Re: [Asterisk-Users] Follow Me & Music on hold

[Kristian Kielhofner]

Me wrote:
OK, I have an extension setup with a follow me like so:
;Operator Going to Sue first, then Mary
exten => 0,1,playback(pls-wait-connect-call)
exten => 0,2,Dial(SIP/103,20,mTt)
exten => 0,3,Dial(SIP/102,20,mTt)
exten => 0,4,VoiceMail([EMAIL PROTECTED])
exten => 0,5,Goto,t|1
This works well except for the fact that the music on hold stops after 
the first timeout and starts over at the beginning of the next line. 
What I mean is that the music sort of skips a beat (so to speak) when * 
stops ring extension 103 and starts ringing extension 102.

Can someone suggest a better/smoother way to do this so the music just 
continues to play until both extensions timeout?

--
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http://www.YourOwnISP.com
What about calling them both at the same time, not sequentially:
exten => 0,1,playback(pls-wait-connect-call)
exten => 0,2,Dial(SIP/103&SIP/102,20,mTt)
exten => 0,3,VoiceMail([EMAIL PROTECTED])
exten => 0,4,Goto,t|1
asterisk -rx "show application Dial"
would have told you this!
--
Kristian Kielhofner
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Re: [Asterisk-Users] Very Cool.........Asterisk Made Wired Magazine

[Wilson Pickett]

> I suppose Asterisk is first going to get press as an über-geek's
> home-brew PBX, but I sometimes wish this weren't so. It is such a
> legitimate technology that it should be getting press in more industry
> publications. Ah well, that'll come in good time I guess.

You're right there, Jim: "if it bleeds, it leads" is the motto of TV
news, that gives an idea of the press mentality. I always thought
Wired was more of a yuppie lifestyle magazine than one to attract
phrackers or real hackers. Still, there may be an influx of newbies
looking to "spoof cid". If this happens, I hope questions will be
answered without scaring anyone who is really interested away.

With the upward spiral of voIP popularity and buzz, asterisk should be
talked about in something like PC Magazine where suits look at stuff
to buiy for their offices. Maybe someone should try to gather some
soho case studies and offer them to a widely-read "serious" computer
magazine, "serious" meaning one read by people who are either small
biz owners or who have decisional power for small and mid level
business strategy.
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[Asterisk-Users] Follow Me & Music on hold

[Me]

OK, I have an extension setup with a follow me like so:
;Operator Going to Sue first, then Mary
exten => 0,1,playback(pls-wait-connect-call)
exten => 0,2,Dial(SIP/103,20,mTt)
exten => 0,3,Dial(SIP/102,20,mTt)
exten => 0,4,VoiceMail([EMAIL PROTECTED])
exten => 0,5,Goto,t|1
This works well except for the fact that the music on hold stops after the 
first timeout and starts over at the beginning of the next line. What I mean 
is that the music sort of skips a beat (so to speak) when * stops ring 
extension 103 and starts ringing extension 102.

Can someone suggest a better/smoother way to do this so the music just 
continues to play until both extensions timeout?

--
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http://www.YourOwnISP.com 

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Re: [Asterisk-Users] I'm stumped

[Wilson Pickett]

> the files:  I have asterisk running as the user asterisk also.

who is the web server running as?

isn't the system() function seeing this:

sudo -u asterisk /usr/sbin/asterisk -rx database put cidname
$PhoneNumber \"$PhoneName\""

when it should be seeing this after the sudo -u asterisk :

"asterisk /usr/sbin/asterisk -rx database put cidname (etc)" <-- whole
cmd in quotes

Try typing this at the linux prompt:

asterisk -rx database put cidname 123 "Julius Ceasar"

I don't think it will work
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Re: [Asterisk-Users] New PRI with DID in US?

[Kevin Blackham]

On Fri, 10 Dec 2004 17:26:48 -0600, Rich Adamson <[EMAIL PROTECTED]> wrote:
> Just turned up a new PRI with DID's in the US. I'm receiving 5 digits
> of the DID numbers as I requested.
> 
> Assuming I have 100 DID numbers but only define 50 of those in
> extensions.conf, is there an easy way to send the incoming calls
> for the 20 undefined numbers to a common resource (ivr, operator,
> or canned message) without having to define each one?

I handle my DIDs with a macro.  A DBget fetches a target for Goto.  If
the key doesn't exist, it jumps to a hangup macro that can either drop
with PRI_CAUSE=1 (invalid) or play Zapateller and ss-noservice.gsm
twice, then hang up with PRI_CAUSE=31, depending on how you want it to
work.  Of course, don't answer first, and if you do Playback(),
remember the noanswer option and play a silence/1 first..

Some samples from database show (numbers changed to protect the
guilty, and I receive full 10 digits)
/DID/9001235900   : mainmenu|s
/DID/9001235904   : 104  
/DID/9001235917   : 117  
/DID/9001235939   : 139  
/DID/9001235942   : 142  
/DID/9001235970   : 170  
/DID/9001235949   : disa|s

[from-pstn]
exten => _NXXNXX,1,Macro(did,${EXTEN});

[macro-did]
exten => s,1,DBget(target=DID/${ARG1})
exten => s,2,Goto(${target},1); db should not include the priority
exten => s,102,SetVar(PRI_CAUSE=1); tells telco to play discon message
exten => s,103,Hangup; 
; target can be simply an extension: my stdexten does more DBget to find channel
exten => _1XX,1,Macro(stdexten,${EXTEN})
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RE: [Asterisk-Users] Log's Message Codes

[Brian West]

noload => chan_oss.so in modules.conf

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Raúl Gómez Cabrera
> Sent: Sunday, December 12, 2004 9:43 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Log's Message Codes
> 
> Hi All,
> 
> Anybody knows where can I find more explanation about the log's message
> codes of Asterisk?
> 
> By the way, anybody had this VERY ANNOYING warning flooding the logs?
> 
> WARNING[23678]: Read error on sound device: File descriptor in bad state
> 
> With the default config of logger.conf it can reach 2GB in a few hours.
> What can it be?
> 
> Thanks for all!
> 
> Raul.
> 
> 
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[Asterisk-Users] DUNDi performance

[Marc Storck]

Hello,
I have a weird problem. My * server, a Pentium Celeron 1200 with 512 MB 
Ram and a Digium E100P card, is performing very well for IAX2, SIP and 
ZAP communication. There is no delay in transcoding, no packet loss etc etc.

Now I added DUNDi, and I added +/- 8 peers in the dundi-test context and 
1 peer in the GPA-bound e164 context. My server shows all but 1 peer as 
OK. DUNDi Ping times are between 20 and 200 ms.

The Problem is, that no server but one can get a stable connection via 
DUNDi to my server. DUNDi ping times for my server are between 3000 an 
7000 ms. Most servers have qualify of 2000ms, some even 500ms, so my 
server is quite always UNREACHABLE for those peers.

When I activate DUNDi DEBUG, I can see that incoming DUNDi packets do 
take all long time before they show up in the * console, and they always 
show up with a whole bunch of others (filling 2-3 screens). But then the 
debug output stops in the middle of 1 debug packet, to continue over 20 
seconds later (if it continues).

The actual CPU load is "load average: 0.00, 0.00, 0.00".
I cannot find the problem, maybe someone over can help me!
Thanks,
Marc
--
CTOMarc Storck
MS Networks SA [EMAIL PROTECTED]
Internet Service Provider  http://www.luxadmin.org
15, route d'Esch   Phone: +352 2727 3030
L-4544 Belvaux Fax:   +352 2727 3060
-- LuxAdmin powered service ---
http://www.Gateway.lu  Your gateway to the net
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RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.

[Matt Hyne]

> I do not have direct access to the Cisco, but I believe that he is
> AS5300

A "show ver" will confirm this and the IOS release. Please publish that info
as well.

I will see if I can get a similar setup going in our lab.

M

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RE: [Asterisk-Users] Very Cool.........Asterisk Made Wired Magazine

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> Hi Guys,
> 
> The article "They've Got Your number" in the Dec 2004 issue of WIRED 
> magazine mentions Asterisk PBX (on p.100).  The article is about phone

> phreaks hijacking cell phones with Bluetooth technology along with 
> spoofing CID to pull some clandestine hacks on the PSTN.  Anyhow, 
> Asterisk is mentioned as the PBX of choice for an outfit: 
> Telephreaks.org out of Florida that has built their own free VoIP 
> service.

I wish they had said a bit more about the usefulness of Asterisk, as
opposed to talking about how some phrackers are using it to spoof CID.
Then again, it's been said that there's no such thing as bad press.

> Quotes:
> 
> "Slestak, Da Beave, and GiD are the crew behind Florida-based 
> Telephreaks.org, a free VoIP service that they've built to run on a 
> roll-your-own, open source private branch exchange
> (PBX) system called Asterisk"  "But with Asterisk, there's no need for

> the phone company to manage your phone lines anymore.  Your can do it 
> yourself."
> 
> Well, it's good that Asterisk made WIRED magazine but really it should

> be on the front page with Mark's smiling mug on the cover like Linus' 
> was on the Nov 2003 issue of WIRED.

Ya, but that'll come in good time. There's no stopping Asterisk now!

> This is a good day for Asterisk because WIRED magazine has a huge 
> subscriber base and this article will be read by a lot of people, and 
> some of those people might have the gumption to check out Asterisk and

> see what it's all about.  If you all see a lot more newbie question 
> pop up in the next few weeks, be kind, help where you can and point to

> the Wiki for more info.

Good advice at any time. Doubly so now.

> I'll be writing a letter to the editor, encouraging them to check out 
> Asterisk as a possible full featured article and I encourage you all 
> to do the same, if you're so inclined.

Call me paranoid, but you might want to give some thought to the fact
that the story you want them to write about Asterisk may not spin the
way you think it will. Asterisk has been gaining a reputation as a
Phrackers PBX, and that kind of reputation may increase interest, but
not necessarily for the better. As Asterisk begins to take market share
away from traditional Telephony platforms, the FUD will increase as
well. The press serve their own ends, and if they decide that the better
spin is a story about the dangers of hackers, they can use Asterisk's
strengths against it. Remember that the press stole the term "hacker"
away from the community and turned something good into something
negative. Perhaps Wired is above all this, but security is a hotter
topic than open-source, and FUD sells magazines.

I suppose Asterisk is first going to get press as an über-geek's
home-brew PBX, but I sometimes wish this weren't so. It is such a
legitimate technology that it should be getting press in more industry
publications. Ah well, that'll come in good time I guess.

Cheers,

Jim.







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[Asterisk-Users] Sipura SPA-2000 won't ring

[Me]

I had a Grandstream 286 at my home hitting my Asterisk box at the office,
all worked well and I received phone calls fine until the device just up and
died.
I replaced this unit with an SPA-2000 because I have been impressed with the
Sipura devices and decided to use them for most of my needs in the future.
Problem is that my phone attached to the device rings shortly after power up
of the device but seems to lose it's head after a period of time and stops
ringing until I power cycle the unit or reboot it.
My Asterisk config is the same regarding NAT for this extension and I have
the Sipura registering with * so I am at a loss as to why Asterisk loses or
stops ringing this device.
I have dug around and can't seem to solve this issue so far, any help would
be appreciated.
--
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[Asterisk-Users] Log's Message Codes

[Raúl Gómez Cabrera]

Hi All,

Anybody knows where can I find more explanation about the log's message
codes of Asterisk?

By the way, anybody had this VERY ANNOYING warning flooding the logs?

WARNING[23678]: Read error on sound device: File descriptor in bad state

With the default config of logger.conf it can reach 2GB in a few hours.
What can it be?

Thanks for all!

Raul.


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Re: [Asterisk-Users] TDM400P FXS polarity reversal?

[Rich Adamson]

> Soren Rathje wrote:
> >
> > Specs for Si3210 (TDM400P FXS Module) says on page 93:
> >
> > ---
> > Register 72. On-Hook Line Voltage
> >
> > Bit 6 VSGN On-Hook Line Voltage.
> > The value written to this bit sets the on-hook line voltage polarity
> > (VTIP–VRING).
> >   0 = VTIP–VRING is positive
> >   1 = VTIP–VRING is negative
> > ---
> >
> 
> The (missing) link..
> 
> https://www.mysilabs.com/public/documents/tpub_doc/dsheet/Wireline/ProSLIC/en/si3210.pdf
> 
> Features
> ...
> ...
> * Software programmable signal generation and audio processing:
>   - DTMF generation and decoding
>   - 12 kHz/16 kHz pulse metering generation
>   - Phase-continuous FSK (caller ID) generation
>   - Dual audio tone generators
>   - Smooth and abrupt polarity reversal  <<== NB!
>   - µ-Law/A-Law and 16-bit linear PCM audio
> ...

Can you translate that into * code?



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Re: [Asterisk-Users] TDM400P FXS polarity reversal?

[Andrew Kohlsmith]

On December 12, 2004 09:59 pm, Rich Adamson wrote:
> Can you translate that into * code?

do_hangup()
{
 if(chan->signalling = FXO_KS)
   {
   if(!chan->reversed)
   {
  setreg(chan->port, funky_do_register, getreg(chan->port, 
funky_do_register) | BATT_REVERSAL);
  set_timer(chan->reversetime);
  chan->reversed = 1;
  }
   else
  {
  setreg(chan->port, funky_do_register, getreg(chan->port, 
funky_do_register) & !BATT_REVERSAL);
  chan->reversed = 0;
  }
}
}

??

Oh, it has to work...  :-)

-A.
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Re: [Asterisk-Users] Re: Cant set H323 up

[kido noagbodji]

what os are you running?

K.
- Original Message - 
From: "Rodolfo Grave" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial
Discussion" <[EMAIL PROTECTED]>
Sent: Monday, December 13, 2004 1:27 AM
Subject: Re: [Asterisk-Users] Re: Cant set H323 up


> Hi
>
> Now I do have compiled all the libraries, and added the
>
> load => chan_h323.so
>
> in the modules.conf file. Actually, now asterisk is attempting to load
> the chan_h323.so module. The problem is that Im getting this error now:
>
>   [chan_h323.so]Dec 13 02:24:01 WARNING[12023]: loader.c:258
> ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object
> file: No such file or directory
>
> I've moved the libpt_linux_x86_r.so.1.5.2 file to /usr/lib,
> /usr/lib/asterisk, /usr/lib/asterisk/modules
>
> After each move, I ran ldconfig the error was always the same...
> does anyone know where does asterisk looks for this file? Or if the
> cause for this is another?
>
> Im using the H323 channel included in the Asterisk tree.
>
> Thanks,
>
> RODOLFO
>
>
> Corvin wrote:
> > Rafael J. Risco G.V. wrote:
> >
> >
> >
> >>On Sat, 11 Dec 2004 16:49:12 +, Corvin <[EMAIL PROTECTED]> wrote:
> >>
> >>>Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisał:
> >>>
> Hi.
> 
> I need to set up H323 on an Asterisk box. I've succesfuly compiled the
> asterisk oh323 (including of course all the dependencies: PWlib and
> OpenH323), and then compiled asterisk. However, asterisk doesn't
report
> a registered H323 channel (when it starts, it reports IAX2, ZAP and
SIP
> channels, however, the 323 word doesn't appear in the whole output).
> 
> Is there anything I'm missing? I've read the documentation on the
wiki,
> and none said nothing about editing a config file. I did noticed that
> they talked about the oh323.conf file, which I dont have.
> >>>
> >
> >
> > BTW. you should check direcory with oh323 a there should be
asterisk-driver
> > directory and there you find sample config.
> > Then you sghould load module in modules.conf.
> >
> > BTW. I can't still compile any h323 driver :(((.
> >
> > Corvin
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[Asterisk-Users] patton smartnode integration

[Michael Lyszczek]

Any have any success using a patton smartnode 4118/js/eiu fxs gateway
with asterisk?  We we're able to get the unit to register with
asterisk, but when trying to place a call, no codec was compatible,
even though I had all of the following enabled on the patton ...


# G.711 A-Law/µ-Law (64kbps)
# G.726 (ADPCM 40, 32, 24, 16 kpbs)
# G.723.1 (5.3 or 6.3 kbps)
# G.729ab (8kbps)

the link to this product is : 

http://commerce.patton.com/pe_products.asp?category=51&MiDAS_SessionID=e41363efa86e409caf79ab1fd9b32e49

?

thanks for any help,
Michael Lyszczek
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Re: [Asterisk-Users] PoE VOIP phones in Australia

[James Andrewartha]

Adam Goryachev wrote:
See the polycom IP 300/500/600 phones. There are many resellers of these
phones in Australia. Note the 300/500 require an additional cable for
PoE.
Are there any that have online stores? I've searched fairly extensively and 
can only find brochureware sites.

Thanks,
James Andrewartha
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Re: [Asterisk-Users] zaptel 0.9.1 compile problem

[Kristian Kielhofner]

Jay Austad wrote:
I'm using gentoo 2004.3, and when I emerge the zaptel driver, compile 
fails with the following output:

  CC [M]  /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o
In file included from 
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:40:
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/zaptel.h:55:2: warning: 
#warning "Zaptel doesn't support DEVFS in post 2.4 kernels.  Disabling 
DEVFS in zaptel"
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c: In function 
`ztdeth_rcv':
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:89: error: 
union has no member named `ethernet'
make[2]: *** [/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o] 
Error 1
make[1]: *** [_module_/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1] 
Error 2
make[1]: Leaving directory `/usr/src/linux-2.6.9-gentoo-r9'
make: *** [linux26] Error 2


Any ideas why this would be?
~jay
Jay,
	Portage is good for many things, but Asterisk and Zaptel is not one of 
them.  Get them both from ftp.asterisk.org and you will be much better off.

--
Kristian Kielhofner
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[Asterisk-Users] Any plans for video in oh323?

[Bruno Hertz]

I did my happy first install of asterisk (cvs), and everything is
working great so far, with one exception.

Since I need h323 support, I first built chan_h323 with openh323
and pwlib pandora, and while the build went ok usage did not.

More specifically, while asterisk would accept h323 calls, no
voice was transmitted, hangup of the client was not recognized and
the server didn't properly shut down any more.

So I switched to oh323 with openh323 and pwlib janus, again the build
went alright, and this time usage did too :)

The only thing (obviously) missing from oh323 is video transport, hence
my question: is this feature planned in the near resp. far future?
I guess quite some people would be happy to see it included ...

Thanks, Bruno.


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Re: [Asterisk-Users] zaptel 0.9.1 compile problem

[Jay Austad]

Done, I got it compiled.  Looks like there was some things changed in 
kernel 2.6.9 that the newer versions of the zaptel driver have been 
modified for.

On Dec 12, 2004, at 8:03 PM, Kristian Kielhofner wrote:
Jay Austad wrote:
I'm using gentoo 2004.3, and when I emerge the zaptel driver, compile 
fails with the following output:
  CC [M]  /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o
In file included from 
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:40:
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/zaptel.h:55:2: 
warning: #warning "Zaptel doesn't support DEVFS in post 2.4 kernels.  
Disabling DEVFS in zaptel"
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c: In 
function `ztdeth_rcv':
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:89: error: 
union has no member named `ethernet'
make[2]: *** 
[/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o] Error 1
make[1]: *** 
[_module_/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1] Error 2
make[1]: Leaving directory `/usr/src/linux-2.6.9-gentoo-r9'
make: *** [linux26] Error 2
Any ideas why this would be?
~jay
Jay,
	Portage is good for many things, but Asterisk and Zaptel is not one 
of them.  Get them both from ftp.asterisk.org and you will be much 
better off.

--
Kristian Kielhofner
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Re: [Asterisk-Users] Re: Ethernet Channel Bank idea

[Gary]

On Sun, 12 Dec 2004 06:22:14 -0500 (EST), Greg Boehnlein wrote:

>On Fri, 10 Dec 2004, nik martin wrote:
>
>> news.gmane.org wrote:
>> > Allied Telesyn VoIP Access Device
>> > http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf
>> >  
>> > 
>> > 
>> > This is a 24-port FXS 1u device, conveniently presented as a single 
>> > RJ-21 TELCO connector.
>> 
>> yeah, but those are expensive as crap.  i was thinking about something 
>> more competetive with a channel bank
>
>You know, if someone had some time on their hands, was good at 
>hardware/software hacking and had the will, the old Livingston/Lucent PM3 
>platform would make an awesome 48 port IAX2 <-> PRI/T1 channel bank.
>
>Basically, the PM3 has 2 T1 ports that can be configured for ISDN PRI. The 
>core of the system runs on an AMD x86 CPU. The plug in Modem cards have 
>Lucent DSP's on them (up to 50 in a box). Flash size is 4 megs, and RAM is 
>usually around 4 megs. That is still quite a bit of horsepower, and the 
>boxes are under $400 now.
>
>The DSP's could be used for Codec Translation, if neccessary, or for echo 
>cancellation.
>
>And, we can get access to the original Lucent ComOS Source code.
>
>Anyone game? :)
>

It is not such a dumb idea !!

I am using The ericsson tigris platform for our few remaining PRI's for
our
dialup pools. We are currently hanging asterisk off these using drop &
insert
(both ways)... Now if we could actually get some code to get the voip
working to/from asterisk we could do away the the PRI cards

Gary
.


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Re: [Asterisk-Users] Re: Cant set H323 up

[Rodolfo Grave]

Hi
Now I do have compiled all the libraries, and added the
load => chan_h323.so
in the modules.conf file. Actually, now asterisk is attempting to load 
the chan_h323.so module. The problem is that Im getting this error now:

 [chan_h323.so]Dec 13 02:24:01 WARNING[12023]: loader.c:258 
ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object 
file: No such file or directory

I've moved the libpt_linux_x86_r.so.1.5.2 file to /usr/lib, 
/usr/lib/asterisk, /usr/lib/asterisk/modules

After each move, I ran ldconfig the error was always the same... 
does anyone know where does asterisk looks for this file? Or if the 
cause for this is another?

Im using the H323 channel included in the Asterisk tree.
Thanks,
RODOLFO
Corvin wrote:
Rafael J. Risco G.V. wrote:

On Sat, 11 Dec 2004 16:49:12 +, Corvin <[EMAIL PROTECTED]> wrote:
Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisał:
Hi.
I need to set up H323 on an Asterisk box. I've succesfuly compiled the
asterisk oh323 (including of course all the dependencies: PWlib and
OpenH323), and then compiled asterisk. However, asterisk doesn't report
a registered H323 channel (when it starts, it reports IAX2, ZAP and SIP
channels, however, the 323 word doesn't appear in the whole output).
Is there anything I'm missing? I've read the documentation on the wiki,
and none said nothing about editing a config file. I did noticed that
they talked about the oh323.conf file, which I dont have.


BTW. you should check direcory with oh323 a there should be asterisk-driver 
directory and there you find sample config.
Then you sghould load module in modules.conf.

BTW. I can't still compile any h323 driver :(((.
Corvin 
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Re: [Asterisk-Users] Using SPANDSP for faxes

[Ariel Batista]

Eric Hall wrote:
I installed spandsp on our asterisk server to get faxes. It works
however the images are a little off. Sometimes a few pages will be
together, pages missing and sentence missing.
Is this normal for this program?
Yes it is with some fax machines.  We had to make our own program that take 
the image and sets it correctly for viewing.  It's not a GPL program it's 
one we got as a test. If all goes well we will post what we are doing to fix 
the problems.  The programmer says it has to do with the libtiff library's.

Any input would be great.

Thank You
Eric
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RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.

[Jorge Verastegui G]

Hi
thanks for your help .


I do not have direct access to the Cisco, but I believe that he is
AS5300

The ios version is 12.2

and the cisco dum config is:

GWSCZ01>en
Password:
GWSCZ01#sh run
Building configuration...

Current configuration : 5053 bytes
!
! Last configuration change at 05:17:58 UTC Mon Apr 16 2001
! NVRAM config last updated at 12:06:13 UTC Sat Apr 14 2001
!
version 12.2
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname GWSCZ01
!
no boot startup-test
logging queue-limit 100

!
!
!
resource-pool disable
spe default-firmware spe-firmware-1
ip subnet-zero
ip cef
no ip domain lookup
!
isdn switch-type primary-net5
!
!
voice service voip
 fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
 sip
!
voice class codec 11
 codec preference 1 g729br8
 codec preference 2 g729r8
 codec preference 3 gsmfr
 codec preference 4 g726r32
 codec preference 6 g726r16
 codec preference 7 g723r63
 codec preference 8 g723r53
 codec preference 9 g726r24
 codec preference 10 g723ar63
 codec preference 11 g723ar53
 codec preference 12 g711ulaw
 codec preference 13 g711alaw
 codec preference 14 clear-channel
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
voice source-group cisco
 access-list 8
 carrier-id target cisco
!
!
!
fax interface-type fax-mail
mta receive maximum-recipients 0
!
!
!
controller E1 7/0
 framing NO-CRC4
 line-termination 75-ohm
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled
 cas-custom 0
  country bolivia
!
controller E1 7/1
 line-termination 75-ohm
 pri-group timeslots 1-31
!
controller E1 7/2
 line-termination 75-ohm
 pri-group timeslots 1-31
 description Embratel
 --More--
!
!
interface FastEthernet0/0
 ip address y.y.y.y 255.255.255.224
 duplex auto
 speed auto
 no cdp enable
 h323-gateway voip interface
 h323-gateway voip id GK01 ipaddr y.y.y.z 1719
 h323-gateway voip h323-id GWSCZ01
 h323-gateway voip tech-prefix 2032#
!
!
ip classless
ip route 0.0.0.0 0.0.0.0 y.y.y.v
no ip http server
!
!
!
!
!
!
call rsvp-sync
!
voice-port 7/0:0
 compand-type a-law
!
voice-port 7/1:D
!
voice-port 7/2:D
!
voice-port 7/3:0
 compand-type a-law
!
voice-port 7/4:0
 compand-type a-law
!
voice-port 7/5:0
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice  voip
 destination-pattern 44T
 voice-class codec 11
 session protocol sipv2
 session target sip-server
 session transport udp
!
dial-peer voice  pots
 destination-pattern T
 direct-inward-dial
 port 7/0:0
!
sip-ua
 retry invite 3
 retry cancel 2
 sip-server ipv4:x.x.x.x
!


On Sun, 2004-12-12 at 20:07, Hatzis, Michael wrote:
> What's the cisco box,52 / 53; version ios? can you post a config dump?
> 
> Regards
> 
>  
> 
> Michael Hatzis
> 
>  0421 476 211
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jorge
> Verastegui G
> Sent: Monday, 13 December 2004 10:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.
> 
> Excuse the insistence but I am more than one week with this problem, and
> I do not have any idea to solve it.
> 
> You know if the configuration with GK in the Cisco, can be interfering
> with the RTP traffic?  
> 
> 
> Thanks in advance
> 
> 
> 
> On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote:
> > Pls, post your Cisco and * config files.
> > 
> >  
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Jorge
> > Verastegui G
> > Sent: Friday, December 10, 2004 12:30 AM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.
> > 
> > Hi, 
> > 
> > I have a serious problem to configure Cisco AS5XXX and Asterisk , 
> > 
> > I trying to use asterisk for 
> > 
> > PSTN(A)> Cisco AS5xxx > ASterisk>PSTN(B) 
> > 
> > (No Nat, no Firewall)
> > 
> > I hear (on the PSTN(A)) clearly what the other person is saying, but
> the
> > other person (on the PSTN(B) side) hears nothing from PSTN(A).
> > 
> > I use tcpdump for debug de rtp trafic, and ouput contains 
> > 
> > 
> > 19:06:00.741293 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:00.763133 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:00.740415 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:00.810312 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:00.860314 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:00.980351 IP (

[Asterisk-Users] Using SPANDSP for faxes

[Eric Hall]

I installed spandsp on our asterisk server to get faxes. It works
however the images are a little off. Sometimes a few pages will be
together, pages missing and sentence missing.

Is this normal for this program? 

Any input would be great.



Thank You
Eric
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RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.

[Hatzis, Michael]

What's the cisco box,52 / 53; version ios? can you post a config dump?

Regards

 

Michael Hatzis

 0421 476 211

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jorge
Verastegui G
Sent: Monday, 13 December 2004 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.

Excuse the insistence but I am more than one week with this problem, and
I do not have any idea to solve it.

You know if the configuration with GK in the Cisco, can be interfering
with the RTP traffic?  


Thanks in advance



On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote:
> Pls, post your Cisco and * config files.
> 
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jorge
> Verastegui G
> Sent: Friday, December 10, 2004 12:30 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.
> 
> Hi, 
> 
> I have a serious problem to configure Cisco AS5XXX and Asterisk , 
> 
> I trying to use asterisk for 
> 
> PSTN(A)> Cisco AS5xxx > ASterisk>PSTN(B) 
> 
> (No Nat, no Firewall)
> 
> I hear (on the PSTN(A)) clearly what the other person is saying, but
the
> other person (on the PSTN(B) side) hears nothing from PSTN(A).
> 
> I use tcpdump for debug de rtp trafic, and ouput contains 
> 
> 
> 19:06:00.741293 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.763133 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.740415 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.810312 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.860314 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.980351 IP (tos 0x0, ttl  64, id 180, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.000313 IP (tos 0x0, ttl  64, id 181, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.014822 IP (tos 0x68, ttl 255, id 1, offset 0, flags [none],
> proto 17, length: 164) y.y.y.y.18975 > x.x.x.x.19927: UDP, length 136
> 19:06:01.020312 IP (tos 0x0, ttl  64, id 182, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.040302 IP (tos 0x0, ttl  64, id 183, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.060343 IP (tos 0x0, ttl  64, id 184, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.083311 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.128314 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.130316 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.165318 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.186312 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 
> 
> Where
> 
>  x.x.x.x = ip address of Astersik
>  y.y.y.y = ip address of Cisco
> 
> 
> Two types of codecs were proven ( ulow, g729 ).
> 
> When use the Asterisk with Sip phones everything works well.
>  
> SipPhone-->Asterisk--->PSTN(B) 
> 
> The configurations, are the usual ones (from the wiki). the version of
> asterisk is 1.0.3, the linux is  FC2.
> 
> 
> Please help me.  

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Re: [Asterisk-Users] [OT] Small SIP phones?

[Antony Stone]

On Sunday 12 December 2004 23:08, Shoval Tomer wrote:

> I may be wrong, but if you always carry your laptop around, why don't
> purchase a USB handset?

The main reason is that (I believe) the quality of audio with a soft phone is 
generally not as good as that from a real SIP phone?

The other reason is that I want to be able to show VoIP in operation to 
clients (which is where I would be taking the phone with me), so a standalone 
phone, which is not dependent on any software installed on my laptop, is a 
much neater arrangement.

> > -Original Message-
> > From: Florian Overkamp [mailto:[EMAIL PROTECTED]
> >
> > On Sun, 2004-12-12 at 21:27, Antony Stone wrote:
> > > Thanks.   In fact I already have a Grandstream ATA-486, which I'm
> > > very pleased with: http://www.grandstream.com/y-ht486.htm   This unit is
> > > even smaller - 105 x 75 x 25mm (or 4" x 2.75" x 1"), however I'm just
> > > wondering if there's a neat all-in-one solution, instead of carrying
> > > around two items? 
> >
> > Three, in fact. The powersupply also adds to the required space. This
> > is one of the biggest advantages of having an all-in-one solution, because
> > you don't have to generate high voltage/high power ring signalling.

True, you don't need the high voltage ringing, but with a standard SIP phone 
you still need a PSU for it.   I couldn't rely on a client's network switch 
supporting PoE for when I wanted to plug one in.

Regards,

Antony.

-- 
Wanted: telepath.   You know where to apply.

 Please reply to the list;
   please don't CC me.
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RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.

[Jorge Verastegui G]

Excuse the insistence but I am more than one week with this problem, and
I do not have any idea to solve it.

You know if the configuration with GK in the Cisco, can be interfering
with the RTP traffic?  


Thanks in advance



On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote:
> Pls, post your Cisco and * config files.
> 
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jorge
> Verastegui G
> Sent: Friday, December 10, 2004 12:30 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.
> 
> Hi, 
> 
> I have a serious problem to configure Cisco AS5XXX and Asterisk , 
> 
> I trying to use asterisk for 
> 
> PSTN(A)> Cisco AS5xxx > ASterisk>PSTN(B) 
> 
> (No Nat, no Firewall)
> 
> I hear (on the PSTN(A)) clearly what the other person is saying, but the
> other person (on the PSTN(B) side) hears nothing from PSTN(A).
> 
> I use tcpdump for debug de rtp trafic, and ouput contains 
> 
> 
> 19:06:00.741293 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.763133 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.740415 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.810312 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.860314 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.980351 IP (tos 0x0, ttl  64, id 180, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.000313 IP (tos 0x0, ttl  64, id 181, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.014822 IP (tos 0x68, ttl 255, id 1, offset 0, flags [none],
> proto 17, length: 164) y.y.y.y.18975 > x.x.x.x.19927: UDP, length 136
> 19:06:01.020312 IP (tos 0x0, ttl  64, id 182, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.040302 IP (tos 0x0, ttl  64, id 183, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.060343 IP (tos 0x0, ttl  64, id 184, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.083311 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.128314 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.130316 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.165318 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.186312 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 
> 
> Where
> 
>  x.x.x.x = ip address of Astersik
>  y.y.y.y = ip address of Cisco
> 
> 
> Two types of codecs were proven ( ulow, g729 ).
> 
> When use the Asterisk with Sip phones everything works well.
>  
> SipPhone-->Asterisk--->PSTN(B) 
> 
> The configurations, are the usual ones (from the wiki). the version of
> asterisk is 1.0.3, the linux is  FC2.
> 
> 
> Please help me.  

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[Asterisk-Users] BRI Problem dialing out

[Hatzis, Michael]

Hi All,

 

I have a slight problem when trying to dial out. When I dial
any number out I get only a dial tone and the number is not dialed I have to then
dial it manually. I have tried my extension.conf with my pstn box and it works
fine but for some reason it won’t with the isdn card. I’m using the
fritz pci card. Has any one else had this problem in the past??. I have also
tried to set up an extension that will open the line then SendDTMF of a number
to dial. “no luck”

 

 

Can any one help

 

 

Regards

 

Michael Hatzis

 

 






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[Asterisk-Users] zaptel 0.9.1 compile problem

[Jay Austad]

I'm using gentoo 2004.3, and when I emerge the zaptel driver, compile 
fails with the following output:

  CC [M]  /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o
In file included from 
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:40:
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/zaptel.h:55:2: warning: 
#warning "Zaptel doesn't support DEVFS in post 2.4 kernels.  Disabling 
DEVFS in zaptel"
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c: In function 
`ztdeth_rcv':
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:89: error: 
union has no member named `ethernet'
make[2]: *** 
[/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o] Error 1
make[1]: *** [_module_/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1] 
Error 2
make[1]: Leaving directory `/usr/src/linux-2.6.9-gentoo-r9'
make: *** [linux26] Error 2


Any ideas why this would be?
~jay
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RE: [Asterisk-Users] PoE VOIP phones in Australia

[Paul Hales]

The Polycom IP600 is fairly available in Australia (at least in Melbourne)

Regards,

PaulH 

-Original Message-
From: James Andrewartha [mailto:[EMAIL PROTECTED] 
Sent: Friday, 10 December 2004 6:32 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PoE VOIP phones in Australia

Hi,

Are there any resellers of phones that can take power over ethernet in
Australia? All I can find for sale online is the BT-10[12], which is cheap
but not featureful enough, and the Snom 190, which is about right, but
neither of them support PoE. I'm particularly intereseted in the Snom 220
with the keypad expansion for our receptionist.

Although, could you make a PoE split-out cable for the Snom 190?

James Andrewartha
DAA Sysadmin
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RE: [Asterisk-Users] [OT] Small SIP phones?

[Shoval Tomer]

I may be wrong, but if you always carry your laptop around, why don't
purchase a USB handset?
It'll give you a mic and phones in one handset, and installs as a sound
card, so it can ring and you don't have to put it next to your ear (the
problem with using head phones is that once they're on the desk and not
on your head you might miss some phone calls).

Some of these come with a softphone, and some can use your softphone of
choice.

As far as I know, their not cheaper the full fledged VOIP phones, but
it'll be the smallest option

> -Original Message-
> From: Florian Overkamp [mailto:[EMAIL PROTECTED]
> Sent: Monday, December 13, 2004 12:21 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] [OT] Small SIP phones?
> 
> Hi
> 
> On Sun, 2004-12-12 at 21:27, Antony Stone wrote:
> > Thanks.   In fact I already have a Grandstream ATA-486, which I'm
very
> pleased
> > with: http://www.grandstream.com/y-ht486.htm   This unit is even
smaller
> -
> > 105 x 75 x 25mm (or 4" x 2.75" x 1"), however I'm just wondering if
> there's a
> > neat all-in-one solution, instead of carrying around two items?
> 
> Three, in fact. The powersupply also adds to the required space. This
is
> one of the biggest advantages of having an all-in-one solution,
because
> you don't have to generate high voltage/high power ring signalling.
> 
> Florian
> 
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Re: [Asterisk-Users] Can't capture "-1" return on Dial command

[Eric Bullen]

>> following setup, and I want to be able to process the audio file after
>> the
>> outbound call has been done regardless how how it ends.
>
> would the hangup priority be appropriate for this?
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>
>
>

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Re: [Asterisk-Users] [OT] Small SIP phones?

[Florian Overkamp]

Hi

On Sun, 2004-12-12 at 21:27, Antony Stone wrote:
> Thanks.   In fact I already have a Grandstream ATA-486, which I'm very 
> pleased 
> with: http://www.grandstream.com/y-ht486.htm   This unit is even smaller - 
> 105 x 75 x 25mm (or 4" x 2.75" x 1"), however I'm just wondering if there's a 
> neat all-in-one solution, instead of carrying around two items?

Three, in fact. The powersupply also adds to the required space. This is
one of the biggest advantages of having an all-in-one solution, because
you don't have to generate high voltage/high power ring signalling.

Florian

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Re: [Asterisk-Users] TDM400P FXS polarity reversal?

[Soren Rathje]

Rich Adamson wrote:
>> Is it possible to do a talk battery polarity reversal on a TDM400P
>> FXS interface?  Everything I can find seems to be referring to the
>> procedure for detecting a battery reversal on a telephone company
>> POTS line using the FXO interface, but not for actually generating
>> one back to a station upon answer supervision.  I would assume that
>> VoicePulse and VoipJet provide a way of signaling far-end
>> supervision back to the originating Asterisk PBX...
>>
>> Basically, my two questions are:
>> (1) Is the hardware capable of even performing a reversal?
>> (2) If the above is true, how would you make it happen in Asterisk?
>
> The Silicon Labs spec sheet does not specifically indicate generating
> a reversal is possible. Therefore, best guess is the integrated
> circuits on the card does not support it, therefore asterisk has no
> means of doing it.
>

Specs for Si3210 (TDM400P FXS Module) says on page 93:

---
Register 72. On-Hook Line Voltage

Bit 6 VSGN On-Hook Line Voltage.
The value written to this bit sets the on-hook line voltage polarity
(VTIP–VRING).
  0 = VTIP–VRING is positive
  1 = VTIP–VRING is negative
---

I wonder if this can control Pol-Rev's.. ?

/Soren

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RE: [Asterisk-Users] four wildcards in a single pc

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
> Hi.
> Please excuse me asking this again. But it really puzzles me.

It is puzzling, no denying it. The development team is still struggling
with these issues, and so far there has not been found a foolproof
solution (at least I can't recall having seen one).

> We're installing asterisk at a branch office at NJ (HQ is at
> Petach-Tikva) It'll need to support 5 POTS lines, 11 analog
> extensions and four VOIP phones. 
> 
> I wanted to go with a T1 card from digium and a channel bank,
> but we have a dead line. It has to be up and running by
> January 1st. I don't have the time to start shopping at ebay,
> where you don't know what you'll get, and you need to
> install, under time pressure something you not familiar with.

For sure, but you should also consider the experience of those who have
been there before you.

> So I thought of installing a combination of four pci cards in
> the machine, and everybody on the list just keeps telling me it won't
> work. 

It _might_ work, but it is almost guaranteed not to work _well_. The
Digium PCI cards are rather different from any PCI card you may have
used in the past.

> I have installed successfully more then four cards in a
> machine before. I had a firewall with eight network
> interfaces (one quad card, one duo and two singles) I have
> machines with two dialogic boards, a pci display card, and a
> network interface. And I know I've had machines at home that
> had a display adaptor, modem, network, scsi, and soundblaster all
> together. 

Yep, so have we all.

The thing is, just ONE of these Digium cards will request more
interrupts all by its lonesome than every single PCI card you've ever
installed in all the machines you've ever owned, all put together! Well,
perhaps not, but seriously, these things work very differently from any
PCI card you've ever seen before.

> Yet, people claim it won't work because of lack of IRQs, and
> that it's not related to Digium.

That isn't strictly correct, but the problem does pertain to IRQs. 

OK, look, you _might_ be able to free up enough IRQs on a PIC-based
motherboard -- if you disable the serial ports, mouse, parallel port and
USB. It's not recommended, but it's theoretically possible. 

And if you have a MoBo that is APIC-compliant, you should be able to
have all the IRQs you can handle, so lack of IRQs doesn't need to be an
issue (make sure you have a BIOS and chipset that's up to the task).

BUT . . . 

Getting dedicated IRQs for the cards is a minor problem compared to what
happens when you have four cards hammering away mercilessly at the
chipset and CPU of your motherboard; 1000 IRQs per second, per card.
Nobody's really sure what's wrong, but it causes problems for pretty
nearly everyone.

What everyone here is saying is that we're all pretty sure you're gonna
run into problems; problems that could easily be avoided by avoiding the
whole TDM400 mess in the first place.

> What am I missing?

The Digium cards are unique in the world of telephony, because instead
of having an expensive DSP chip on board, they use the CPU to provide
this functionality. The challenge comes from the fact that voice is
intolerant of delay. In order to ensure that the voice processing that
goes on in the CPU is handled with no perceivable delay, the zaptel
cards have to establish a kind of pseudo-synchronous clocking with the
CPU. Unfortunately, the signalling bus on a PC isn't synchronous, at
least not in that way. The "clock" that the zaptel cards use is the IRQ
of the card, literally requesting the CPU interrupt what it's doing and
pay attention to it 1000 times per second, regardless of what it's
doing. You are proposing the use of FOUR of these cards. 

Since this has caused trouble for nearly everyone who has tried it,
everyone is suggesting that you might want to give the matter some
careful thought. There _are_ less painful ways.

Cheers,

Jim.


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RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.

[Jorge Verastegui G]

Hi, 

I already tested those (ulow and  g729) , and the rtp.conf  

[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=16384
rtpend=2
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
~


Best regards,

Jorge

On Sun, 2004-12-12 at 16:54, Henry Devito wrote:
> voice class codec 11
>  codec preference 1 g729br8
>  codec preference 2 g729r8
>  codec preference 3 gsmfr
>  codec preference 4 g726r32
>  codec preference 6 g726r16
>  codec preference 7 g723r63
>  codec preference 8 g723r53
>  codec preference 9 g726r24
>  codec preference 10 g723ar63
>  codec preference 11 g723ar53
>  codec preference 12 g711ulaw
>  codec preference 13 g711alaw
>  codec preference 14 clear-channel
> 
> Why so many codecs listed in class 11?  Asterisk can only use ulaw,alaw, or
> gsm unless licensed.  Try only listing g711ulaw for testing purposes.
> 
> 
> Make sure you have all of the correct ports for RTP listed in your rtp.conf
> file also. 
> 
> Just a thought or two.
> 
> 
> 
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[Asterisk-Users] IAXPeerGraph - a beta of another windows monitor app

[Matt Riddell]

We've just completed another Windows monitor app. This one has a 
scrolling taskman-like interface.

Once again the zip file just contains the .exe file and the INSTALL.txt 
file.

Oh, and by the way, the blue light that flashes next to the green 
connect light (it is black in the picture) toggles between blue and 
black whenever it receives a response from the simple manager proxy with 
IAX peer information inside of it. Note that if you have multiple 
clients connecting to the same server, the pulses sent out by 
IAXPeerGraph and IAXPeers will be received by all peer clients.

You can't really tell from the picture, but the line will change colour 
depending on how high the ping is (i.e. if it is low, the colour will be 
green, med - yellow, bad red).  This is kinda experimental because it is 
continuously variable - let me know how it goes for you.

You can download it directly from 
http://www.sineapps.com/down/IAXPeerGraph.zip or via the main page of 
the daily news (as listed in my sig)

If you need any help or would like some changes made, please don't 
hesitate to contact us.

Also, if anyone could take a screen shot (press print screen key, paste 
into windows paint etc) of it running in Windows XP and mail it to me, 
it would be much appreciated.

--
Cheers,
Matt Riddell
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RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.

[Jorge Verastegui G]

Hi,
thanks for your help .

here is the cisco config

GWSCZ01>en
Password:
GWSCZ01#sh run
Building configuration...

Current configuration : 5053 bytes
!
! Last configuration change at 05:17:58 UTC Mon Apr 16 2001
! NVRAM config last updated at 12:06:13 UTC Sat Apr 14 2001
!
version 12.2
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname GWSCZ01
!
no boot startup-test
logging queue-limit 100

!
!
!
resource-pool disable
spe default-firmware spe-firmware-1
ip subnet-zero
ip cef
no ip domain lookup
!
isdn switch-type primary-net5
!
!
voice service voip
 fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
 sip
!
voice class codec 11
 codec preference 1 g729br8
 codec preference 2 g729r8
 codec preference 3 gsmfr
 codec preference 4 g726r32
 codec preference 6 g726r16
 codec preference 7 g723r63
 codec preference 8 g723r53
 codec preference 9 g726r24
 codec preference 10 g723ar63
 codec preference 11 g723ar53
 codec preference 12 g711ulaw
 codec preference 13 g711alaw
 codec preference 14 clear-channel
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
voice source-group cisco
 access-list 8
 carrier-id target cisco
!
!
!
fax interface-type fax-mail
mta receive maximum-recipients 0
!
!
!
controller E1 7/0
 framing NO-CRC4
 line-termination 75-ohm
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled
 cas-custom 0
  country bolivia
!
controller E1 7/1
 line-termination 75-ohm
 pri-group timeslots 1-31
!
controller E1 7/2
 line-termination 75-ohm
 pri-group timeslots 1-31
 description Embratel
 --More--
!
!
interface FastEthernet0/0
 ip address y.y.y.y 255.255.255.224
 duplex auto
 speed auto
 no cdp enable
 h323-gateway voip interface
 h323-gateway voip id GK01 ipaddr y.y.y.z 1719
 h323-gateway voip h323-id GWSCZ01
 h323-gateway voip tech-prefix 2032#
!
!
ip classless
ip route 0.0.0.0 0.0.0.0 y.y.y.v
no ip http server
!
!
!
!
!
!
call rsvp-sync
!
voice-port 7/0:0
 compand-type a-law
!
voice-port 7/1:D
!
voice-port 7/2:D
!
voice-port 7/3:0
 compand-type a-law
!
voice-port 7/4:0
 compand-type a-law
!
voice-port 7/5:0
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice  voip
 destination-pattern 44T
 voice-class codec 11
 session protocol sipv2
 session target sip-server
 session transport udp
!
dial-peer voice  pots
 destination-pattern T
 direct-inward-dial
 port 7/0:0
!
sip-ua
 retry invite 3
 retry cancel 2
 sip-server ipv4:x.x.x.x
!
-
Where

  y.y.y.z = ip address of gk h323
  y.y.y.v = ip default gateway
  x.x.x.x = ip address of Astersik
  y.y.y.y = ip address of Cisco


And this is the asterisk configuration

sip.conf
[general]
context=default
port=5060
bindaddr=x.x.x.x
srvlookup=yes
videosupport=no

[y.y.y.y]
type=user
host=y.y.y.y
canreinvite=no
context=fromsip
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ulaw
allow=alaw


--
 extensions.conf

[fromsip]
exten => _X.,1,Dial(Zap/g2/${EXTEN:2})



I'd be very grateful  for your help on this matter
Best regards,

Jorge

On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote:
> Pls, post your Cisco and * config files.
> 
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jorge
> Verastegui G
> Sent: Friday, December 10, 2004 12:30 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.
> 
> Hi, 
> 
> I have a serious problem to configure Cisco AS5XXX and Asterisk , 
> 
> I trying to use asterisk for 
> 
> PSTN(A)> Cisco AS5xxx > ASterisk>PSTN(B) 
> 
> (No Nat, no Firewall)
> 
> I hear (on the PSTN(A)) clearly what the other person is saying, but the
> other person (on the PSTN(B) side) hears nothing from PSTN(A).
> 
> I use tcpdump for debug de rtp trafic, and ouput contains 
> 
> 
> 19:06:00.741293 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.763133 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.740415 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.810312 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.860314 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:00.980351 IP (tos 0x0, ttl  64, id 180, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.000313 IP (tos 0x0, ttl  64, id 181, offset 0, flags [DF],
> proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> length 32
> 19:06:01.014822 IP (tos 0x68, ttl 255, id 1, offset 0, flags [none],
>

RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.

[Henry Devito]

voice class codec 11
 codec preference 1 g729br8
 codec preference 2 g729r8
 codec preference 3 gsmfr
 codec preference 4 g726r32
 codec preference 6 g726r16
 codec preference 7 g723r63
 codec preference 8 g723r53
 codec preference 9 g726r24
 codec preference 10 g723ar63
 codec preference 11 g723ar53
 codec preference 12 g711ulaw
 codec preference 13 g711alaw
 codec preference 14 clear-channel

Why so many codecs listed in class 11?  Asterisk can only use ulaw,alaw, or
gsm unless licensed.  Try only listing g711ulaw for testing purposes.


Make sure you have all of the correct ports for RTP listed in your rtp.conf
file also. 

Just a thought or two.



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Re: [Asterisk-Users] Digium Card Error

[Charles S. Antrim]

Thanks Lee

-Original Message-
From: Lee <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Date: Sun, 12 Dec 2004 12:26:15 -0800
Subject: Re: [Asterisk-Users] Digium Card Error

> On Sun, 12 Dec 2004 09:00:50 -0600, Steven Critchfield
> <[EMAIL PROTECTED]> wrote:
> > This is where reading the mailing list is important. We just covered
> > that for a person Friday. Look in /etc/modprobe.conf and add the
> modules
> > you need.
> 
> Actually, the two fixes that worked to solve this were:
> 
>   1. Add the following to /etc/rc.d/rc.local:
>/sbin/modprobe wcfxo
> or
>   2.  cd to /usr/src/zaptel-1.0.3 and do a "make config"
> 
> -- 
> Lee
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Re: [Asterisk-Users] Pattern-matching in the dial-plan

[Peter Svensson]

On Sun, 12 Dec 2004, Wilson Pickett wrote:

> > dialled is a "#" or not.  It's as if the parser assumes everything
> > after the "." will match and doesn't look any further.  Is this expected
> > behaviour?  

> Yes, the dot says match ANYTHING from here on AFAIK

To be precise it will match one or more digits, ognoring the rest of the
pattern as well. There is no match zero-or-more wildcard and no wildcard
that tries to continue to read the pattern after the wildcard.

The former can easily be implemented, we had to do it to handle some cases 
of overlap dialing. We'll clean it up and submit it later.

The latter case could probably be implemented in the 
  ast_extension_match
  ast_extension_close
  EXTENSION_MATCH_CORE
functions in pbx.c. Alternativly a regular or extended regexp could be 
added as a an extension switch, similar to pbx_loopback.c. Yet another 
option would be to integrate it with the res_perl in asterisk-addons.

Peter


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RE: [Asterisk-Users] [OT] Small SIP phones?

[Clay Reiche]

I don't know of a small phone, but you use a WorlACCXX TA200 device (pretty
small) along with any standard analogue phone.
http://www.worldaccxx.com I have one and carry it around in my laptop bag.
Demensions are 6"x4.5"x1.25"

Clay

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antony Stone
Sent: Sunday, December 12, 2004 3:01 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] [OT] Small SIP phones?

Hi.

Does anyone know of any small SIP phones (and preferably have some experience
of using them and happy to recommend them)?

By 'small' I mean a single-piece phone, with dial buttons in the handset, so 
that it can be carried around easily in a laptop bag.   Something like 
http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a
standard analogue telephone).

Ideally I'd like something without a cradle, which can simply be put on a
desk and answered by picking it up.

Thanks,

Antony.

--
"Linux is going to be part of the future. It's going to be like Unix was."

 - Peter Moore, Asia-Pacific general manager, Microsoft
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[Asterisk-Users] I'm stumped

[Henry Devito]

I am trying to use the simple CID name management script on the wiki.
http://www.voip-info.org/wiki-Asterisk+tips+managing+CID+names   I can not
see what is wrong.  The values never get entered in the database.  Here are
the files:  I have asterisk running as the user asterisk also.

---cid-store.php



Storing Asterisk CID data


Asterisk phone book
 "" && $PhoneName <> "") {
   system("sudo -u asterisk /usr/sbin/asterisk -rx " .
escapeshellarg("database put cidname $PhoneNumber \"$PhoneName\"") . " &>
/tmp/error");
  print "Successfully stored $PhoneNumber as $PhoneName.";
} else {
   print "Please enter both phone number and name!";
}
?>



-My sudoers files-
asterisk  ALL=(ALL) NOPASSWD: /usr/sbin/asterisk





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Re: [Asterisk-Users] [OT] Small SIP phones?

[Antony Stone]

On Sunday 12 December 2004 20:12, Clay Reiche wrote:

> I don't know of a small phone, but you use a WorlACCXX TA200 device (pretty
> small) along with any standard analogue phone.
> http://www.worldaccxx.com I have one and carry it around in my laptop bag.
> Demensions are 6"x4.5"x1.25"

Thanks.   In fact I already have a Grandstream ATA-486, which I'm very pleased 
with: http://www.grandstream.com/y-ht486.htm   This unit is even smaller - 
105 x 75 x 25mm (or 4" x 2.75" x 1"), however I'm just wondering if there's a 
neat all-in-one solution, instead of carrying around two items?

Regards,

Antony.

> -Original Message-
>
> Hi.
>
> Does anyone know of any small SIP phones (and preferably have some
> experience of using them and happy to recommend them)?
>
> By 'small' I mean a single-piece phone, with dial buttons in the handset,
> so that it can be carried around easily in a laptop bag.   Something like
> http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a
> standard analogue telephone).
>
> Ideally I'd like something without a cradle, which can simply be put on a
> desk and answered by picking it up.

-- 
Never automate fully anything that does not have a manual override capability. 
Never design anything that cannot work under degraded conditions in emergency.

 Please reply to the list;
   please don't CC me.
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Re: [Asterisk-Users] Digium Card Error

[Lee]

On Sun, 12 Dec 2004 09:00:50 -0600, Steven Critchfield
<[EMAIL PROTECTED]> wrote:
> This is where reading the mailing list is important. We just covered
> that for a person Friday. Look in /etc/modprobe.conf and add the modules
> you need.

Actually, the two fixes that worked to solve this were:

1. Add the following to /etc/rc.d/rc.local:
 /sbin/modprobe wcfxo
or
2.  cd to /usr/src/zaptel-1.0.3 and do a "make config"

-- 
Lee
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[Asterisk-Users] [OT] Small SIP phones?

[Antony Stone]

Hi.

Does anyone know of any small SIP phones (and preferably have some experience 
of using them and happy to recommend them)?

By 'small' I mean a single-piece phone, with dial buttons in the handset, so 
that it can be carried around easily in a laptop bag.   Something like 
http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a 
standard analogue telephone).

Ideally I'd like something without a cradle, which can simply be put on a desk 
and answered by picking it up.

Thanks,

Antony.

-- 
"Linux is going to be part of the future. It's going to be like Unix was."

 - Peter Moore, Asia-Pacific general manager, Microsoft
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Re: [Asterisk-Users] TDM400P FXS polarity reversal?

[Soren Rathje]

Soren Rathje wrote:
>
> Specs for Si3210 (TDM400P FXS Module) says on page 93:
>
> ---
> Register 72. On-Hook Line Voltage
>
> Bit 6 VSGN On-Hook Line Voltage.
> The value written to this bit sets the on-hook line voltage polarity
> (VTIP–VRING).
>   0 = VTIP–VRING is positive
>   1 = VTIP–VRING is negative
> ---
>

The (missing) link..

https://www.mysilabs.com/public/documents/tpub_doc/dsheet/Wireline/ProSLIC/en/si3210.pdf

Features
...
...
* Software programmable signal generation and audio processing:
  - DTMF generation and decoding
  - 12 kHz/16 kHz pulse metering generation
  - Phase-continuous FSK (caller ID) generation
  - Dual audio tone generators
  - Smooth and abrupt polarity reversal  <<== NB!
  - µ-Law/A-Law and 16-bit linear PCM audio
...
...

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Re: [Asterisk-Users] Many similar contexts - can I use Macro or some other template concept ?

[Wilson Pickett]

> Are there any other ways of context templates filled with data in dialplan ?

Rob you really should read some of the beginning material to find this
stuff out. Here is a great article for the basic concepts (including
what macros are for IIRC):

http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html

Here is a site with documents that have answers to all the questions
you are likely to ask in the next few days:

http://asteriskdocs.org

Read online or download the latest PDF and read it through a few
times. That's why they spent all the time and effort writing,
perfecting and putting this online.

There are many, many other things you could read to get up to speed on this.
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Re: [Asterisk-Users] Can't capture "-1" return on Dial command

[Wilson Pickett]

> following setup, and I want to be able to process the audio file after the
> outbound call has been done regardless how how it ends.

would the hangup priority be appropriate for this?
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Re: [Asterisk-Users] MySQL

[Greg - Cirelle Enterprises]

At 06:29 PM 12/9/04, you wrote:
Sure. (I really need to write a wiki on this.)
You have two choices here before we start. You can use RealTime one of 2
ways: ODBC or direct MySQL. Currently these are the only two supported
methods.
Since I don't use ODBC and as the author of the MySQL RealTime driver, I'm
going to instruct on how to use/install it.
The RealTime MySQL driver can be found inside asterisk-addons. Just do the
standard make, make install.
Now copy asterisk-addons/configs/res_mysql.conf.sample to
/etc/asterisk/res_mysql.conf (or whereever your conf dir is).
Edit the res_mysql.conf to your liking.
Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime
config stuff. If you want voicemail, add this line:
voicemail => mysql,asterisk,voicemail_users
No such file res_mysql.conf
only cdr_mysql_conf.sample
Greg
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Re: [Asterisk-Users] Totally LOST with dialplan and Extensions.

[Wilson Pickett]

> ; outbound
> ; Firefly (Freshtel)
> [89280250] ; Firefly
> context=89280250

Where is this context? If you change it to default, it should work if
the rest is right.

Otherwise, post what you see as console messages when you try to dial.
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Re: [Asterisk-Users] TDM400P FXS polarity reversal?

[Rich Adamson]

> Is it possible to do a talk battery polarity reversal on a TDM400P FXS
> interface?  Everything I can find seems to be referring to the
> procedure for detecting a battery reversal on a telephone company POTS
> line using the FXO interface, but not for actually generating one back
> to a station upon answer supervision.  I would assume that VoicePulse
> and VoipJet provide a way of signaling far-end supervision back to the
> originating Asterisk PBX...
> 
> Basically, my two questions are:
> (1) Is the hardware capable of even performing a reversal?
> (2) If the above is true, how would you make it happen in Asterisk?

The Silicon Labs spec sheet does not specifically indicate generating
a reversal is possible. Therefore, best guess is the integrated circuits
on the card does not support it, therefore asterisk has no means of
doing it.



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[Asterisk-Users] Re: Cant set H323 up

[Corvin]

Rafael J. Risco G.V. wrote:


> 
> On Sat, 11 Dec 2004 16:49:12 +, Corvin <[EMAIL PROTECTED]> wrote:
>> Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisał:
>> > Hi.
>> >
>> > I need to set up H323 on an Asterisk box. I've succesfuly compiled the
>> > asterisk oh323 (including of course all the dependencies: PWlib and
>> > OpenH323), and then compiled asterisk. However, asterisk doesn't report
>> > a registered H323 channel (when it starts, it reports IAX2, ZAP and SIP
>> > channels, however, the 323 word doesn't appear in the whole output).
>> >
>> > Is there anything I'm missing? I've read the documentation on the wiki,
>> > and none said nothing about editing a config file. I did noticed that
>> > they talked about the oh323.conf file, which I dont have.
>> 


BTW. you should check direcory with oh323 a there should be asterisk-driver 
directory and there you find sample config.
Then you sghould load module in modules.conf.

BTW. I can't still compile any h323 driver :(((.

Corvin 
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RE: [Asterisk-Users] Will Adtran TSU 600 work with *?

[Robert Augustyn]

Let me know how it works for you.
Thanks
robertShoval Tomer <[EMAIL PROTECTED]> wrote:









People on the list tend to think you can’t make many cards work on a regular desktop.
 
If you’re willing to wait a couple of week I might have an answer for you.
 





From: Robert Augustyn [mailto:[EMAIL PROTECTED] Sent: Saturday, December 11, 2004 7:13 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Will Adtran TSU 600 work with *?
 


Hi, 
I am looking at getting adtran tsu 600 p/n 1200.076L2 for my small office
It comes with 6 FXS ports and I would use 2 X100Ps for FXO ports.
Would that work ? Is there anything I would have to be aware of in such configuration?
What would be a better solution?
robert
 
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Re: SV: [Asterisk-Users] How to Playback Mailbox Owners Name?

[Howard Lowndes]

On Mon, 2004-12-13 at 02:10, Thorben G. Jensen wrote:
> -Oprindelig meddelelse-
> Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] PÃ vegne af Howard Lowndes
> Sendt: 12. december 2004 13:07
> Til: Asterisk Users Mailing List - Non-Commercial Discussion
> Emne: Re: [Asterisk-Users] How to Playback Mailbox Owners Name?
> 
> On Sun, 2004-12-12 at 21:45, Thorben G. Jensen wrote:
> > How do I Playback the Mailbox Owners Name?
> > 
> >  
> > 
> > Ex.: I want a message saying "I am sorry but" + Mailbox Owner Name +
> > "has gone to lunch"
> 
> You could get them to record their temp message in the voicemail
> services; option 0, IIRC.
> 
> I understand that, but they all have recorded their name and I just would 
> like to use that recording.
> 

Well, record pre-greet.gsm and post-greet.gsm, then:

cat pre-greet.gsm greet.gsm post-greet.gsm >temp.gsm

Bingo - your personalised temp greeting message.

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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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[Asterisk-Users] TDM400P FXS polarity reversal?

[Strom Carlson]

Hello all,

Is it possible to do a talk battery polarity reversal on a TDM400P FXS
interface?  Everything I can find seems to be referring to the
procedure for detecting a battery reversal on a telephone company POTS
line using the FXO interface, but not for actually generating one back
to a station upon answer supervision.  I would assume that VoicePulse
and VoipJet provide a way of signaling far-end supervision back to the
originating Asterisk PBX...

Basically, my two questions are:
(1) Is the hardware capable of even performing a reversal?
(2) If the above is true, how would you make it happen in Asterisk?

-- 
Strom Carlson
http://www.stromcarlson.com/
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[Asterisk-Users] ACK from asterisk not matched to transaction by SER / LCS2005

[Public Dump]

Thanks for the info !
Is there any way to work around the bug, maybe by rewriting the SIP
Message in SER ? 
Or some kind to temporary third party patch ?

Chris.

--

Message: 9
Date: Sun, 12 Dec 2004 10:34:53 +0100
From: "Olle E. Johansson" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] ACK from asterisk not matched to
transaction by  SER / LCS2005
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[EMAIL PROTECTED]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Public Dump wrote:
> For reasons unknown to me, SER and subsequently a Microsoft Live 
> Communcations Server 2005 seems to have problems, matching a SIP ACK 
> request from asterisk to the ongoing SIP transaction, I have attached 
> the complete log, but the essential lines are:
>  
That's a bug in Asterisk that is in the bug tracker and needs to be
fixed. Asterisk is sending the ACK to the AOR in the invite, not to the
contact address in the 200 OK.

Regards,
/Olle
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Re: [Asterisk-Users] How to Playback Mailbox Owners Name?

[Soren Rathje]

Thorben G. Jensen wrote:
> How do I Playback the Mailbox Owners Name?
>
>
>
> Ex.: I want a message saying "I am sorry but" + Mailbox Owner Name +
> "has gone to lunch"
>

Extension "999" in voicemail context "internal"

exten => 999,1,SetVar(VM_CONTEXT=internal)
exten => 999,2,Playback(im-sorry)
exten =>
999,3,Playback(/var/spool/asterisk/voicemail/${VM_CONTEXT}/${EXTEN}/greet)
exten => 999,4,Playback(flagged-for-lea)
exten => 999,5,Hangup()

That should get you going.. :-)

/Søren

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RE: [Asterisk-Users] can a TDM400P FXS drop voltage on hangup?

[Henry Devito]

 

 

 

>But if I could get the FXS to drop
voltage instead of play Congestion (or a second of Congestion in case a person
is listening, and then drop voltage) that >would be even simpler.  But can I
make that happen, and how?

 

I have the same setup at one of my sites,  I
tried to make the FXS drop voltage on a hangup , but I couldn’t make it
happen, seems like the talk voltage is always there on the fxs port.  I
ended up finding a setting in the legacy pbx in the CO line setup for the
disconnect supervision and set it for not received, this seemed to help a lot. 

 






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Re: [Asterisk-Users] Digium Card Error

[Steven Critchfield]

On Sun, 2004-12-12 at 00:36 -0800, Charles S. Antrim wrote:
> I have success installing and compiling, but if I reboot I have to modprobe 
> again to get he 
> drivers loaded for the module I am using.  I am using rhes31 and a tdm card 
> with one fxo and 
> one fxs.

This is where reading the mailing list is important. We just covered
that for a person Friday. Look in /etc/modprobe.conf and add the modules
you need.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] Totally LOST with dialplan and Extensions.

[David Uzzell]

Ok I have spent the last week working on getting my small PBX to work.
I will in the end only have 4 SIP extensions being either softphones of 
IP phones. Currently only 1 SIP config for testing.

And at the this point it should be all fairly easy with all inbound and 
outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via 
IAX. Inbound does work in it's current basic state.

There will be NO ZAP devices, so I have ztdummy running.
I would say that for the outbound dialing I have either missed out 
something plainly obvious or a simple typo which would be the challange.

I would think that all the problems are in the extensions.conf file 
which really has me confused and totally lost.

I don't expect answers just pointers in the correct direction so that I 
can get it to work for the outbound calling to work, I have the inbound 
working which was a task but I was able with some pointers to get it 
working.

I would like to thank you all for your casting experianced eyes to look 
over this. What ever is worked out I will make sure the info gets onto 
the Wiki for Freshtel and for a SIP to IAX to PSTN config so that others 
can look up the basic configs to do this type of setup. There does not 
seem to be from what I can find this basic configs for IAX without FXS & 
FXO devices.

cheers
David
SIP.CONF
[general]
context=default
realm=monitor.diversified.com.au
bindaddr=203.29.98.221
srvlookup=yes
maxexpirey=180
defaultexpirey=160
disallow=all
allow=speex
allow=gsm
allow=ilbc
allow=ulaw
allow=ilbc
[801]
type=friend
regexten=801
username=801
secret=password
callerid=801
host=dynamic
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=speex
allow=ulaw
allow=alaw
IAX.CONF
[general]
tos=lowdelay
jitterbuffer=no
disallow=all
allow=speex
allow=ilbc
allow=gsm
allow=adpcm
allow=alaw
register => 89280250:[EMAIL PROTECTED]
register => 89280250:[EMAIL PROTECTED]
[guest]
type=user
context=default
auth=none
;inbound
[firefly]
type=friend
host=cts-au.freshtel.net
context=default
; outbound
; Firefly (Freshtel)
[89280250] ; Firefly
context=89280250
qualify=no
username=89280250
secret=password
auth=md5
type=friend
host=gateway.freshtel.net
EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
[globals]
SpeakingClock=123
[default]
exten => s,1,Wait,1
exten => s,n,Answer
exten => s,n,DigitTimeout,5
exten => s,n,ResponseTimeout,10
exten => s,n,WaitExten
exten => s,n,Dial(SIP/801)
exten => 13,1,DateTime()
exten => 13,2,Wait(1)
exten => 13,3,DateTime()
exten => 13,4,Hangup
exten => t,1,Goto(#,1)
exten => i,1,Playback(invalid)
exten => 600,1,Playback(demo-echotest)
exten => 600,n,Echo
exten => 600,n,Playback(demo-echodone)
exten => 600,n,Goto(s,6)
exten => ${SpeakingClock},1,Wait(1)
exten => ${SpeakingClock},2,setvar(FutureTime=$[${EPOCH} + 10])
exten => ${SpeakingClock},3,Wait,3
exten => ${SpeakingClock},4,SayUnixTime(${FutureTime},,R)
exten => ${SpeakingClock},5,playback(vm-and)
exten => ${SpeakingClock},6,SayUnixTime(${FutureTime},,S)
exten => ${SpeakingClock},7,playback(seconds)
exten => ${SpeakingClock},8,playback(beep)
exten => ${SpeakingClock},9,wait(2)
exten => ${SpeakingClock},10,goto(1)
exten => _394.,1,SetCallderId(89280250)
exten => 
_394.,2,Dial(IAX2/89280250:[EMAIL PROTECTED]/${EXTEN:3},60,r)

[outgoing-firefly-peers]
exten => _62,1,Macro(outgoingfirefly,${EXTEN:2},70) ; Firefly
[macro-outgoingfirefly]
exten => s,1,SetCallerID("89280250" <89280250>)
exten => 
s,2,Dial(IAX2/89280250:[EMAIL PROTECTED]/${ARG1},${ARG2},r)
exten => s,3,Congestion

[macro-outgoingfreshtel]
exten => s,1,SetCallerID("89280250" <89280250>)
exten => 
s,2,Dial(IAX2/89280250:[EMAIL PROTECTED]/${ARG1},${ARG2},r)
exten => s,3,Congestion

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Re: [Asterisk-Users] Pattern-matching in the dial-plan

[Wilson Pickett]

> dialled is a "#" or not.  It's as if the parser assumes everything
> after the "." will match and doesn't look any further.  Is this expected
> behaviour?  
Yes, the dot says match ANYTHING from here on AFAIK
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[Asterisk-Users] can a TDM400P FXS drop voltage on hangup?

[Warren Burstein]

I
thought I had posted this, but I didn’t see it in the archives, so I
guess I hadn’t.

 

I’ve
got FXS lines going to a legacy IVR.  When I Dial into one of these lines
and then hang up, FXS plays the Congestion tone until the IVR drops voltage.   I
would like the IVR to hang up sooner.  I could do this by either making
the IVR recognize the standard Congestion tone, or changing the Congestion tone
to be one that the IVR already recognizes (by the way, I was surprised to find
that Zap tones are compiled in, not in indications.conf – any thought of
changing this (with backward compatability, of course)?  I might be able
to do this myself).

 

But
if I could get the FXS to drop voltage instead of play Congestion (or a second
of Congestion in case a person is listening, and then drop voltage) that would be
even simpler.  But can I make that happen, and how?

 

thanks

  






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SV: [Asterisk-Users] How to Playback Mailbox Owners Name?

[Thorben G. Jensen]

-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Howard Lowndes
Sendt: 12. december 2004 13:07
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [Asterisk-Users] How to Playback Mailbox Owners Name?

On Sun, 2004-12-12 at 21:45, Thorben G. Jensen wrote:
> How do I Playback the Mailbox Owners Name?
> 
>  
> 
> Ex.: I want a message saying "I am sorry but" + Mailbox Owner Name +
> "has gone to lunch"

You could get them to record their temp message in the voicemail
services; option 0, IIRC.

I understand that, but they all have recorded their name and I just would like 
to use that recording.

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[Asterisk-Users] MeetMe performance

[Jason Lixfeld]

Hey folks,
Using FreeBSD 5.2.1 and I've got the current zaptel driver installed 
from ports (0.8_1) and current ports asterisk (1.0.1).  I've set 
options HZ=1000 in my kernel config, recompiled and rebooted and as far 
as I can tell, I've done everything right but when I try to use the 
conference, the audio is very delayed, choppy and segmented -- totally 
unusable.

At the suggestion of someone on #asterisk, I cvsup'd * against digium 
and used that instead of ports, but that didn't seem to help either.

FYI:  When I said above "when I try to use the conference" I meant 
using two non-voip phones, specifically a cell phone and a land line.  
I'd dial the number for my asterix box which is in itself a b channel 
on a PRI answered by a T100P on a friend's * box and sent via IAX over 
to my * box.  Not sure if that matters, but I figure I'd mention it 
anyway.

Anyone have any ideas here?
# meetme.conf
[rooms]
conf => 97531,24680
# extensions.conf
[conf]
exten => 1,1,Answer
exten => 1,2,Wait(1)
exten => 1,3,Authenticate(5447847)
exten => 1,4,MeetMe(97531,Mas,24680)
exten => 1,5,Playback(vm-goodbye)
exten => 1,6,Hangup()
exten => 2,1,MeetMe(97531,Ms,24680)
[EMAIL PROTECTED]://~ ]$ kldstat
Id Refs AddressSize Name
 15 0xc040 5e16d8   kernel
 24 0xc231e000 2f000zaptel.ko
 31 0xc234f000 6000 wcfxo.ko
 41 0xc2355000 a000 wcfxs.ko
 51 0xc235f000 2000 ztdummy.ko
[EMAIL PROTECTED]://~ ]$
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[Asterisk-Users] 3com NBX and Asterisk Integration.

[Steve Totaro]

I sent this once already but it didnt show up in my 
mailbox so I am not sure if ever made it to the list.  If it did, my 
apologies.
 


I have an * system and an nbx.  My plan is to 
use several grandstream 286s and plug the the phone cords into the NBX's 
analog FXO card.  It works fine but after I make a call, the port stays 
open on the NBX and I cannot make another call for around five minutes or I have 
to manually reset the port on the NBX.
 
I assume the Grandstream is still sending a busy 
signal to the NBX so the port wont close.  I know you can do things with 
Zap channels to monitor the line but is it possible with SIP?  I am pretty 
sure that this needs to be implemented on the 286 or the NBX itself and not 
*.  Here is what 3com says about it.  Maybe someone can help without 
having to buy the graybar device listed in the article.
 


  
  

  SOLUTION OFFERING FOR Telco LINES NOT PROVIDING DISCONNECT:- This 
  solution may not work in all implementations, and most likely does NOT 
  work for PBX Analog Extensions coming into the NBX, as they are not actual 
  phone company lines, they do not go into recordings or produce re-order 
  tones- This solution is a workaround to be attempted, with the 
  explicit understanding that 3Com Corporation does NOT endorse this product 
  in any way, but prefers to provide a solution for continued use of the 
  NBX system:The disconnect product is available through GrayBar (800 
  825 5517).Please shop around and get the unit you feel may be best for 
  your situation.Product Manufacturer: Electronic Tele-Communications, 
  Inc.Model: MAX Terminator Disconnect Unit"Scans for progress tones 
  - dial tone, reorder tone, busy signal - and generates a disconnect signal 
  within seconds of recognizing the tone. The disconnect signal is passed 
  down the line to the peripheral equipment, allowing proper operation. 
  Prevents voice mail systems from recording annoying telephone tones after 
  a caller hangs up. Also helps alleviate tied up phone lines.- Provides 
  disconnect detection for phone systems and peripherals- Helps maintain 
  accurate call count statistics- Services 12 lines; up to 120 lines 
  with rack-mount unit- Progress tone scanner- Stand alone 
  chassis- Rack-mounted cards (card file accommodates up to 10 
  cards)- Easy installation"GrayBar Catalog # and Description 
  (GrayBar 800 825 5517 for pricing and availability - ask for similar 
  products too)ETC - 8208 - SN12 Standard Chassis Unit (12 lines) 11.2 " 
  long x 8.5" wide x 1.5" deep or rack mount for more lines:ETC - 1144 - 
  CN12 - Rack Card (12 lines per card)ETC - 1145 - 
  AC - Rack Card File - 
  accommodates up to 120 lines (10 cards)
   
  this thing costs over a 
grand!
  

 
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[Asterisk-Users] RE: Voice Prompt Info

[Warren Burstein]

Ariel Batista wrote:

> Warren Burstein wrote:
>> One more thing about prompts, it's better to say "for sales press 5" 
>> than "press 5 for sales", because by the time you hear "sales" you've 
>> already forgotten what number it was.
>
> If you add the sounds all you need is For Sales recorded the new sounds
> have press # already. So you don't need to get any additional recorded
> items except the one that says "For Sales" by Allison. If you want have
> her record "Press" as an additional recorded item.

She already did a message containing just "press", sounds/vm-press.gsm in
asterisk-1.0.0 (not asterisk-sounds).

I was thinking about a project I have to bring up very soon won't be time to
wait for new recordings, so we are going to record the messages ourselves,
but yeah, if we're making messages for everyone to use, we already have
"press" and "press 1".

Anyway, the departments we need are support, sales, projects. systems, and
operator.  There already is

%for-tech-support.gsm%For technical support
%to-reach-operator.gsm%to reach an operator

so all we would like to add is sales, projects, and systems.  Maybe so all
the messages are similar, add "for an operator" as well.

Although we might decide to stick with our own recordings, in case later we
need to change a message ASAP.

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Re: [Asterisk-Users] RE: Polycom 500 - Dialtone while connected

[Julien Goodwin]

On Sun, Dec 12, 2004 at 05:50:16AM -0600, Rich Adamson arranged a set of bits 
into the following:
> > > I'm interested too. Any chance to put the archive in a ftp site?.
> > > 
> > 
> > I am also interested in getting the 1.3.4 firmware. It annoys me that I 
> > can't just get it from Polycom's website, and forces me to rethink 
> > deploying their phones for customers.
> 
> Send emails to the Polycom sales, support and other groups, and complain
> to them. Maybe if enough folks do that they will rethink their policy.
> 
> They claim to be handling it the way they do because they want to
> "maintain high quality customer support" through certified dealers.
> That might be true for their more sophisticated products, but it
> certainly does not appear to be working for their IP phones. I'd bet
Doesn't work for their better phones either...
The problem is that they assume that the reseller knows more then the
customer, something that hasn't been true for a *long* time.
Polycom are second only to Cisco in the shere stupidity of their
managemnt (yes I haven't delt with Avaya or some of the more traditional
companies who are apparently just as bad) with regards to VoIP,
believing that their customers will want to buy their softswitches, and
are not buying VoIP for say _the_flexibility_.

I've tried telling Cisco that if we could simply have a copy of the
Skinny protocol docs (which do exist and are distributed to some
companies) that they would have increased sales due to the people who
want features SIP can't provide, or the possibility of the integrated
applications. But they don't listen and don't seem to care. Fortunatly
the word I'm hearing (at least in .au large commercial) is that many
large companies that have gone cisco are getting very annoyed at them
for promising and not delivering, and if they continue their next
upgrade will be explicitly *not* cisco. [Those are direct words from a
few large scale PBX integrators I know, not myself]

If they would just realise that if they had their products actually
realise the potentional that VoIP offers they'd increase sales where it
matters, on the equipment that's VISIBLE, with THEIR branding on it, not
* or whoever makes the switch. And at least for * they dont need to do
anything, just release the docs _they_already_have_.

> a fair number of folks reselling their IP phones aren't certified 
> and they are picking up the product through (back-door) distributors.
> (That's got to be part of the reason why resellers do not include 
> copies of the required (license) software when shipping product, if
> when its stipulated on a purchase order.)
I assume you mean even when, and if a vendor did that I'd just return it
and not pay the bill. If it's not what I ordered then it doesn't meet
the contract and so I won't pay. Any vendor that still complained would
never get my business, not that of anyone I know.


pgpqdB3QeuM5p.pgp
Description: PGP signature
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[Asterisk-Users] Pattern-matching in the dial-plan

[The Traveller]

Hey all,

I'm trying to add some logic to a dial-plan to allow the caller to
terminate a number with a "#", but also accept it without this
terminator.  While trying this, I noticed that, for example,
extension "_[*0-9]XXX.#" always seems to match, whether the last digit
dialled is a "#" or not.  It's as if the parser assumes everything
after the "." will match and doesn't look any further.  Is this expected
behaviour?  If so, what would be the best solution to my problem?
I currently solved it by avoiding the "." and matching every possible
number-length seperately, both with and without the "#"-terminator.
It works, but seems like it should be doable with just 2 matches.

The box I'm trying this on is running the CVS HEAD of about a week
ago.  Thanks in advance for any suggestions.



   Grtz,

 Oliver
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[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *

[Olle E. Johansson]

Welcome to the Asterisk users community!

Asterisk.org is a fast moving project. New code is added every day.
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Our community is also growing fast and we're having a lot
of interaction, on the IRC and on the mailing lists.
It's great to have you participating in this Open Source project
- building an Open Source PBX. Here are a few things to know and
remember while working with the project.
** The mailing list is growing
The lead programmer of Asterisk, Mark Spencer at Digium, inc, half a
year ago wrote:
The Asterisk community is growing at a remarkable pace.  I know there are
thousands of you out there -- in fact there are over eight *thousand*
subscribers to asterisk-users alone, and almost one *thousand* registered
users on the bug tracker.
Today, we propably have over 10,000 readers. This means that everything
anyone write to this mailing list, is sent to thousands of mailboxes
that is already flowing over with messages.
** Think before sending a message, think twice
I would like to stress the fact that you have to think before you send a
message to such a big list. Do *not* send out personal replies on the list.
If you offer services to someone, do *not* CC: or reply to the list, it
will annoy more potential customers than get you new customers. If you
send out a message by mistake, you don't have to apologize to all of us,
we understand you're embarassed. We will get more annoyed by your apology
than over your first message.
** Try finding the answer first, then ask the list
The Asterisk Wiki at http://www.voip-info.org project is an important
knowledge base for the project.
Go there to find your answer first, then search the mailing list
archives (Google or http://search.voip-forum.com) and then
go to the IRC channel. The IRC channel is populated with Asterisk gurus
around the clock (literally) and they'll help you move forward.
* IRC info: http://www.asterisk.org/index.php?menu=support#irc
* There's many links to Asterisk web pages on the documentation
  page at http://www.asterisk.org
* The Asterisk FAQ is found on the wiki
  http://www.voip-info.org/wiki-Asterisk+FAQ
* The Asterisk documentation project (which needs your help)
  is at http://www.asteriskdocs.org
  Their handbook "The hitchhiker's guide to Asterisk" is already
  well worth reading.
Finally, if you don't find the answer elsewhere, try the list.
** Mailing lists
For developers, there is a developer's list, asterisk-dev.
For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a
list called asterisk-bsd. There is also a business list
for those that want to ask for commercial services and
inform their community about new services.
You'll find all lists on http://lists.digium.com, which is the
site where you manage your subscription to this list as well.
Please, do not crosspost the same message to multiple mailing
lists. It will not help you, it will only add to the mail flow
and get people that read both lists irritated.
** Reporting bugs
If you think you have found a bug, report it. We need bug reports.
Read this document http://www.digium.com/bugtracker.html and then
go to the bugtracker http://bugs.digium.com to file a report.
If you are unsure, find a bug marshal on the IRC channel to help
you. They're appointed to support you with how to handle bugs.
Please check the bugtracker thoroughly before posting a new bug;
often, your bug or feature already exists but is simply slowly
making it's way through the system.  Duplicate reports slow things
down for everyone, so please spend a few minutes searching first.
The bug tracker is also a place where you add your contribution
to Asterisk. If you have coded extra functionality, make sure you
give it back to the project so it can be added to the code base.
This is how Asterisk grows, free contributions and consultants
that are paid to add functionality on a case by case basis.
** Be a community member - contribute!
The Asterisk software growth is very much based on user contributions.
That's really how we all pay for the software - and get revenue back.
If you develop custom functionality, you can rest assured that there
is someone out there that wants it, needs it and will be helped by it.
Don't forget to contribute. Open Source is both giving and taking.
The financial model behind it all is really cooperative in some way.
As one member to the community said to a contractor:
  "Hey, I'm paying you to deliver code to me, then I'm giving it
   away to the community. How did this happen?"
It's the Open Source business model. And if it didn't work, we
wouldn't have a lot of the software platforms that we all use
in our business systems - Linux, Apache, MySQL, PostgreSQL and
Asterisk.
** Remember: It's Open Source, it's voluntary
Asterisk.org is a Open Source project. This means you can't request
help from people, demand new functions or s

Re: [Asterisk-Users] gap in priorities - what happens

[Brancaleoni Matteo]

Hi,

Il giorno dom, 12-12-2004 alle 14:38 +0200, Warren Burstein ha scritto:
> When I first saw the priority numbers in extensions.conf, I thought BASIC,
> if a number is missing, * will fall thru to the next number.  I learned that
> this is not so, if you have nothing between 1 and 3, you don't ever get to
> 3.
that's true.

> But I'm wondering what does happen?  Hangup and wait for next offhook?
> Undefined?
Timeout is called.
Ie if exists the exten "t", after the timeout (default 5 secs, if
I remember correctly) will be executed, otherwise hangup

Matteo.



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[Asterisk-Users] gap in priorities - what happens

[Warren Burstein]

When I first saw the priority numbers in extensions.conf, I thought BASIC,
if a number is missing, * will fall thru to the next number.  I learned that
this is not so, if you have nothing between 1 and 3, you don't ever get to
3.

But I'm wondering what does happen?  Hangup and wait for next offhook?
Undefined?


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Re: [Asterisk-Users] Re: Ethernet Channel Bank idea

[Rich Adamson]

> > > Allied Telesyn VoIP Access Device
> > > http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf
> > >  
> > > 
> > > 
> > > This is a 24-port FXS 1u device, conveniently presented as a single 
> > > RJ-21 TELCO connector.
> > 
> > yeah, but those are expensive as crap.  i was thinking about something 
> > more competetive with a channel bank
> 
> You know, if someone had some time on their hands, was good at 
> hardware/software hacking and had the will, the old Livingston/Lucent PM3 
> platform would make an awesome 48 port IAX2 <-> PRI/T1 channel bank.
> 
> Basically, the PM3 has 2 T1 ports that can be configured for ISDN PRI. The 
> core of the system runs on an AMD x86 CPU. The plug in Modem cards have 
> Lucent DSP's on them (up to 50 in a box). Flash size is 4 megs, and RAM is 
> usually around 4 megs. That is still quite a bit of horsepower, and the 
> boxes are under $400 now.
> 
> The DSP's could be used for Codec Translation, if neccessary, or for echo 
> cancellation.
> 
> And, we can get access to the original Lucent ComOS Source code.

One of Livingston's developers use to hang around this list. Haven't
seen him post for awhile so not sure if he's still hanging out or
not. Maybe he'll read this and comment.



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Re: [Asterisk-Users] How to Playback Mailbox Owners Name?

[Howard Lowndes]

On Sun, 2004-12-12 at 21:45, Thorben G. Jensen wrote:
> How do I Playback the Mailbox Owners Name?
> 
>  
> 
> Ex.: I want a message saying âI am sorry butâ + Mailbox Owner Name +
> âhas gone to lunchâ

You could get them to record their temp message in the voicemail
services; option 0, IIRC.

> 
>  
> 
> Thanks
> 
> Thorben
> 
> 
> 
> __
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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] RE: Polycom 500 - Dialtone while connected

[Rich Adamson]

> > I'm interested too. Any chance to put the archive in a ftp site?.
> > 
> 
> I am also interested in getting the 1.3.4 firmware. It annoys me that I 
> can't just get it from Polycom's website, and forces me to rethink 
> deploying their phones for customers.

Send emails to the Polycom sales, support and other groups, and complain
to them. Maybe if enough folks do that they will rethink their policy.

They claim to be handling it the way they do because they want to
"maintain high quality customer support" through certified dealers.
That might be true for their more sophisticated products, but it
certainly does not appear to be working for their IP phones. I'd bet
a fair number of folks reselling their IP phones aren't certified 
and they are picking up the product through (back-door) distributors.
(That's got to be part of the reason why resellers do not include 
copies of the required (license) software when shipping product, if
when its stipulated on a purchase order.)


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[Asterisk-Users] Problems getting Asterisk Realtime to work

[Jason Goecke]

I have installed the CVS Head as of 12/12/04, as well
as the asterisk-addons to ensure that
/usr/lib/asterisk/modules/res_config_mysql.so exists.

I have configured the following (after building a new
DB with the appropriate SQL examples, with mods to
drop the invalid keys, on the Wiki):

- /etc/asterisk/res_mysql.conf

[general]
dbhost = 127.0.0.1
dbname = my_db
dbuser = my_uname
dbpass = my_secret
dbport = 3306
dbsock = /tmp/mysql.sock

- /etc/asterisk/extconfig.conf

; Extconfig.conf for realtime configuration

voicemail => mysql,my_db,voicemail_users

(Just want to try something simple such as voicemail
for the initial testing.)

I have removed the [default] section from my
voicemail.conf. When I try to access voicemail after
restarting Asterisk, no voicemail config is found.

Anyone have any luck?
-
I notice I get this error at startup:

parse error: No category context for line 1 of
/etc/asterisk/extconfig.conf

If I change my extconfig.conf to:

; Extconfig.conf for realtime configuration

[default]
voicemail => mysql,my_db,voicemail_users

The error goes away, but the config still does not
work. Can't find anything on the new Wiki pages on the
subject though.
-
Also posted here:

http://asterisk.xvoip.com/viewtopic.php?t=764&start=0&postdays=0&postorder=asc&highlight=
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Re: [Asterisk-Users] RE: Polycom 500 - Dialtone while connected

[Greg Boehnlein]

On Thu, 9 Dec 2004, Jorge Mendoza wrote:

> Andrei,
> 
> I'm interested too. Any chance to put the archive in a ftp site?.
> 
> Jorge Mendoza

I am also interested in getting the 1.3.4 firmware. It annoys me that I 
can't just get it from Polycom's website, and forces me to rethink 
deploying their phones for customers.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] Re: Ethernet Channel Bank idea

[Greg Boehnlein]

On Fri, 10 Dec 2004, nik martin wrote:

> news.gmane.org wrote:
> > Allied Telesyn VoIP Access Device
> > http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf
> >  
> > 
> > 
> > This is a 24-port FXS 1u device, conveniently presented as a single 
> > RJ-21 TELCO connector.
> 
> yeah, but those are expensive as crap.  i was thinking about something 
> more competetive with a channel bank

You know, if someone had some time on their hands, was good at 
hardware/software hacking and had the will, the old Livingston/Lucent PM3 
platform would make an awesome 48 port IAX2 <-> PRI/T1 channel bank.

Basically, the PM3 has 2 T1 ports that can be configured for ISDN PRI. The 
core of the system runs on an AMD x86 CPU. The plug in Modem cards have 
Lucent DSP's on them (up to 50 in a box). Flash size is 4 megs, and RAM is 
usually around 4 megs. That is still quite a bit of horsepower, and the 
boxes are under $400 now.

The DSP's could be used for Codec Translation, if neccessary, or for echo 
cancellation.

And, we can get access to the original Lucent ComOS Source code.

Anyone game? :)

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] Very Cool.........Asterisk Made Wired Magazine

[Greg Boehnlein]

On Fri, 10 Dec 2004, JR Richardson wrote:

> Hi Guys,
> 
> The article "They've Got Your number" in the Dec 2004 issue of WIRED
> magazine mentions Asterisk PBX (on p.100).  The article is about phone
> phreaks hijacking cell phones with Bluetooth technology along with spoofing
> CID to pull some clandestine hacks on the PSTN.  Anyhow, Asterisk is
> mentioned as the PBX of choice for an outfit: Telephreaks.org out of Florida
> that has built their own free VoIP service.
> 
> Quotes:
> 
> "Slestak, Da Beave, and GiD are the crew behind Florida-based
> Telephreaks.org, a free VoIP service that they've built to run on a
> roll-your-own, open source private branch exchange (PBX) system called
> Asterisk"  "But with Asterisk, there's no need for the phone company to
> manage your phone lines anymore.  Your can do it yourself."
> 
> Well, it's good that Asterisk made WIRED magazine but really it should be on
> the front page with Mark's smiling mug on the cover like Linus' was on the
> Nov 2003 issue of WIRED.

Well.. it took 11 years from the first Linux release for Linus to make it 
onto the cover. Mark has a way to go yet! ;)

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] Can I re-write an incoming caller-id?

[Ronald Wiplinger]

Eric Wieling aka ManxPower wrote:
Dan Weber wrote:
check out application SetCallerID

Thanks that worked, ... for some lines only.
I have still the problem that I cannot get the caller-id from the pstn 
line. Maybe I do something wrong with the procedure.

I changed in zapata.conf to all combinations:
cidsignaling =bell, v23 and dtmf
cidstart = ring and polarity
after saving I reloaded in the running asterisk console and tried a call.
There is no caller id to see!!!
A parallel phone to the pstn line shows the caller-id though. So it must 
be just a setting!!!

Since I use "9" as the prefix to dial the pstn line I setup my 
extensions.conf

exten => s,1,NoOp(${CALLERIDNUM})
exten => s,2,Wait(1)
exten => s,3, SetCallerId(9${CALLERIDNUM})
...
and see now the "9" on my display but not the callerid of the caller.
bye
Ronald

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[Asterisk-Users] Caller ID info ZAP --> SIP??

[stuff]

Hi everyone,

I've been toying with * for quite some time now.  I've got two Cisco
7940's with the SIP firmware playing nice with *.  I can also make
outbound calls via IAXTel (toll-free calls only) and all other calls I
have routed out my X100P-clone adapter.

Here's my question...  Is there a way to capture the inbound callerid from
my phone line (coming in on the X100P) and have it appear on my SIP phones
properly?

Or maybe I'm just doing this wrong.  I want all my IP phones to ring when
someone from the outside calls my number.  When this fails... send them to
voicemail.  Right now I just dial my IP phone when the outside line rings.
 If there is a different way about doing this -- please let me know!

This is how my test config is right now:

extensions.conf
[incoming]
exten => s,1,Dial(SIP/2002,20)

The phone does ring... but for caller id info it just shows "asterisk".

Any help would be appreciated.

-= EB =-
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[Asterisk-Users] How to Playback Mailbox Owners Name?

[Thorben G. Jensen]

How do I Playback the Mailbox Owners Name?

 

Ex.: I want a message saying “I am sorry but“
+ Mailbox Owner Name + “has gone to lunch”

 

Thanks

Thorben






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Re: [Asterisk-Users] ACK from asterisk not matched to transaction by SER / LCS2005

[Olle E. Johansson]

Public Dump wrote:
For reasons unknown to me, SER and subsequently a Microsoft Live 
Communcations Server 2005 seems to have problems, matching a SIP ACK 
request from asterisk to the ongoing SIP transaction, I have attached 
the complete log, but the essential lines are:
 
That's a bug in Asterisk that is in the bug tracker and needs to be
fixed. Asterisk is sending the ACK to the AOR in the invite, not to
the contact address in the 200 OK.
Regards,
/Olle
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Re: [Asterisk-Users] Problem with TDM400P and cidstart=polarity

[Peter Svensson]

On Sat, 11 Dec 2004, Rickard Kristiansson wrote:

> I'm testing a TDM400P with FXO module to receive incoming calls from an 
> analogue line and send it to a SIP device.
> To recieve callerid, I need to use cidsignalling=dtmf and cidstart=polarity.
> The problem is that when a call is finished, the TDM400P seems to require 
> about 20 seconds to prepare for the next incoming call. If a new call comes 
> in within 20 seconds after the previous call was hungup, the TDM400P answers 
> with a modem carrier, sounding like you're calling a modem pool..!  The 
> caller hangs up and retries the call, and the next time everything is OK. 
> If the second call comes in later than 20 seconds after the previous call 
> was finished, or if I remove the cidstart=polarity (and don't get callerid) 
> everything works fine.
> I can't see any difference in the Asterisk debug logs worth mentioning... 
> Has anyone experienced anything similar..?

It may be that the remote hangup supervision for your line is signalled
via a polarity reversal as well and Asterisk misstankenly thinks that is
the start of the callerid for the next call. The 20 seconds would then be
the time it takes for Asterisk to abandon that phantom call.

A similar issue has been discussed in 
  http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002847

My comments there describe how the Swedish PSTN works. The signalling
there is built around polarity reversals. At the moment I do not think
Asterisk handles all cases. Since we are isdn based I have not looked
further into it.

Peter

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[Asterisk-Users] Problems getting Asterisk Realtime to work

[Jason Goecke]

I have installed the CVS Head as of 12/12/04, as well
as the asterisk-addons to ensure that
/usr/lib/asterisk/modules/res_config_mysql.so exists.

I have configured the following (after building a new
DB with the appropriate SQL examples, with mods to
drop the invalid keys, on the Wiki):

- /etc/asterisk/res_mysql.conf

[general]
dbhost = 127.0.0.1
dbname = my_db
dbuser = my_uname
dbpass = my_secret
dbport = 3306
dbsock = /tmp/mysql.sock

- /etc/asterisk/extconfig.conf

; Extconfig.conf for realtime configuration

voicemail => mysql,my_db,voicemail_users

(Just want to try something simple such as voicemail
for the initial testing.)

I have removed the [default] section from my
voicemail.conf. When I try to access voicemail after
restarting Asterisk, no voicemail config is found.

Anyone have any luck?
-
I notice I get this error at startup:

parse error: No category context for line 1 of
/etc/asterisk/extconfig.conf

If I change my extconfig.conf to:

; Extconfig.conf for realtime configuration

[default]
voicemail => mysql,my_db,voicemail_users

The error goes away, but the config still does not
work. Can't find anything on the new Wiki pages on the
subject though.
-
Also posted here:

http://asterisk.xvoip.com/viewtopic.php?t=764&start=0&postdays=0&postorder=asc&highlight=
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[Asterisk-Users] DIALSTATUS missing an important condition?

[chris vince]

I have recently built my first asterisk system and am very impressed with 
its capabilities.

However, I have run into one problem that hopefully someone can help me 
with.

I am trying to use the DIALSTATUS function to route incoming calls to the 
appropriate Voice Mail (busy or unavailable) or to an Unavailable Number 
recording if the number is not assigned.

However, I find that DIALSTATUS seems to generate an CHANUNAVAIL status for 
any 1 of 2 conditions:
1) the dialled user is not logged in (and hence no channel) or
2) the dialled user does not exist at all (ie the number is not assigned in 
sip.conf) (and hence no channel)

Obviously for condition 1 the call should be sent to VM unavailable, whereas 
for condition 2 I would like to send it to a "number you have dialled is not 
in service" recording - with no Voice Mail involved.

I have managed to get this scenario working but I don't think my solution is 
very elegant or even correct (although it seems to work).

Here are the relevant parts of my extensions.conf. My VM box numbers are 
exactly the same as the phone number so I only use ARG1 in the macro. (2000 
= 2000 etc.). I only have SIP phones at the moment and they are all 
allocated in the 20XX numbering range.

[altea_extensions]
;This is a "catchall" for any 4 digit number dialled starting with "20"
;Using it removes the need to provide a routing plan for each phone
exten => _20XX,1,ResponseTimeout,1	; Response Timeout for non working 
numbers
exten => _20XX,2,Macro(stdexten_sip,${EXTEN})	;send to macro for processing

;following is needed if an extension is unassigned (ie not datafilled) 
because
;DIALSTATUS cannot (?) differentiate between an unassigned # or 1 that is 
not answered or not logged in
;an unassigned (non working number) causes a timeout in the std-extn macro 
and it drops back here
;where I provide a "not in service" recording

exten => t,1,Macro(not_in_service);send to "number not in service" 
recording
exten => t,2,Hangup

[macro-stdexten_sip]
; Standard extension macro for SIP phones (modified):
;   ${ARG1} = Dialled number
;
exten => s,1,Dial(SIP/${ARG1},20,tT)		; Ring the interface, 20 secs maximum
exten => s,2,Goto(s-${DIALSTATUS},1); Jump on Status 
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1})	; no answer = unavailable
exten => s-BUSY,1,Voicemail(b${ARG1})		; busy
exten => s-CHANUNAVAIL,1,Voicemail(u${ARG1})	;no channel (not logged in) = 
unavailable
exten => s-CONGESTION,1,Macro(120_ipm)	;Don't know what this is but will 
include anyway

Am I missing something here or should there be another condition such as 
"Unassigned"? Asterisk seems to know that the number is unassigned because 
it writes a "No such host" message into the log. Is there any way of 
trapping this message in Call Processing to route this call correctly?

Or am I getting to deep here and there's a real simple way to do it that 
I've missed?

chris
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Re: [Asterisk-Users] Digium Card Error

[Brancaleoni Matteo]

Hi,

Il giorno dom, 12-12-2004 alle 00:36 -0800, Charles S. Antrim ha
scritto:
> I have success installing and compiling, but if I reboot I have to modprobe 
> again to get he 
> drivers loaded for the module I am using.  I am using rhes31 and a tdm card 
> with one fxo and 
> one fxs.

perhaps you have to build a script that loads modules on boot?
see in zaptel src dir, there's a zaptel.sysconfig & zaptel.init
demo examples. Not installed by default.

Matteo.

-- 

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[Asterisk-Users] Digium Card Error

[Charles S. Antrim]

I have success installing and compiling, but if I reboot I have to modprobe 
again to get he 
drivers loaded for the module I am using.  I am using rhes31 and a tdm card 
with one fxo and 
one fxs.

tia


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RE: [Asterisk-Users] Will Adtran TSU 600 work with *?

[Shoval Tomer]

People on the list tend to think you can’t
make many cards work on a regular desktop.

 

If you’re willing to wait a couple
of week I might have an answer for you.

 











From: Robert Augustyn
[mailto:[EMAIL PROTECTED] 
Sent: Saturday, December 11, 2004
7:13 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Will
Adtran TSU 600 work with *?



 





Hi, 

I am
looking at getting adtran tsu 600 p/n 1200.076L2 for my small office

It comes
with 6 FXS ports and I would use 2 X100Ps for FXO ports.

Would
that work ? Is there anything I would have to be aware of in such
configuration?

What
would be a better solution?

robert

 






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This message has been scanned for viruses and 
dangerous content by MailScanner, and is 
believed to be clean. 
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