not quite as complex as
the real phone systems of the day.  For example, it is true that a
ringback tone commonly used is 420Hz modulated by 40Hz.  This is what
shows up in [us-old].  But that modulated tone was generally overlaid on
top of real ringing, i.e. 20Hz.  So, using the Asterisk example of
420*40, it would seem that a decent ringback would be (420*40)*20.  But,
of course, that doesn't appear to exist.  If it does, I am missing the
boat on how to do it properly.

So, I have a question:  Is it possible to either (a) do the double
modulation as listed above, or (b) provide recorded wav or gsm sounds as
a background fill while a phone is being rung?  I have recordings of
various types of older central office ringback tones that I'd just love
to be able to put into Asterisk.

I know this sounds a bit arcane.  But Asterisk can do so many things to
order that it really ought to be able to do this, don't you think?

Thanks,

Greg

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[Asterisk-Users] NAT and FWD

[Anton Krall]

Guys.
 
Im trying to figure out how to confgure FWD and NAT. I tried some configs
and tested thru the FWD webpage the incoming call.. I got the incoming fine
except it kept asking me for my name and seems it didnt get it.. (nat
problems, one way sound only)?
 
Also, I cant dialout.. 
 
Can somebody using NAT send me some example configs?
 
Thx!
 
__
Anton Krall
 

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RE: [Asterisk-Users] Voice Prompts with no sound

[Anton Krall]

Well.. I disabled certain modules, removed any app not supposed to be
running.. Seems the problem is solved.. For now at least hahaha

Thx to everybody that sent their comments.
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ray Brannam
Sent: Domingo, 20 de Febrero de 2005 05:55 p.m.
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Voice Prompts with no sound


On Sun, 20 Feb 2005 14:45:20 -0600 (CST)
[EMAIL PROTECTED] wrote:

> Message: 1
> Date: Sun, 20 Feb 2005 12:06:16 -0600
> From: "Anton Krall" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Voice Prompts with no sound
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>   
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="US-ASCII"
> 
> I have a weird problem... very puzzling..
>  
> Yesterday I had sound problems with the voice prompts, I couldnt hear 
> them, so I rebooted the system and voila, I was able to hear 
> everything.. so I went to bad.. and I just woke up and tried the system
again and its back!!!
> I dial the voicemail system and I cant hear the voice welcome.. I can 
> hear any voice prompts
>  
> Has anybody had this kind of problems?
>  
>  
> __
> Anton Krall
>  
Anton,
I have had this problem a couple times and in my case it was contention
forthe sound card.  I solved this by starting * first then the other
application that uses the sound device, in this case KDE. The result is I do
not have sound on this machine except for the things * uses it for.
I would love to have both, but in this case having VM prompts is better then
having system sounds.

Cheers,
Ray
--
Ray Brannam <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Mandrake & CAPI

[Craig Guy]

Or you could go to a 2.6 kernel and use the mISDN drivers.

Craig

- Original Message - 
From: "Razza" <[EMAIL PROTECTED]>
To: 
Sent: Sunday, February 20, 2005 8:00 PM
Subject: [Asterisk-Users] Mandrake & CAPI


> All,
> I have been trying to get CAPI4Linux working on my machine and being
> frank am failing miserably! I am looking for any help available, I am
> real newbie (so please be gentle) and choose to run Mandrake 9.2 as it
> appears quite friendly (or so I thought!).
>
> I have been following the guidance found at
> http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI for the
> AVM card (actually I have a BT Speedway - apparently the same thing).
>
> I guess the best approach is to detail what I have done in tandem with
> the guidance? So here we go -
>
> Type -
> # modprobe capi
>
> Great! I get no response (which is expected!), so move to step 2
> (http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install)
>
> Guidance states 'Download and install your kernel sources' - I installed
> these as part of the original installation, so I'll ignore.
>
> I download and install the CAPI driver -
> # cd /usr/src
> # wget
> ftp://ftp.avm.de/cardware/fritzcrd.pci/linux/suse.82/fcpci-suse8.2-03.11
> .02.tar.gz
> # tar -xzvf fcpci-suse8.2-03.11.02.tar.gz
> # cd fritz
> Great! Looking good!
>
> Guidance states modify the makefile in /usr/src/src.drv as follows -
> Replace -
>  CARD_PATH   = /lib/modules/`uname -r`/misc
> with  -
>  CARD_PATH   = /lib/modules/$(uname -r)/kernel/drivers/isdn/avmb1
>
> I am aware this chap is running Debian and I am running Mandrake, so
> after searching decided to modify the line as such -
>  CARD_PATH   = /lib/modules/2.4.22-10mdk/kernel/drivers/isdn/avmb1
>
> Guidance states modify the KRNLINCL lines for the correct include path -
>
> KRNLINCL= /usr/src/kernel-headers-`uname -r`/include
> #KRNLINCL= /lib/modules/`uname -r`/build/include
> #KRNLINCL= /usr/src/linux/include
>
> And modify the lines as thus -
> DEFINES = -DMODULE -D__KERNEL__ -DNDEBUG \
>  -D__$(CARD)__ -DTARGET=\"$(CARD)\"
> CCFLAGS = -c $(DEFINES) -O2 -Wall -I $(KRNLINCL)
> With -
> DEFINES = -DMODULE -DMODVERSIONS -D__KERNEL__ -DNDEBUG \
>  -D__$(CARD)__ -DTARGET=\"$(CARD)\"
> CCFLAGS = -c $(DEFINES) -march=i686 -O2 -Wall -I $(KRNLINCL) \
>-include $(KRNLINCL)/linux/modversions.h
>
> Again aware of the Debian V's Mandrake configuration, I searched the web
> and found the following guidance for Mandrake (using the google
> translation feature - http://translate.google.com/translate?hl=en
>  in.de/&prev=/search%3Fq%3Dcapi%2Bmandrake%26hl%3Den%26lr%3D%26rls%3DRNWE
> ,RNWE:2004-35,RNWE:en>
> &sl=de&u=http://ixi.thepenguin.de/&prev=/search%3Fq%3Dcapi%2Bmandrake%26
> hl%3Den%26lr%3D%26rls%3DRNWE,RNWE:2004-35,RNWE:en )
>
> And made the following changes to the makefile in /usr/src/src.drv as
> that seemed more appropriate and saved the file -
>
> KRNLINCL =/usr/src/linux/include
>
> DEFINES = Dmodule Dmodversions D__kernel __ Dndebug \
> D__$(card) __ Dtarget=\"$(card) \ "
>
> CCFLAGS = C $(defines) -march=i586 -O2 barrier i $(krnlincl) \
> include/usr/src/linux/include/linux/modversions.h
>
> Going back to the original Guidance
> (http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install)
> I am instructed to modify the defs.h file in /usr/src/fritz/src.drv as
> follows -
> #if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 0)
> with
> #if LINUX_VERSION_CODE < KERNEL_VERSION(2, 4, 23)
>
> Great, I'm now ready to run the make command! Unfortunately the first
> couple of responses are as follows which to me looks very bad? And not
> sure what to do next?
>
> [EMAIL PROTECTED] src.drv]# make
> cc C Dmodule Dmodversions D__kernel__ DNDEBUG D Dtarget=\"\"
> -march=i586 -O2 barrier i /usr/src/linux/include
> include/usr/src/linux/include/linux/modversions.h main.c -o main.o
> cc: C: No such file or directory
> cc: Dmodule: No such file or directory
> cc: Dmodversions: No such file or directory
> cc: D__kernel__: No such file or directory
> cc: DNDEBUG: No such file or directory
> cc: D: No such file or directory
> cc: Dtarget="": No such file or directory
> cc: barrier: No such file or directory
> cc: i: No such file or directory
> cc: include/usr/src/linux/include/linux/modversions.h: No such file or
> directory
>
>
> For completeness I Have included the makefile and defs.h files
>
>  Makefile 
> SOURCES  = main.c driver.c tables.c queue.c lib.c tools.c
> OBJECTS  = $(patsubst %.c,%.o,$(SOURCES))
> LIBRARY  = ../lib/$(CARD)-lib.o
>
> CARD_PATH = /lib/modules/2.4.22-10mdk/kernel/drivers/isdn/avmb1
> CS_PATH  = /lib/modules/`uname -r`/pcmcia-external
>
> KRNLINCL = /usr/src/linux/include
>
> DEFINES  = Dmodule Dmodversions D__kernel__ DNDEBUG \
> D__$(CARD)__ Dtarget=\"$(CARD)\"
> CCFLAGS  = C $(DEFINES) -

Re: [Asterisk-Users] Voice Prompts with no sound

[Ray Brannam]

On Sun, 20 Feb 2005 14:45:20 -0600 (CST)
[EMAIL PROTECTED] wrote:

> Message: 1
> Date: Sun, 20 Feb 2005 12:06:16 -0600
> From: "Anton Krall" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Voice Prompts with no sound
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>   
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="US-ASCII"
> 
> I have a weird problem... very puzzling..
>  
> Yesterday I had sound problems with the voice prompts, I couldnt hear them,
> so I rebooted the system and voila, I was able to hear everything.. so I
> went to bad.. and I just woke up and tried the system again and its back!!!
> I dial the voicemail system and I cant hear the voice welcome.. I can hear
> any voice prompts 
>  
> Has anybody had this kind of problems?
>  
>  
> __
> Anton Krall
>  
Anton,
I have had this problem a couple times and in my case it was contention
forthe sound card.  I solved this by starting * first then the other
application that uses the sound device, in this case KDE. The result is
I do not have sound on this machine except for the things * uses it for.
I would love to have both, but in this case having VM prompts is better
then having system sounds.

Cheers,
Ray
-- 
Ray Brannam <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] help with @home

[GLNX]

Remember you can also test *77 recording thru a softphone (in fact as I 
did, to submit some recordings I'm using for other usages)

Regards
Kurt Fankhauser wrote:
I'll buy a IP phone tomarrow so i can do that
-Original Message-
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*dean collins
*Sent:* Sunday, February 20, 2005 2:40 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [Asterisk-Users] help with @home
Can you work through a process of elimination if you record the
file using an internal extension by dialing *77 and seeing if that
works?
 

 

 


*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Kurt Fankhauser
*Sent:* Sunday, February 20, 2005 7:42 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] help with @home
 

just reinstalled @home and i have a one of those 100 cards,
anyways when i call from the pstn the box picks up but i hear
nothing, then it clicks a couple times, then nothing again, i am
trying to get the digital receptionist to work but it won't save
my wav file to the @home box and all the radio buttons under
incoming calls are greyed out. the greyed out thing seems to be my
biggest problem right now, also do you have to use a ip phone to
record your greeting because this wav file stuff isn't working.
 

Kurt Fankhauser
WaveLinc
www.wavelinc.com 
114 S. Walnut St.
Bucyrus, OH 44820
419-562-6405
 

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Re: [Asterisk-Users] Voice Prompts with no sound

[timebandit001]

> Yesterday I had sound problems with the voice prompts, I couldnt hear them,
> so I rebooted the system and voila, I was able to hear everything.. so I
> went to bad.. and I just woke up and tried the system again and its back!!!
> I dial the voicemail system and I cant hear the voice welcome.. I can hear
> any voice prompts
> 
> Has anybody had this kind of problems?
Only thing I can see is that you have a codec problem. Maybe you are
allowing a codec that isn't supported by your phone.

Maybe I'm wrong, but at least check that. Get on the console and see
what codec the call is being handled with

hth
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Re: [Asterisk-Users] Conecting to asterisk server through NAT using IAX

[timebandit001]

> I use linksys router.
> Now, I am trying to connect from outside to my asterisk server.
> I use Diax as iax client.
> For some reason I cannot connect to my server from outside.
> On my router I forward those ports to my asterisk server.
> 5060-5063
> 1-2
> 5036
> 4569
For IAX, only port you have to forward is 4569 UDP

Notice the UDP, not TCP

I'm using Linksys WRT54G and it works without a hitch.

> It works ok with broadvoice, but clinets cannot connect to the server.
> This is my iax.conf file
> [general]
> port=5036

well, here's your problem, port=5036. This is not the standard IAX
port. comment that line or replace it with port=4569

hth
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RE: [Asterisk-Users] help with @home

[dean collins]

Title: Message








Just download a free softphone and do it
that way eg xten

 

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt Fankhauser
Sent: Sunday, February 20, 2005
9:18 PM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] help
with @home



 



I'll buy a IP phone tomarrow so i can do
that





-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean collins
Sent: Sunday, February 20, 2005
2:40 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] help
with @home

Can you work through a process of
elimination if you record the file using an internal extension by dialing *77
and seeing if that works?

 

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt Fankhauser
Sent: Sunday, February 20, 2005
7:42 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] help
with @home



 



just reinstalled @home and i have a one of those 100 cards,
anyways when i call from the pstn the box picks up but i hear nothing, then it
clicks a couple times, then nothing again, i am trying to get the digital
receptionist to work but it won't save my wav file to the @home box and all the
radio buttons under incoming calls are greyed out. the greyed out thing seems
to be my biggest problem right now, also do you have to use a ip phone to
record your greeting because this wav file stuff isn't working.





 



Kurt Fankhauser

WaveLinc
www.wavelinc.com
114 S. Walnut St.
Bucyrus, OH
 44820
419-562-6405 



 










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Re: [Asterisk-Users] help with @home

[Erwin de Raad]

Message> I'll buy a IP phone tomarrow so i can do that

No need:
http://www.xten.net/index.php?menu=products&smenu=download

Regards,
Erwin
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Re: [Asterisk-Users] help with @home

[Erwin de Raad]

Message> also do you have to use a ip phone to record your greeting because
this wav file stuff isn't working.

I didn't try uploading. You can just setup a SIP softphone and dial *77 when
looking at the menu you want to record in the GUI.

Regards,
Erwin

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RE: [Asterisk-Users] help with @home

[Kurt Fankhauser]

I think the box is answering calls but I don't think the digital
receptionist is working properly.

Kurt

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Thompson
Sent: Sunday, February 20, 2005 3:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] help with @home


Kurt Fankhauser wrote:
> just reinstalled @home and i have a one of those 100 cards, anyways
when 
> i call from the pstn the box picks up but i hear nothing, then it
clicks 
> a couple times, then nothing again, i am trying to get the digital 
> receptionist to work but it won't save my wav file to the @home box
and 
> all the radio buttons under incoming calls are greyed out. the greyed 
> out thing seems to be my biggest problem right now, also do you have
to 
> use a ip phone to record your greeting because this wav file stuff
isn't 
> working.

Are you logged into the console while your testing the dialing in? What 
messages are you seeing?

If asterisk is already running in the background, do a "asterisk -r" 
before you start to dial in.

If there is some other interface in the @home distribution for 
monitoring asterisk, you'll have to say what app you're using and what 
you're seeing.

At any rate, without log, error, or console messages there's not alot we

can do for you.


-- 
Andrew Thompson
http://aktzero.com/
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RE: [Asterisk-Users] help with @home

[Kurt Fankhauser]

Title: Message



I'll 
buy a IP phone tomarrow so i can do that

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of dean 
  collinsSent: Sunday, February 20, 2005 2:40 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] help with @home
  
  Can you work through 
  a process of elimination if you record the file using an internal extension by 
  dialing *77 and seeing if that works?
   
   
   
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 7:42 
  PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] help with 
  @home
   
  
  just reinstalled @home and i have 
  a one of those 100 cards, anyways when i call from the pstn the box picks up 
  but i hear nothing, then it clicks a couple times, then nothing again, i am 
  trying to get the digital receptionist to work but it won't save my wav file 
  to the @home box and all the radio buttons under incoming calls are greyed 
  out. the greyed out thing seems to be my biggest problem right now, also do 
  you have to use a ip phone to record your greeting because this wav file stuff 
  isn't working.
  
   
  Kurt 
  Fankhauser
  WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 
  
  
   
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RE: [Asterisk-Users] No Sounds; stumping "The Tryant" ; Possible heat problem

[Anton Krall]

Just to be sure.. I checked the cards...   

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 04:18 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" ;Possible
heat problem

Okay, now you are getting off track.

Hold old is the motherboard? 
How big is the case?
How big is the power supply?

If it is a smaller case and server then sometimes heat can be an issue when
you are on the threshold of the temperature limit. Things will work
mysteriously and then not work.

Make sure all your cards in their slots tightly and screwed down.

Try running with the case cover off the server. If it then runs fine, you
have an overheating problem.

As an example my eth1 PCI network card was failing intermittently. Turns out
it was really the Sound Blaster card that was loose and causing problems.
Took the sound blaster out and eth1 is solid as a rock.

Race Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Sunday, February 20, 2005 5:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

It is weird.. I did a full asterisk reinstall... (no asterisk at home
now)... And well Problem persists but this is weird, when it happens, I
reboot the machine, starts working again and sometimes sound stops,
sometimes it doesn’t... This machine seems to have an attitude :)

Last reboot one of the x100p cards complained during the modprobe wcfxo...
:) then it didn’t :) 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 02:07 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

Yes, running extra code/libraries/.so means more variables, which we are
trying to eliminate.

Really Anton I am stumped.

Does anyone know? Do you have to have the gsm codec to hear the .gsm sound
files.

Is there an [EMAIL PROTECTED] mailing list?

I found these:
http://www.uninett.no/voip/asterisk.html
No sound on SIP
I had a "allow=all" codecs in the 'sip.conf' while which sort of "stopped"
all sound out. I commented it out, and it was up and running on the sound.
Now I just allow for the g.711 codec with

disallow=all
allow=ulaw 





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Sunday, February 20, 2005 2:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

Ok Noload modems, alsa and oss... No errors... Is this ok? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Domingo, 20 de Febrero de 2005 01:18 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

I don’t know if it has something to do but I see 2 mpg123 processes
running:

 3552 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 3553 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 
Everytime I start asterisk..

Also, if I enable alsa I get this error:

 [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301
__load_resource:
/usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file:
No
such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459
load_modules: Loading module chan_alsa.so failed!
[EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested!
Ouch ... error while writing audio data: : Broken pipe

And asterisk quits... 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 12:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

This is getting hard.

So what do we have?

1. The Asterisk server and the phones are using good CODECS.
2. Sound is moving from phone to phone.
3. Sound from the prompts is not playing back to the phones.

So let's go with the first principle: Eliminate the variables.

Do this:

Extensions.conf

[NoSound]

exten => 222,1,Wait(2)
exten => 222,2,Answer
exten => 222,3,Playback(vm-isunavail)
exten => 222,4,Hangup

And see what happens.

I might be missing something. Anyone know how .gsm files are translated to
ulaw/alaw in asterisk?

Race "The Tyrant" Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 10:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Di

[Asterisk-Users] HFC-S ISDN card on *@home

[Erwin de Raad]

It seems so simple, but I'm having no luck installing a HFC-s ISDN BRI card
on [EMAIL PROTECTED] 0.5.
I probably have to install BRI-stuff from Junghanns.net but that also
downloads and installs another copy of * from Digium.

I'm not sure if zaphfc has to be installed *before* Asterisk or if it's OK
to do this afterwards.

I've seen this question before, but: Anyone successfully installed a HFC-s
card on [EMAIL PROTECTED] Please post the steps you had to take. I'm sure quite 
some
list-members are interested!

With kind regards
Erwin
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Re: [Asterisk-Users] help with @home

[Andrew Thompson]

Kurt Fankhauser wrote:
just reinstalled @home and i have a one of those 100 cards, anyways when 
i call from the pstn the box picks up but i hear nothing, then it clicks 
a couple times, then nothing again, i am trying to get the digital 
receptionist to work but it won't save my wav file to the @home box and 
all the radio buttons under incoming calls are greyed out. the greyed 
out thing seems to be my biggest problem right now, also do you have to 
use a ip phone to record your greeting because this wav file stuff isn't 
working.
Are you logged into the console while your testing the dialing in? What 
messages are you seeing?

If asterisk is already running in the background, do a "asterisk -r" 
before you start to dial in.

If there is some other interface in the @home distribution for 
monitoring asterisk, you'll have to say what app you're using and what 
you're seeing.

At any rate, without log, error, or console messages there's not alot we 
can do for you.

--
Andrew Thompson
http://aktzero.com/
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Re: [Asterisk-Users] wiki down?

[Peter Bowyer]

On Sun, 20 Feb 2005 22:45:42 +0100, Roy Sigurd Karlsbakk
<[EMAIL PROTECTED]> wrote:
> It seems to me wiki downtime is somehow regular.
> Is this the fact?
> If so, should it be moved?

Just to add some balance to this threadJim and colleagues, thanks
for hosting the Wiki. You should take it as a compliment that when
it's down occasionally, so many people notice.

Peter

-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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RE: [Asterisk-Users] help with @home

[dean collins]

Title: Message








Can you work through a process of
elimination if you record the file using an internal extension by dialing *77
and seeing if that works?

 

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt Fankhauser
Sent: Sunday, February 20, 2005
7:42 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] help
with @home



 



just reinstalled @home and i have a one of those 100 cards,
anyways when i call from the pstn the box picks up but i hear nothing, then it
clicks a couple times, then nothing again, i am trying to get the digital
receptionist to work but it won't save my wav file to the @home box and all the
radio buttons under incoming calls are greyed out. the greyed out thing seems
to be my biggest problem right now, also do you have to use a ip phone to
record your greeting because this wav file stuff isn't working.





 



Kurt Fankhauser

WaveLinc
www.wavelinc.com
114 S. Walnut St.
Bucyrus, OH
 44820
419-562-6405 



 








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[Asterisk-Users] Sparc hardware, Linux and X100P REVISITED

[Robert Burcham]

I was studying the asterisk-users list archives
to learn if anyone has had success with an X100P on a
sparc. I noticed some postings on the subject.  I am
wondering if anyone has learned anything new?

I have an Ultra-60 running Gentoo with 2.6.10 and
udev. I built * 1.0.5 and have been enjoying
various SIP configurations, with 2 sipura phones and 2
UIP200 phones (got them working!) in my home, bridging
in FWD and now Voiptalk too.

I bought 2 X100P clones via ebay, and put one in my
U60. I can see it with lspci:

0001:00:02.0 Communication controller: Tiger Jet
Network Inc. Tiger3XX Modem/ISDN interface

I successfully built zaptel, and can modprobe zaptel
and wcfxo:

# lsmod
Module  Size  Used by
wcfxo  14680  0
zaptel195424  1 wcfxo
crc_ccitt   2752  1 zaptel

However I can't ztcfg with any success:

# ztcfg -v

Zaptel Configuration
==


1 channels configured.

ZT_CHANCONFIG failed on channel 1: Invalid argument
(22)
Did you forget that FXS interfaces are configured with
FXO signalling
and that FXO interfaces use FXS signalling?


In fact dmesg never shows wcfxo completely setting
up the card:

# dmesg

Zapata Telephony Interface Registered on major 196
wcfxo: DAA mode is 'FCC'
Found a Wildcard FXO: Generic Clone
PCI Target abort
PCI Target abort
PCI Target abort

I have to think that the PCI driver does not get along
with the wcfxo driver for the X100P clone. Also,
strange things happen while the driver is loaded
too...
consoles dropping, etc.

As for pure SIP related functions, * 1.0.5 on sparc
has performed very well, with all functions (moh, vm,
xfer, call park, etc) working admirably.

Has anyone been able to get any farther than I have
with the X100P on a sparc?

Rob



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Re: [Asterisk-Users] Detecting if a call is active on chan_sip before trying INVITE? (was Sip question - allow only 1 incoming call to sip phone)

[Kevin P. Fleming]

Roy Sigurd Karlsbakk wrote:
Is it possible to test if a call to SIP/xxx is in place before dialling 
out? This could help a lot to centralize administation of whether or not 
to use call waiting instead of configuring the ATAs.
app_groupcount can be used to provide call counting in any fashion you 
desire.
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RE: [Asterisk-Users] No Sounds; stumping "The Tryant" ; Possible heat problem

[Race Vanderdecken]

Okay, now you are getting off track.

Hold old is the motherboard? 
How big is the case?
How big is the power supply?

If it is a smaller case and server then sometimes heat can be an issue
when you are on the threshold of the temperature limit. Things will work
mysteriously and then not work.

Make sure all your cards in their slots tightly and screwed down.

Try running with the case cover off the server. If it then runs fine,
you have an overheating problem.

As an example my eth1 PCI network card was failing intermittently. Turns
out it was really the Sound Blaster card that was loose and causing
problems. Took the sound blaster out and eth1 is solid as a rock.

Race Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Sunday, February 20, 2005 5:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

It is weird.. I did a full asterisk reinstall... (no asterisk at home
now)... And well Problem persists but this is weird, when it
happens, I
reboot the machine, starts working again and sometimes sound stops,
sometimes it doesn’t... This machine seems to have an attitude :)

Last reboot one of the x100p cards complained during the modprobe
wcfxo...
:) then it didn’t :) 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 02:07 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

Yes, running extra code/libraries/.so means more variables, which we are
trying to eliminate.

Really Anton I am stumped.

Does anyone know? Do you have to have the gsm codec to hear the .gsm
sound
files.

Is there an [EMAIL PROTECTED] mailing list?

I found these:
http://www.uninett.no/voip/asterisk.html
No sound on SIP
I had a "allow=all" codecs in the 'sip.conf' while which sort of
"stopped"
all sound out. I commented it out, and it was up and running on the
sound.
Now I just allow for the g.711 codec with

disallow=all
allow=ulaw 





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Sunday, February 20, 2005 2:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

Ok Noload modems, alsa and oss... No errors... Is this ok? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Domingo, 20 de Febrero de 2005 01:18 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

I don’t know if it has something to do but I see 2 mpg123 processes
running:

 3552 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 3553 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 
Everytime I start asterisk..

Also, if I enable alsa I get this error:

 [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301
__load_resource:
/usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file:
No
such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459
load_modules: Loading module chan_alsa.so failed!
[EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested!
Ouch ... error while writing audio data: : Broken pipe

And asterisk quits... 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 12:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

This is getting hard.

So what do we have?

1. The Asterisk server and the phones are using good CODECS.
2. Sound is moving from phone to phone.
3. Sound from the prompts is not playing back to the phones.

So let's go with the first principle: Eliminate the variables.

Do this:

Extensions.conf

[NoSound]

exten => 222,1,Wait(2)
exten => 222,2,Answer
exten => 222,3,Playback(vm-isunavail)
exten => 222,4,Hangup

And see what happens.

I might be missing something. Anyone know how .gsm files are translated
to
ulaw/alaw in asterisk?

Race "The Tyrant" Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, February 19, 2005 10:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Race.

Here are thre results of the tests:

Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
-
0x0 (nothing) 

Seems both can speak ulaw 

Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}

[Steven Critchfield]

On Sun, 2005-02-20 at 22:59 +0100, Julius Schwartzenberg wrote:

> I'm using a pretty old system and I have good experiences with Slackware 
>   on other systems. Here are the specs of the system I'm using:
> IBM/Cyrix PR-200 (@150MHz), 64MB RAM, two HDs which are combined ~2GB.

Here is your trouble. The Cyrix chip is what is the newer Via chipsets
are based on. It isn't a real pentium chipset and needs to get tuned
down via the CFLAGS to 586 or lower. 

You will probably hit the limits of that machine really quickly. You may
want to find a slightly better machine for testing.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] No Sounds; stumping "The Tryant"

[Anton Krall]

It is weird.. I did a full asterisk reinstall... (no asterisk at home
now)... And well Problem persists but this is weird, when it happens, I
reboot the machine, starts working again and sometimes sound stops,
sometimes it doesn’t... This machine seems to have an attitude :)

Last reboot one of the x100p cards complained during the modprobe wcfxo...
:) then it didn’t :) 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 02:07 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

Yes, running extra code/libraries/.so means more variables, which we are
trying to eliminate.

Really Anton I am stumped.

Does anyone know? Do you have to have the gsm codec to hear the .gsm sound
files.

Is there an [EMAIL PROTECTED] mailing list?

I found these:
http://www.uninett.no/voip/asterisk.html
No sound on SIP
I had a "allow=all" codecs in the 'sip.conf' while which sort of "stopped"
all sound out. I commented it out, and it was up and running on the sound.
Now I just allow for the g.711 codec with

disallow=all
allow=ulaw 





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Sunday, February 20, 2005 2:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

Ok Noload modems, alsa and oss... No errors... Is this ok? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Domingo, 20 de Febrero de 2005 01:18 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

I don’t know if it has something to do but I see 2 mpg123 processes
running:

 3552 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 3553 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 
Everytime I start asterisk..

Also, if I enable alsa I get this error:

 [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301
__load_resource:
/usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file:
No
such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459
load_modules: Loading module chan_alsa.so failed!
[EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested!
Ouch ... error while writing audio data: : Broken pipe

And asterisk quits... 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 12:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

This is getting hard.

So what do we have?

1. The Asterisk server and the phones are using good CODECS.
2. Sound is moving from phone to phone.
3. Sound from the prompts is not playing back to the phones.

So let's go with the first principle: Eliminate the variables.

Do this:

Extensions.conf

[NoSound]

exten => 222,1,Wait(2)
exten => 222,2,Answer
exten => 222,3,Playback(vm-isunavail)
exten => 222,4,Hangup

And see what happens.

I might be missing something. Anyone know how .gsm files are translated to
ulaw/alaw in asterisk?

Race "The Tyrant" Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 10:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Race.

Here are thre results of the tests:

Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
-
0x0 (nothing) 

Seems both can speak ulaw and ilbc but still, no prompt on the phone :(

Any ideas? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 07:15 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Correct.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

This is a very good place to start Race. So if I understand you correctly,
Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw
might help so I can listen to the prompts right?

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 06:55 p.m.
To: 'Asterisk Users Mailing

Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}

[Julius Schwartzenberg]

Thanks a lot for your message.
Race Vanderdecken schreef:
Ouch,
Do you know how to use gdb, the Gnu Debugger?
That will give you a clue as to where the segmentation fault is coming
from.
No, I once used it being instructed exactly by a developer to solve a 
problem in Dosemu, but I never did anything else with it.
I understand that I need to recompile Asterisk with debugging support. 
Could you give me some pointers on what to do next?

Good, then let me move on to the insults and ranting.
1. Why are you running on Slackware? 
	Are you trying to prove a point or just enjoy being frustrated?
	Open Source is like "Broad Spectrum Pesticide", it works but
your results may vary and you may end up killing your lawn.
I'm using a pretty old system and I have good experiences with Slackware 
 on other systems. Here are the specs of the system I'm using:
IBM/Cyrix PR-200 (@150MHz), 64MB RAM, two HDs which are combined ~2GB.

2. The dearth of information of value in your posting is amazing.
I went to http://www.automated.it/guidetoasterisk.htm (a good
start, good effort Mr. Powell.) As stated above, you life might be
easier using FEDORA, not an endorsement of Red Hat, rather a plea for a
unified Linux base (please don't say Debian, self-installing the
micro-chip in my head was easier.) (it is the new Anti-AMD Intel Rantino
chip for those interested.)
I've never used Fedora on older systems, but I thought it wouldn't run 
very well on the system I'm using.
(Good thing you don't have an Anti-IBM chip ;)
 	
3. "I've never installed or used Asterisk before, so I do not know much
about it."
	1. What is your goal with installing Asterisk? 
We have about 8 telephones that use the plain telephone system to call 
each other and externally. Some of them are analog and others are 
digital (ISDN). I've also still got the old ISDN card from before we had 
ADSL. (Eicon Diva 2.01 ISA, seems to work with the hisax module.)
Since I read that Asterisk worked with any ISDN adapter that was 
supported by ISDN4Linux, I thought it might be possible to hook it up in 
such a way that the phones could call the Asterisk system and that 
Asterisk would forward the call to a computer (and maybe even over the 
internet). Also the other way around would be neat.

	2. Do you have Digium or other hardware installed?
No. Only the ISDN adapter.
	3. Are you running SIP/H323/MGCP?
No. I've experimented with SIP before, but only with a softphone, using 
an account from SIPPhone.com. It would be nice if I could call my 
Asterisk system using SIP!

	4. Did you modify any files?
None from Asterisk.
4. What was the last thing on the *CLI> screen before the seg fault?	
The command to run Asterisk. It immediatly gives the error when I try to 
run it.

Is Asterisk able to do, what I thought it would do or am I just messing?
Come on Mr. Caesar throw us a bone here.
All Hail,
Race "The Tyrant" Vanderdecken
Thanks again,
Julius
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[Asterisk-Users] Conecting to asterisk server through NAT using IAX

[Bartosz Wegrzyn - asterisk]

Hello,

I have asterisk setup with Broadvoice.
It works great as PBX and Outgoing calling server for all local clients
withing 192.168.1.0 network. My asterisk is running over NAT.
I use linksys router.
Now, I am trying to connect from outside to my asterisk server.
I use Diax as iax client.
For some reason I cannot connect to my server from outside.
On my router I forward those ports to my asterisk server.
5060-5063
1-2
5036
4569

It works ok with broadvoice, but clinets cannot connect to the server.
This is my iax.conf file
[general]
port=5036
tos=lowdelay
jitterbuffer=no
disallow=all
allow=ulaw
allow=ilbc
allow=gsm
allow=adpcm
allow=alaw

register => xxx:[EMAIL PROTECTED]

[guest]
type=user
context=abcxyz
auth=none

[voicepulse-in-01] ; <-- Name must be [voicepulse-in-01]
type=user
context=voicepulse-incoming ; <-- Should match the context you
   ; are using in extensions.conf
auth=rsa
inkeys=voicepulse01

[tester]
type=friend
context=sip
auth=plaintext
secret=secrwt
host=dynamic
allow=all
nat =1

Clients cannot connect to asterisk. WHY???
Am I doing something wrong?

Please help.

Thanks

Bart


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[Asterisk-Users] Detecting if a call is active on chan_sip before trying INVITE? (was Sip question - allow only 1 incoming call to sip phone)

[Roy Sigurd Karlsbakk]

Disable the call waiting feature in the phone, so it will signal "486 
- Busy here" to additionally incoming calls.
Is it possible to test if a call to SIP/xxx is in place before dialling 
out? This could help a lot to centralize administation of whether or 
not to use call waiting instead of configuring the ATAs.

roy
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Re: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4 C.O. lines

[Sergey Kuznetsov]

Yesterday, I've checked tariffs from Bell Canada, For Full voice T1 it
was costs around $1000 + tax.

$216 - is access fee, $34 per channel.

You can get the PRIs from Allstream with 3 years commitment ~$600 per
month.

Andrew Kohlsmith wrote:

  On February 20, 2005 11:44 am, Jim Van Meggelen wrote:
  
  
I like the thinking; the challenge is often where in the world you are,
and how much competition there is. Here in Ontario, T1's were generally
priced such that fractional T1s hardly saved anything. There is more
competition now, so prices are changing, but I still can't see frac T1
service competing with such a small number of analog circuits. I know
there are places where such a thing could be had very competitively, so
your advice is still good.

  
  
I think you'd be surprised.  Even in Listowel a CT1 for POTS termination was 
on-par with having the individual analogue lines brought out.  You'll pay a 
little more for the smartjack lease but it eliminates a lot of headaches.

Hell the PRI here in cow-town Listowel was in-line with POTS until you 
included the D channel price of $500 -- The B chans were all $55/mo which is 
exactly what a business line costs.  I imagine CT1 instead of PRI service 
would have been significantly cheaper, *AND* I wouldn't have to pay for all 
those extra DIDs.

-A.
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-- 
All the Best!
Sergey.
=
Sergey Kuznetsov
President/CEO
 High Intellectual Technologies, Inc.

   Web: http://www.hitcalls.com
E-mail: [EMAIL PROTECTED]
Business phone: (416) 548-9700
  Mobile phone: (647) 287-8448


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Re: [Asterisk-Users] wiki down?

[Roy Sigurd Karlsbakk]

It seems to me wiki downtime is somehow regular.
Is this the fact?
If so, should it be moved?
roy
On Feb 19, 2005, at 10:02 PM, James H. Thompson wrote:
Wiki is back up.
Between comment SPAM storms, over eager robots ignoring robots.txt, 
and mysql issues, it has been an interesting week.
 
 
Jim
 
James H. Thompson
[EMAIL PROTECTED]
[EMAIL PROTECTED]

 
- Original Message -
 From: Roy Sigurd Karlsbakk
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Saturday, February 19, 2005 8:13 AM
Subject: [Asterisk-Users] wiki down?
hi
is the wiki down again?
roy
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[Asterisk-Users] help with @home

[Kurt Fankhauser]

Title: Message



just reinstalled 
@home and i have a one of those 100 cards, anyways when i call from the pstn the 
box picks up but i hear nothing, then it clicks a couple times, then nothing 
again, i am trying to get the digital receptionist to work but it won't save my 
wav file to the @home box and all the radio buttons under incoming calls are 
greyed out. the greyed out thing seems to be my biggest problem right now, also 
do you have to use a ip phone to record your greeting because this wav file 
stuff isn't working.
 
Kurt Fankhauser
WaveLincwww.wavelinc.com114 S. Walnut 
St.Bucyrus, OH 44820419-562-6405 
 
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RE: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> I have no problem with Slackware,

Me neither. I learned Linux with Slack. Found it to be extremely
friendly. And that was 10 years ago. Last time I chacked, it was still
friendly (and not at all GUI, unless you want it served that way)

> But when you are learning to drive a car you should first try
> a Chevy with an automatic transmission first before strapping
> on a 6 speed Ferrari.

Popular opinion holds that people who learn to drive standard first
generally end up being better drivers. And why wouldn't you want to
learn on a Ferrari since you can get one for free!?!

> Humor helps in teaching and getting a person to step out of a
> rut they are having a problem in and gives them a chance to
> rethink what might be going on.

Ya, but humour should be dispensed carefully, lest offence be given.

> Remember, my goal is to reduce the number of variables in the system.

The problem I see with Fedora is that you can install it successfully
without learning anything about Linux. Slackware is rather good for
learning Linux, because it is friendly and helpful, but still expects
you to make the decisions. I'd argue that a familiarity with the shell
is going to be essential for even a basic Asterisk install. It's not a
pre-qualifier so much as an essential skill.

LOL! You're just bored and are trolling for a holy war, eh? Well, I
guess we gotta shake off these Febraury blah's somehow.

GENTOO IS FOR WANNABE NEWBIES!!! (that oughta stir things up)


--
Jim Van Meggelen
[EMAIL PROTECTED]


> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Andrew Kohlsmith
> Sent: Sunday, February 20, 2005 1:39 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [Asterisk-Users] Segmentation fault {Writer
> given gnu-lashing}
> 
> On February 20, 2005 01:11 pm, Race Vanderdecken wrote:
>> 1. Why are you running on Slackware?
>>  Are you trying to prove a point or just enjoy being frustrated? 
>> Open Source is like "Broad Spectrum Pesticide", it works but your
>> results may vary and you may end up killing your lawn.
> 
> Got a problem with Slackware?  It works *very* well with Asterisk.

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005
 

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RE: [Asterisk-Users] Snom phone hint exten question

[James Bean]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jon Radon
> Sent: Monday, 21 February 2005 2:55 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Snom phone hint exten question
> 
> I haven't used it in a while, but I had to put 
> subscribecontext=sip for the phone's (in your case the snom) 
> sip entry.
> 
> This seems like it has been removed from the wiki.  Has it 
> changed or is this incorrect?
> 
> http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+ph
> one+snom&diff=7
> 
> 
> On Sat, 19 Feb 2005 21:36:04 +1000, James Bean 
> <[EMAIL PROTECTED]> wrote:
> > Putting bt-karen in the destination of the snom doesn't work, i.e.
> > pushing the button the phone says no such destination.
> > 
> > exten => 691,hint,SIP/bt-karen
> > exten => 691,1,SetMusicOnHold(random)
> > exten => 691,2,Dial(SIP/bt-karen,30,tr) exten => 
> 691,10,voicemail,u691
> > 
> > Is in the extensions.conf but in the snom I have destination as 691.
> > 
> > In the sip.conf it is setup as
> > 
> > [bt-karen]
> > type=friend
> > secret=
> > host=dynamic
> > callerid="Karen Colomb" <691>
> > defaultip=192.168.69.251
> > dtmfmode=info
> > mailbox=691
> > 
> > Hope this helps.
> > 
> > James
> 
> 
> --
> Is it something someone said, was it something someone said?
> ___
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> 
> 

Thanks for the link, it had some very userful information in it,
unforunately the lights on my snom are still dead as a door nail.

Ok the snom phone has one of its LED's set to Destination 691 (it
changes that into the sip address and it dials the extension when I hit
the button on the snom no problems, and the led works)

Does anyone know where I have gone wrong.

Configurations I have enabled are voicemail and call parking.

My sip.conf is

[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = all
allow = ilbc
allow = alaw
allow = ulaw
nat=disable
srvlookup=no
localnet=192.168.69.0/255.255.255.0
subscribecontext = sip

[snom-james]
type=friend
secret=
host=dynamic
callerid="James Bean" <690>
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=690

[bt-karen]
type=friend
secret=
host=dynamic
callerid="Karen Colomb" <691>
defaultip=192.168.69.251
dtmfmode=info
mailbox=691

My extensions.conf is

[pstn]

exten => s,hint,SIP/bt-karen
exten => s,1,SetMusicOnHold(random)
exten => s,2,Dial(SIP/snom-james&SIP/bt-karen,45,t) 
exten => s,4,VoiceMail(u690) 
exten => s,5,Hangup

[internal]

exten => i,1,Playback(invalid)
exten => i,2,Hangup
exten => t,1,Hangup

exten => 098,1,WaitMusicOnHold(45)
exten => 099,1,Echo ;simple echo test when you dial 099 on your
phone

exten => 1690,1,VoicemailMain,s690
exten => 1691,1,VoicemailMain,s691

[outgoing]

exten => _9X.,hint,SIP/bt-karen
exten => _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten => _9X.,2,Congestion()
exten => _9X.,3,Hangup

[sip]

exten => 690,hint,SIP/snom-james
exten => 690,1,SetMusicOnHold(random)
exten => 690,2,Dial(SIP/snom-james,30,Ttr)
exten => 690,3,Voicemail,u690
exten => 690,103,Voicemail,b690

exten => 691,hint,SIP/bt-karen
exten => 691,1,SetMusicOnHold(random)
exten => 691,2,Dial(SIP/bt-karen,30,Ttr)
exten => 691,3,Voicemail,u691
exten => 691,103,Voicemail,b691

include => internal
include => outgoing
include => parkedcalls
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Re: [Asterisk-Users] Adtran Total Access MGCP Config?

[Leo Ann Boon]

Dave Weis wrote:

I've never set up an mgcp device before. I have an Adtran IAD with the 
MGCP firmware on it. I have it configured in mgcp.conf like this:

[general]
port = 2427
bindaddr = 0.0.0.0
[adtran]
host = 192.168.2.2
context = default
canreinvite = no
line => aaln/1
line => aaln/2
Check that the name adtran can be resolved by your DNS or /etc/hosts. 
Otherwise just put in the IP address.

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RE: [Asterisk-Users] Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11?

[Jim Van Meggelen]

Title: Message



Just plug 
it in. The RJ11 is narrower than the RJ48, but has the exact same connection 
mechanism. it'll fit perfectly (the centre two pins are the 
contacts)
 

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  [EMAIL PROTECTED]Sent: February 20, 2005 1:51 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionCc: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] Digium TDM400P has RJ45 interface,how to connect to analog 
  phone RJ11?
  Hello,
   I bought a TDM400P, and intended to use it with my analog phone, 
  which is RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone 
  to the TDM400P card, which has an RJ45 interface? Also, since it's an 11B 
  card, I also intend to bring in an analog line into the RJ45, so i am still 
  left with the same questionhow do I go from the RJ11 standard analog to 
  the RJ45 on the TDM400P card? I'd appreciate any response.
   
  thx
  chuks
  --No virus found in this incoming message.Checked by 
  AVG Anti-Virus.Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 
  18/02/2005
  


--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005
 
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RE: [Asterisk-Users] A bit of a survey: What do do if you needmorethan 4 C.O. lines

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> On Sun, 2005-02-20 at 11:47 -0500, Jim Van Meggelen wrote:
>> Well, I appreciate everyone's input, and I'll give the matter some
>> more thought. 
>> 
>> Just so no one stays up at night worrying, this is not a situation I
>> am facing, it is simply a hypothetical scenario.
>> 
>> As with so many things, there is always a trade-off between economy
>> and functionality. The Adit 600 and T1 integration is certainly
>> quality, but I have not been able find an economical way to do this
>> (purchasing used equipment on eBay is fine for smaller deployments
>> and lab gear, but not a very sound logistics strategy, and awfully
>> difficult to explain to a customer).
> 
> This would be one of those cases where you keep a couple in
> stock and watch the ebay auctions when your stock goes low.
> You will find that your customers that are looking for the
> cheapest solutions possible will not baulk at used equipment.
> It is highly likely that they will price you against a used key
> system or pbx. 

Certainly keeping spares in stock is good advice, and I don't mind using
pre-owned equipment if it's solid stuff (which I know the adit is). I'm
going to think about this some.

As for price, that's always the challenge. Thing is, the lowest price
does not always win. Still, being able to keep costs low is always going
to be an advantage.


--
Jim Van Meggelen
[EMAIL PROTECTED]


-- 
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Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005
 

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[Asterisk-Users] possible attack, or just dumb log question?

[RJ]

I've got a strange situation that started yesterday -- I have a ton
of calls listed in the log for number = 18883629704

It initially looked like I was getting an incoming call on Zap/4 (LD
trunk) from 18883629704, which was going to an extension at Zap/2, 
and then trying to dial out again back to the 18883629704 number 
(the 'dial' application was called, with the argument 
Zap/4/18883629704).

I found one reference to on Google to this number under the topic 
"New ECM technique", describing what looks like some
kind 
of attack on some unknown system (the domain is down, but it
was in the google cache)..

The outgoing attempts weren't working (apparently because they
were coming in on the same trunk that's used for LD outgoing),
but it was still disconcerting...

So I tried to block receiving any calls from 18883629704 in
the dialplan by giving them the congestion application, and also
blocking outgoing calls to it the same way, as

exten => s/3202594099,1,Congestion  
exten => s/8883629704,1,Congestion
exten => s,1,Answer()
exten => s,2,NoOp(INCOMING call at ${DATETIME} from ${CALLERID}:
Name: ${CALLERIDNAME}, Number: ${CALLERIDNUM})
exten => s,3,DigitTimeout(10)   
exten => s,4,ResponseTimeout(20)    
exten => s,5,Background(splash)
...

and 

exten => 18883629704,1,Hangup()

in the [outgoing] context. 

But I'm still getting these things, every 45 minutes or so, in pairs
about a minute or so apart.  At least now they're not trying to dial
out, and the hangup seems to be working, but why is there all
this activity?  And why am I getting the incoming digits that it's 
trying to dial?  It looks like they're not getting the congestion thing
at all?

I put the logging into verbose debug mode, and got the following,
which doesn't make a lot of sense.  Shouldn't there be a log
entry for the Zap/4 (incoming trunk) call before it gets rung to the 
Zap/2 (station) extension?  

Thanks in advance for any help!

rj


2005-02-20 14:05:46 DEBUG[28229]: Monitor doohicky got event
Ring/Answered on channel 2
2005-02-20 14:05:46 DEBUG[28229]: Device 'Zap/2' changed to state '2'
2005-02-20 14:05:46 DEBUG[28229]: Device 'Zap/2' changed to state '2'
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 1 on Zap/2-1
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 3 on Zap/2-1
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 6 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: DTMF digit: 2 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: DTMF digit: 9 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: DTMF digit: 7 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: DTMF digit: 0 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: DTMF digit: 4 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: Enabled echo cancellation on channel 2
2005-02-20 14:05:49 DEBUG[28229]: Launching 'Hangup'
2005-02-20 14:05:49 DEBUG[28229]: Spawn extension
(default,18883629704,1) exited non-zero on 'Zap/2-1'
2005-02-20 14:05:49 DEBUG[28229]: Hanging up channel 'Zap/2-1'
2005-02-20 14:05:49 DEBUG[28229]: zt_hangup(Zap/2-1)
2005-02-20 14:05:49 DEBUG[28229]: Hangup: channel: 2 index = 0, normal
= 16, callwait = -1, thirdcall = -1
2005-02-20 14:05:49 DEBUG[28229]: disabled echo cancellation on channel
2
2005-02-20 14:05:49 DEBUG[28229]: Set option TDD MODE, value: OFF(0) on
Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: Updated conferencing on 2, with 0
conference users
2005-02-20 14:05:49 DEBUG[28229]: Device 'Zap/2' changed to state '0'
2005-02-20 14:05:49 DEBUG[28229]: Device 'Zap/2' changed to state '0'
2005-02-20 14:05:50 DEBUG[28229]: Monitor doohicky got event Hook
Transition Complete on channel 2
2005-02-20 14:05:54 DEBUG[28229]: Monitor doohicky got event On hook on
channel 2
2005-02-20 14:05:54 DEBUG[28229]: disabled echo cancellation on channel
2
2005-02-20 14:06:06 DEBUG[28229]: Monitor doohicky got event
Ring/Answered on channel 2
2005-02-20 14:06:06 DEBUG[28229]: Device 'Zap/2' changed to state '2'
2005-02-20 14:06:06 DEBUG[28229]: Device 'Zap/2' changed to state '2'
2005-02-20 14:06:08 DEBUG[28229]: DTMF digit: 1 on Zap/2-1
2005-02-20 14:06:08 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:06:08 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:06:08 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:06:09 DEBUG[28229]: DTMF digit: 3 on Zap/2-1
2005-02-20 14:06:09 DEBUG[28229]: DTMF digit: 6 on Zap/2-1
2005-02-20 14:06:09 DEBUG[28229]: DTMF digit: 2 on Zap/2-1
2005-02-20 14:06:09 DEBUG[28229]: DTMF digit: 9 on Zap/2-1
2005-02-20 14:06:09 DEBUG[28229]: DTMF digit: 7 on Zap/2-1
2005-02-20 14:06:10 DEBUG[28229]: DTMF digit: 0 on Zap/2-1
2005-02-20 14:06:10 DEBUG[28229]: DTMF digit: 4 on Zap/2-1
2005-02-20 14:06:10 DEBUG[28229]: Enabled echo cancellation on channel 2
2005-02-20 14:06:10 DEBUG[28229]: Launching 'Hangup'
2005-02-20 14:06:10 DEBUG[28229]: Spawn extension
(default,1888362970

RE: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4 C.O. lines

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> Jim Van Meggelen wrote:
> 
>> Yep, that's a possibility, but it's rather more kludgy than I'd like.
>> (heck, the channel bank and T1 is more kludgy than I'd like - an
>> integrated card would be my preference).
> 
> I haven't followed this thread closely but have you looked into the
> Voicetronix OpenSwitch cards? 
> 
> http://www.voicetronix.com.au/hda.htm

I've looked at them, but never heard much about them. Is anyone using
them? Can anyone give a comparison vs. the TDM400?


-- 
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Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005
 

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Re: [Asterisk-Users] What happens if quadbri or octobri loses power - do they have power failure feature ?

[Brancaleoni Matteo]

Hi,

> I wonder if I use quadbri or octobri cards to insert Asterisk between ISDN
> PBX and ISDN line - if power of Asterisks fails - will those card connect
> PBX directly to ISDN line ? 
No, you need a isdn failover switch

> If not are there any other simple switching
> devices, that would do this (in power fail it will connect ISDN PBX to ISDN
> lines directly) ?
Yes, klaus (author of bristuff) has/will have a solution for that.
Hardware isdn failover switch.

I don't know if I can reveal some details on this magic,
so please contact him for further details

Matteo.


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[Asterisk-Users] What happens if quadbri or octobri loses power - do they have power failure feature ?

[Robert Rozman]

Hi,

I mistakenly posted this to Dev list

I wonder if I use quadbri or octobri cards to insert Asterisk between ISDN
PBX and ISDN line - if power of Asterisks fails - will those card connect
PBX directly to ISDN line ? If not are there any other simple switching
devices, that would do this (in power fail it will connect ISDN PBX to ISDN
lines directly) ?

Thanks in advance,

regards,

Rob.

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[Asterisk-Users] Recording of calls stopped - normal behaviour?

[Eric Bishop]

Hi all,

I have call recording enabled via the Monitor command and it seems,
the call stops being recorded after the call is transferred. Is this
normal behavior? If so how can I continue recording of calls after
they have been trasnferred
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 7, Issue 260

[David Cook]

> From: "James Bean" <[EMAIL PROTECTED]>
> Has anyone every setup an external open/close relay, off say a serial
> interface, and have an extension trigger the relay?

The following will do the trick. Just add a 5vdc solid state relay
('cause you can't sink too much current out of the RS232C port).
Substitute "2", "4" or "6" in the code below to turn on either DTR, RTS
or both signals. "0" for off.

Change SWDEV in the lpswitch.h file to be the serial port you intend to 
use for the relay. I'm using some optically isolated relays I found in
town for $5.00 Cdn. The box to put it in cost more than the relay.

There is a bunch of extra defines in the .h file that were needed for
the larger project this was part of. Just ignore them, they won't hurt.

Call this program from your dialplan, and voila.

Compile with cc -i lpon.c -o lpon


  /*

   * lpon.c   Lineprinter ON

   *  *** test program only **

   *

   *  (c) David Cook, 1994

   *

   *  Set signlal lines on serial port to turn on 5vdc

   *  signal. Used for solid-state relay (low current

   *  draw on RS232C port) to switch high voltage/high

   *  current load for printer.

   *

   *  Part of an intelligent print spooler to only power

   *  on/off high draw printer when required.

   *

   * Usage:   lpon  

   *  For example, lpon /dev/cua4 4 to set bit 3 on

   *  port /dev/cua4.

   *  "4" = 0100 or bit 3 which is DTR

   *  "2" = 0010 or bit 2 which is RTS

   *  "6" = 0110 or both DRT & RTS

   */

  #include 

  #include 

  #include 

  #include 

  #include 

  #include 

  #include 

  #include 

  #include 



  #include "lpswitch.h"



  /* Main program. */

  int main(int argc, char **argv)

  {

struct termios port_config;

int fd;

int set_bits = 6;



/* Open monitor device. */

if ((fd = open(SWDEV, O_RDWR | O_NDELAY)) < 0) {

  fprintf(stderr, "lpswtich: %s: %s\n", SWDEV, sys_errlist[errno]);

  exit(1);}



cfmakeraw( &port_config );

port_config.c_iflag=port_config.c_iflag|IXON;

port_config.c_oflag=port_config.c_oflag|CLOCAL|~CRTSCTS;

tcsetattr( fd, TCSANOW, &port_config );

ioctl(fd, TIOCMSET, &set_bits );

sleep(5);

close(fd);

}


/* lpswitch.h
 * include file for lpswitchd configuration
 * (c) 1994, David Cook <[EMAIL PROTECTED]>
 */

#include

#define FILTERDEUG  0   /* filter app debug   */
#define DAEMONDEBUG 0   /* daemon app debug   */
#define VERSION "0.6"   /* appl version number*/
#define LOCKF   "/var/run/lpswitchd.pid" /* lock/PID file  */
#define READYFILE   "/tmp/lpready"  /* printer ready file */
#define RQSTFILE"/tmp/lprequest" /* lprequest file */
#define LPDEV   "/dev/lp0"  /* physical lp device */
#define SWDEV   "/dev/ttyC0"/* switch port-HW box */
#define SPEED   B9600   /* port baud rate */
#define RESET   B0  /* reset by 0 speed   */
#define WARMUP  45  /* 45 sec warmup delay*/
#define IDLE1200/* 1200 seconds (20min)
   idle delay */
#define XON 17  /* XON character  */
#define XOFF19  /* XOFF character */
#define ABORTTIME   90  /* Max before abort   */

dbc.
David Cook
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RE: [Asterisk-Users] No Sounds; stumping "The Tryant"

[Race Vanderdecken]

Yes, running extra code/libraries/.so means more variables, which we are
trying to eliminate.

Really Anton I am stumped.

Does anyone know? Do you have to have the gsm codec to hear the .gsm
sound files.

Is there an [EMAIL PROTECTED] mailing list?

I found these:
http://www.uninett.no/voip/asterisk.html
No sound on SIP
I had a "allow=all" codecs in the 'sip.conf' while which sort of
"stopped" all sound out. I commented it out, and it was up and running
on the sound. Now I just allow for the g.711 codec with

disallow=all
allow=ulaw 





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Sunday, February 20, 2005 2:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

Ok Noload modems, alsa and oss... No errors... Is this ok? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Domingo, 20 de Febrero de 2005 01:18 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

I don’t know if it has something to do but I see 2 mpg123 processes
running:

 3552 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 3553 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 
Everytime I start asterisk..

Also, if I enable alsa I get this error:

 [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301
__load_resource:
/usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file:
No
such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459
load_modules: Loading module chan_alsa.so failed!
[EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested!
Ouch ... error while writing audio data: : Broken pipe

And asterisk quits... 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 12:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

This is getting hard.

So what do we have?

1. The Asterisk server and the phones are using good CODECS.
2. Sound is moving from phone to phone.
3. Sound from the prompts is not playing back to the phones.

So let's go with the first principle: Eliminate the variables.

Do this:

Extensions.conf

[NoSound]

exten => 222,1,Wait(2)
exten => 222,2,Answer
exten => 222,3,Playback(vm-isunavail)
exten => 222,4,Hangup

And see what happens.

I might be missing something. Anyone know how .gsm files are translated
to
ulaw/alaw in asterisk?

Race "The Tyrant" Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, February 19, 2005 10:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Race.

Here are thre results of the tests:

Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
-
0x0 (nothing) 

Seems both can speak ulaw and ilbc but still, no prompt on the phone :(

Any ideas? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 07:15 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Correct.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

This is a very good place to start Race. So if I understand you
correctly,
Ill do the sip debug but maybe trying to force both to use ilbc or
ulaw/alaw
might help so I can listen to the prompts right?

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 06:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Grasshopper, 

You have your first clue, the live test works.

Do you understand how SIP works? During the INVITE sequence the Asterisk
and
the phone trade RTP CODEC information. RTP is the protocol that actually
carries the sounds, SIP only does the handshaking for the call.
A CODEC is what the RTP is carrying between the pones.

If you do "sip debug" inside of the asterisk command line interface
 
*CLI> sip debug

Then you will see the SIP Messages and the Codec agreements.

...
16 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.102 : 5060 (non-NAT)

RE: [Asterisk-Users] A bit of a survey: What do do if you need morethan 4 C.O. lines

[Steven Critchfield]

On Sun, 2005-02-20 at 11:47 -0500, Jim Van Meggelen wrote:
> Well, I appreciate everyone's input, and I'll give the matter some more
> thought.
> 
> Just so no one stays up at night worrying, this is not a situation I am
> facing, it is simply a hypothetical scenario.
> 
> As with so many things, there is always a trade-off between economy and
> functionality. The Adit 600 and T1 integration is certainly quality, but
> I have not been able find an economical way to do this (purchasing used
> equipment on eBay is fine for smaller deployments and lab gear, but not
> a very sound logistics strategy, and awfully difficult to explain to a
> customer).

This would be one of those cases where you keep a couple in stock and
watch the ebay auctions when your stock goes low. You will find that
your customers that are looking for the cheapest solutions possible will
not baulk at used equipment. It is highly likely that they will price
you against a used key system or pbx.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] No Sounds; stumping "The Tryant"

[Anton Krall]

Ok Noload modems, alsa and oss... No errors... Is this ok? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Domingo, 20 de Febrero de 2005 01:18 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

I don’t know if it has something to do but I see 2 mpg123 processes running:

 3552 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 3553 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 
Everytime I start asterisk..

Also, if I enable alsa I get this error:

 [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301 __load_resource:
/usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file: No
such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459
load_modules: Loading module chan_alsa.so failed!
[EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested!
Ouch ... error while writing audio data: : Broken pipe

And asterisk quits... 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 12:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

This is getting hard.

So what do we have?

1. The Asterisk server and the phones are using good CODECS.
2. Sound is moving from phone to phone.
3. Sound from the prompts is not playing back to the phones.

So let's go with the first principle: Eliminate the variables.

Do this:

Extensions.conf

[NoSound]

exten => 222,1,Wait(2)
exten => 222,2,Answer
exten => 222,3,Playback(vm-isunavail)
exten => 222,4,Hangup

And see what happens.

I might be missing something. Anyone know how .gsm files are translated to
ulaw/alaw in asterisk?

Race "The Tyrant" Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 10:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Race.

Here are thre results of the tests:

Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
-
0x0 (nothing) 

Seems both can speak ulaw and ilbc but still, no prompt on the phone :(

Any ideas? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 07:15 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Correct.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

This is a very good place to start Race. So if I understand you correctly,
Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw
might help so I can listen to the prompts right?

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 06:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Grasshopper, 

You have your first clue, the live test works.

Do you understand how SIP works? During the INVITE sequence the Asterisk and
the phone trade RTP CODEC information. RTP is the protocol that actually
carries the sounds, SIP only does the handshaking for the call.
A CODEC is what the RTP is carrying between the pones.

If you do "sip debug" inside of the asterisk command line interface
 
*CLI> sip debug

Then you will see the SIP Messages and the Codec agreements.

...
16 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.102 : 5060 (non-NAT) Found RTP audio format 0 Found RTP
audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP
audio format 101 Peer audio RTP is at port 192.168.1.102:10054 Found
description format pcmu Found description format pcma Found description
format gsm Found description format g729 Found description format
telephone-event
--
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10e
(gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)

...
...
m=audio 19958 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

Asterisk is the " us - 0x8000e (gsm|ulaw|alaw|h263)"
Phone is the " peer - audio=0x10e (gsm|ulaw|alaw|g729)"

Above is the trace of my SNOM 200 -- see the "combined - 0xe
(gsm|ulaw|alaw)"?

The phone can do g729, but asterisk can't, so asterisk and th

RE: [Asterisk-Users] No Sounds; stumping "The Tryant"

[Anton Krall]

I don’t know if it has something to do but I see 2 mpg123 processes running:

 3552 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 3553 pts/1S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 
Everytime I start asterisk..

Also, if I enable alsa I get this error:

 [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301 __load_resource:
/usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file: No
such file or directory
Feb 20 13:16:44 WARNING[3573]: loader.c:459 load_modules: Loading module
chan_alsa.so failed!
[EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested!
Ouch ... error while writing audio data: : Broken pipe

And asterisk quits... 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 12:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

This is getting hard.

So what do we have?

1. The Asterisk server and the phones are using good CODECS.
2. Sound is moving from phone to phone.
3. Sound from the prompts is not playing back to the phones.

So let's go with the first principle: Eliminate the variables.

Do this:

Extensions.conf

[NoSound]

exten => 222,1,Wait(2)
exten => 222,2,Answer
exten => 222,3,Playback(vm-isunavail)
exten => 222,4,Hangup

And see what happens.

I might be missing something. Anyone know how .gsm files are translated to
ulaw/alaw in asterisk?

Race "The Tyrant" Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 10:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Race.

Here are thre results of the tests:

Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
-
0x0 (nothing) 

Seems both can speak ulaw and ilbc but still, no prompt on the phone :(

Any ideas? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 07:15 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Correct.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

This is a very good place to start Race. So if I understand you correctly,
Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw
might help so I can listen to the prompts right?

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 06:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Grasshopper, 

You have your first clue, the live test works.

Do you understand how SIP works? During the INVITE sequence the Asterisk and
the phone trade RTP CODEC information. RTP is the protocol that actually
carries the sounds, SIP only does the handshaking for the call.
A CODEC is what the RTP is carrying between the pones.

If you do "sip debug" inside of the asterisk command line interface
 
*CLI> sip debug

Then you will see the SIP Messages and the Codec agreements.

...
16 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.102 : 5060 (non-NAT) Found RTP audio format 0 Found RTP
audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP
audio format 101 Peer audio RTP is at port 192.168.1.102:10054 Found
description format pcmu Found description format pcma Found description
format gsm Found description format g729 Found description format
telephone-event
--
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10e
(gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)

...
...
m=audio 19958 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

Asterisk is the " us - 0x8000e (gsm|ulaw|alaw|h263)"
Phone is the " peer - audio=0x10e (gsm|ulaw|alaw|g729)"

Above is the trace of my SNOM 200 -- see the "combined - 0xe
(gsm|ulaw|alaw)"?

The phone can do g729, but asterisk can't, so asterisk and the phone agree
on a non-g729 codec, ulaw. 

Eventually the phone agrees to "a=rtpmap:0 PCMU/8000", it is going to talk
sound using ulaw at 8000hz. (again, I might be a little wrong on the extact
details.)

If the phones agree to use G729 then the playback won't work because you
don't have a g729 license, $10 from Digium.

Remember that aster

Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}

[Bruno Hertz]

On Sun, 2005-02-20 at 13:51 -0500, Paul wrote:

> Or maybe a double fool because he also disrespected Debian GNU/Linux in 
> his reply. 

*And* recommended Fedora, which makes it triple. I just dumped FC3 and
replaced it with Debian because Fedora's kernels constantly gave me
issues, e.g. with proprietary AVM kernel drivers which didn't even work.
On the other hand, no probs whatsoever with Debian.



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RE: [Asterisk-Users] No Sounds; stumping "The Tryant"

[Anton Krall]

Ok... I added the extension and here are the results:

-- Executing Wait("SIP/intruder-phone1-8613", "2") in new stack
-- Executing Answer("SIP/intruder-phone1-8613", "") in new stack
-- Executing Playback("SIP/intruder-phone1-8613", "vm-isunavail") in new
stack
-- Playing 'vm-isunavail' (language 'en')

On the sip phone I hear no prompts or recordings.  :(

I tried rebooting the system, and weird, it worked once, and then, it
stopped working.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 12:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

This is getting hard.

So what do we have?

1. The Asterisk server and the phones are using good CODECS.
2. Sound is moving from phone to phone.
3. Sound from the prompts is not playing back to the phones.

So let's go with the first principle: Eliminate the variables.

Do this:

Extensions.conf

[NoSound]

exten => 222,1,Wait(2)
exten => 222,2,Answer
exten => 222,3,Playback(vm-isunavail)
exten => 222,4,Hangup

And see what happens.

I might be missing something. Anyone know how .gsm files are translated to
ulaw/alaw in asterisk?

Race "The Tyrant" Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 10:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Race.

Here are thre results of the tests:

Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
-
0x0 (nothing) 

Seems both can speak ulaw and ilbc but still, no prompt on the phone :(

Any ideas? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 07:15 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Correct.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

This is a very good place to start Race. So if I understand you correctly,
Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw
might help so I can listen to the prompts right?

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 06:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Grasshopper, 

You have your first clue, the live test works.

Do you understand how SIP works? During the INVITE sequence the Asterisk and
the phone trade RTP CODEC information. RTP is the protocol that actually
carries the sounds, SIP only does the handshaking for the call.
A CODEC is what the RTP is carrying between the pones.

If you do "sip debug" inside of the asterisk command line interface
 
*CLI> sip debug

Then you will see the SIP Messages and the Codec agreements.

...
16 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.102 : 5060 (non-NAT) Found RTP audio format 0 Found RTP
audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP
audio format 101 Peer audio RTP is at port 192.168.1.102:10054 Found
description format pcmu Found description format pcma Found description
format gsm Found description format g729 Found description format
telephone-event
--
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10e
(gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)

...
...
m=audio 19958 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

Asterisk is the " us - 0x8000e (gsm|ulaw|alaw|h263)"
Phone is the " peer - audio=0x10e (gsm|ulaw|alaw|g729)"

Above is the trace of my SNOM 200 -- see the "combined - 0xe
(gsm|ulaw|alaw)"?

The phone can do g729, but asterisk can't, so asterisk and the phone agree
on a non-g729 codec, ulaw. 

Eventually the phone agrees to "a=rtpmap:0 PCMU/8000", it is going to talk
sound using ulaw at 8000hz. (again, I might be a little wrong on the extact
details.)

If the phones agree to use G729 then the playback won't work because you
don't have a g729 license, $10 from Digium.

Remember that asterisk is a third party to a conference and if your
conference is using g729, then asterisk can't do that.

In the sip.conf, 

Disallow=all
Allow=gsm
Allow=ulaw
Allow=alaw

This will force the phone and asterisk to speak gsm, ulaw or alaw.

I had the same experience with no sound when I first connected a Cisco 79

RE: [Asterisk-Users] Simulated dialtone like in other PBX

[Anton Krall]

Thx Sergey!! Ill give it a try

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergey
Kuznetsov
Sent: Domingo, 20 de Febrero de 2005 07:34 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Simulated dialtone like in other PBX

Easy as piece of cake.

Remove ignorepat=>9

add:

exten => 9,1,DISA(no-password|my_outbound_context)

[my_outbound_context]

exten => NXX, 1, blah-blah-blah

All the Best!
Sergey.

Peter Svensson wrote:

>On Sun, 20 Feb 2005, Anton Krall wrote:
>
>  
>
>>Im new to asterisk but is it possible to simulate a dialtone for 
>>example, in other PBX when you pick up the phone you can hear a 
>>certain dialup, which is the PBX dialtone, and when you hit 9, you can 
>>hear the PSTN dialtone, is this possible?
>>
>>
>
>I'm not sure I understand your question. 
>
>Do you want to be able to hit 9 and get a an outside line with dialtone? 
>Just add an extension to do that. For isdn you need to enable overlap 
>dialing.
>
>Or do you want Asterisk to provide a dialtone after the user have hit 9 
>as the first digit of a number? User the ignorepat option in the dialplan.
>
>Or do you want Asterisk to provide a _different_ dialtone after the 
>user have hit 9 as the first digit of a number? This may be possible, 
>but I think some hack may be needed.
>
>Peter
>
>
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RE: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}

[Race Vanderdecken]

I have no problem with Slackware,

But when you are learning to drive a car you should first try a Chevy
with an automatic transmission first before strapping on a 6 speed
Ferrari.

Humor helps in teaching and getting a person to step out of a rut they
are having a problem in and gives them a chance to rethink what might be
going on.

Remember, my goal is to reduce the number of variables in the system. 

Race



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Sunday, February 20, 2005 1:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [Asterisk-Users] Segmentation fault {Writer given
gnu-lashing}

On February 20, 2005 01:11 pm, Race Vanderdecken wrote:
> 1. Why are you running on Slackware?
>  Are you trying to prove a point or just enjoy being frustrated?
>  Open Source is like "Broad Spectrum Pesticide", it works but
> your results may vary and you may end up killing your lawn.

Got a problem with Slackware?  It works *very* well with Asterisk.

-A.
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Re: [Asterisk-Users] Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11?

[El Panitaxx --]

Push it with enough force, it will come in. 


On Sun, 20 Feb 2005 11:51:05 -0700, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> 
> Hello, 
>  I bought a TDM400P, and intended to use it with my analog phone, which is
> RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the
> TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I
> also intend to bring in an analog line into the RJ45, so i am still left
> with the same questionhow do I go from the RJ11 standard analog to the
> RJ45 on the TDM400P card? I'd appreciate any response. 
>   
> thx 
> chuks 
> ___
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Re: [Asterisk-Users] Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11?

[steve]

On Sun, 20 Feb 2005 [EMAIL PROTECTED] wrote:

> 
> Hello,
>  I bought a TDM400P, and intended to use it with my analog phone, which is 
> RJ11 ofcourse. So, the question now, how do I plug in
> my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since 
> it's an 11B card, I also intend to bring in an
> analog line into the RJ45, so i am still left with the same questionhow 
> do I go from the RJ11 standard analog to the RJ45 on
> the TDM400P card? I'd appreciate any response.


Just plug the RJ11 into the socket - it will go in fine and work.

This is a designed-in feature of the RJ series connectors, if I'm not 
mistaken.

Steve

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RE: [Asterisk-Users] No Sounds; stumping "The Tryant"

[Race Vanderdecken]

This is getting hard.

So what do we have?

1. The Asterisk server and the phones are using good CODECS.
2. Sound is moving from phone to phone.
3. Sound from the prompts is not playing back to the phones.

So let's go with the first principle: Eliminate the variables.

Do this:

Extensions.conf

[NoSound]

exten => 222,1,Wait(2)
exten => 222,2,Answer
exten => 222,3,Playback(vm-isunavail)
exten => 222,4,Hangup

And see what happens.

I might be missing something. Anyone know how .gsm files are translated
to ulaw/alaw in asterisk?

Race "The Tyrant" Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, February 19, 2005 10:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Race.

Here are thre results of the tests:

Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
-
0x0 (nothing) 

Seems both can speak ulaw and ilbc but still, no prompt on the phone :(

Any ideas? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 07:15 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Correct.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

This is a very good place to start Race. So if I understand you
correctly,
Ill do the sip debug but maybe trying to force both to use ilbc or
ulaw/alaw
might help so I can listen to the prompts right?

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 06:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Grasshopper, 

You have your first clue, the live test works.

Do you understand how SIP works? During the INVITE sequence the Asterisk
and
the phone trade RTP CODEC information. RTP is the protocol that actually
carries the sounds, SIP only does the handshaking for the call.
A CODEC is what the RTP is carrying between the pones.

If you do "sip debug" inside of the asterisk command line interface
 
*CLI> sip debug

Then you will see the SIP Messages and the Codec agreements.

...
16 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.102 : 5060 (non-NAT) Found RTP audio format 0 Found
RTP
audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found
RTP
audio format 101 Peer audio RTP is at port 192.168.1.102:10054 Found
description format pcmu Found description format pcma Found description
format gsm Found description format g729 Found description format
telephone-event
--
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10e
(gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)

...
...
m=audio 19958 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

Asterisk is the " us - 0x8000e (gsm|ulaw|alaw|h263)"
Phone is the " peer - audio=0x10e (gsm|ulaw|alaw|g729)"

Above is the trace of my SNOM 200 -- see the "combined - 0xe
(gsm|ulaw|alaw)"?

The phone can do g729, but asterisk can't, so asterisk and the phone
agree
on a non-g729 codec, ulaw. 

Eventually the phone agrees to "a=rtpmap:0 PCMU/8000", it is going to
talk
sound using ulaw at 8000hz. (again, I might be a little wrong on the
extact
details.)

If the phones agree to use G729 then the playback won't work because you
don't have a g729 license, $10 from Digium.

Remember that asterisk is a third party to a conference and if your
conference is using g729, then asterisk can't do that.

In the sip.conf, 

Disallow=all
Allow=gsm
Allow=ulaw
Allow=alaw

This will force the phone and asterisk to speak gsm, ulaw or alaw.

I had the same experience with no sound when I first connected a Cisco
7960,
I could here other people, but not the prompts. Asterisk will allow G729
to
pass through, but it will not allow G729 to originate and terminate
without
the license (I might be a little mistaken here...)

I hope this helps. I have not use [EMAIL PROTECTED], it might be different.

Let me know,

Race "The Tyrant" Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, February 19, 2005 7:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Hi Race..

In this case, the asterisk|home comes preconfigured with some stuff
different than the asterisk tar file.

I check and the phone supports all mentioned codecs, I

RE: [Asterisk-Users] External relay triggered by Asteriskextension-question

[Jay Milk]

Sorry, I didn't say I was using it with * -- just on a PC with a
different app.  I don't think it would be difficult to use something
like lcdproc or even their test-app --
http://www.crystalfontz.com/software/633_WinTest/index.html (link to
linux source at the bottom), and use agi to call the application.

Basically, you got all the dots and all the connections.

> -Original Message-
> From: James Bean [mailto:[EMAIL PROTECTED] 
> Sent: Sunday, February 20, 2005 5:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] External relay triggered by 
> Asteriskextension-question
> 
> 
> Very friggen cool, that you very much for the information it 
> looks like it will do the job nicely.
> 
> What did you use in your extensions list to activate the relay?
> 
> James 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf 
> Of Jay Milk
> > Sent: Sunday, 20 February 2005 6:24 PM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [Asterisk-Users] External relay triggered by 
> > Asterisk extension-question
> > 
> > Done something similar in a different application, but *
> > should handle it --
> > 
> > In my case, I took a crystalfontz LCD, type 633, and used two
> > of the four fan-outputs to drive two 12V relays.  As a nice 
> > extra, you get temperature capabilities thrown in, so you can 
> > monitor your set-up.  The LCD runs on serial, of course.
> > 
> > As an alternative, you can use any of the many available
> > relay boards -- $50 gets you this:
> > http://www.phanderson.com/iom141.html
> > 
> > > -Original Message-
> > > From: James Bean [mailto:[EMAIL PROTECTED]
> > > Sent: Saturday, February 19, 2005 11:34 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: [Asterisk-Users] External relay triggered by Asterisk
> > > extension -question
> > > 
> > > 
> > > 
> > > Has anyone every setup an external open/close relay, off
> > say a serial
> > > interface, and have an extension trigger the relay?
> > > 
> > > Why I ask is I have a student accomodation where I am 
> installing an
> > > asterisk box to supply phone services to the tenants, there 
> > is already
> > > an intercom system in the main hallways that triggers the
> > downstairs
> > > door and gate using a standard relay open/close trip, so I
> > was hoping
> > > to get the linux box with asterisk to trip the same type of relay.
> > > 
> > > Is there any door phones that are speaker driven only and 
> sip based
> > > that anyone knows about as well?
> > > 
> > > James
> > > 
> > > ___
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> > > http://lists.digium.com/mailman/listinfo/aster> isk-users To 
> > > UNSUBSCRIBE or update options visit:
> > >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> > 
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> > 
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Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}

[Paul]

Brian Capouch wrote:
Race Vanderdecken wrote:
Good, then let me move on to the insults and ranting.
1. Why are you running on Slackware? Are you trying to prove a 
point or just enjoy being frustrated?
Open Source is like "Broad Spectrum Pesticide", it works but
your results may vary and you may end up killing your lawn.

Why do you not follow Ann Landers simple adage, "Better to keep one's 
mouth shut and be thought a fool, than to open it and remove any doubt?"

Or maybe a double fool because he also disrespected Debian GNU/Linux in 
his reply. Is ignorance really bliss?

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