[Asterisk-Users] Spandsp....And garble incoming fax

Fri Jan 14 08:14:04 MST 2005

Hello:

   I have successfully install spandsp and patch asterisk with it. But when
I received a Fax is garble or shrink. Does any one know why???... Am using a
PRI T100P card to receive the fax and save it to a tiff file... Any help
will be greatly appreciated. Here are the versions.

Latest csv from asterisk,
spandsp-0.0.1k.tar.gz
redhat 7.3
T100P has its own IRQ.

Any help will be greatly appreciated... 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
asterisk-users-request at lists.digium.com
Sent: Friday, January 14, 2005 2:28 AM
To: asterisk-users at lists.digium.com
Subject: Asterisk-Users Digest, Vol 6, Issue 199

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Today's Topics:

   1. Re: Problem patching asterisk CVS with SpanDSP (Matt Riddell)
   2. DIAX 0.9.9g more features and higher stability (Dan)
   3. R2/MFC Mexico FREE calls to test chan_unicall (Gonzalo Gasca Meza)
   4. Re: Updated kphone 4.0.5, asterisk v1.0.3 (Howard Lowndes)
   5. RE: [Asterisk-biz] SS7 and Asterisk solution (Rob Lith)
   6. RE: TE410P card in an HP-Compaq DL380 G4 server (Joshua McAdam)
   7. Polycom Shared Call Appearance (John Bittner)
   8. Re: SER vs Asterisk for SIP (Julio Tejera)
   9. Re: How to set asterisk NOT to answer incoming lines?
      (Steven Critchfield)
  10. Limit outgoing trunk calls (Mike Sander)
  11. RE: Agentcallbackogin	withoutanyuserinputafter	extension is
      dialed. (Florian Overkamp)

----------------------------------------------------------------------

Message: 1
Date: Fri, 14 Jan 2005 19:00:11 +1300
From: Matt Riddell <matt.riddell at sineapps.com>
Subject: Re: [Asterisk-Users] Problem patching asterisk CVS with
	SpanDSP
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <41E75FEB.3040305 at sineapps.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Keith LeClaire Jr wrote:
> I'm trying to patch the current asterisk CVS with spandsp-0.0.1k.tar.gz.
> Everything compiles fine but when I go to patch the asterisk/apps/Makefile
> it fails:

:-))))))))))))))))))))))

Sorry, that's my excuse for the biggest smile ever.

I just posted the solution yesterday/day before for this exact thing.

Have you just subscribed or were you here yesterday too?

:-)

Drop me a line off list if you would like me to talk you though this 
(free of course).  The reason I say off-list is because the solution 
will already end up in the mailing list...

This is one of the simplest patches in the world to apply.  I can talk 
you through it, or you could have a look (hint +xxx means add xxx, don't 
forget that the spaces are actually tabs in the Makefile).

-- 
Cheers,

Matt Riddell
_______________________________________________

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

------------------------------

Message: 2
Date: Fri, 14 Jan 2005 08:05:36 +0200
From: "Dan" <danto at rdslink.ro>
Subject: [Asterisk-Users] DIAX 0.9.9g more features and higher
	stability
To: <asterisk-users at lists.digium.com>
Message-ID: <003501c4f9ff$27c1de50$0121a8c0 at dantenc4010>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
	reply-type=original

Hi all,

DIAX 0.9.9g is available for download (including the updated help file and
web page) from the following locations:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro

What's new in 0.9.9g (from 0.9.9f):

- during a call, accept DTMF tones as monitored events to trigger output
commands
- call timer on the phone display
- Swedish language added
- can run a command from the monitoring definition form, to test it
- ENTER key validate all fields in the Registration form
- you can select both preffered and accepted codecs
- do not autoresize main form when receiving a call and monitoring activated
- use /m switch to start DIAX minimized
- saving only main form position, all others auto positioning relative to
the main form

solved bugs:
- crash when trying to dial without registration server defined
- Config Audio form positioning issue
- not saving the main form when closing the app from the systray
- X10 send error if CM11/12 interface has some commands in the receiver
buffer
- error if trying to delete for the second time the log file
- unexpected crashes when registered with IAXTEL and/or other remote servers

As usual, please send me your feedback.

Best regards,
Dan

------------------------------

Message: 3
Date: Thu, 13 Jan 2005 22:07:46 -0800 (PST)
From: Gonzalo Gasca Meza <xomeboy at yahoo.com>
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test
	chan_unicall
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <20050114060746.34380.qmail at web60707.mail.yahoo.com>
Content-Type: text/plain; charset="us-ascii"

Miguel,

Congrats, i was testing your R2/MFC link, and I was able to made lots of
calls, all of them worked fine.Thanks for setting up this link.

When i hang up, there were no dead air, music on hold worked fine, when I
called to a conference worked fine also, busy line Telmex recording worked
also fine. Please let me know if there is anything I can help you with or if
you want to test something.

Thanks again!

---------------------------------
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Message: 4
Date: Fri, 14 Jan 2005 17:14:26 +1100
From: Howard Lowndes <lannet at lannet.com.au>
Subject: Re: [Asterisk-Users] Updated kphone 4.0.5, asterisk v1.0.3
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <1105683264.4323.11.camel at lan-255-17.lan.lannet.com.au>
Content-Type: text/plain

On Fri, 2005-01-14 at 15:09, Andrew McRory wrote:
> I have uploaded kphone and asterisk CVS stable. These packages are built
> for Fedora Core 1 and this asterisk release should fix the non-root
> permissions problem I worte about...
> 
> 	ftp://ftp.linuxsys.com/pub/releases/FC1/

OK, there are a number of issues I have detected.

The error message about closing other applications using the sound card
is definitly repated to the SIP SUBSCRIBE packets.

When I run it from an xterm, on hangup it seg faults.  This does not
happen when I run it from a KDE panel button.

The DTMF tones generated from the on-screen keypad appear not to be
recognised by *.
-- 
Howard.
LANNet Computing Associates;
Your Linux people <http://www.lannetlinux.com>
------------------------------------------
"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."

------------------------------

Message: 5
Date: Fri, 14 Jan 2005 08:16:22 +0200
From: "Rob Lith" <rob at connection-telecom.com>
Subject: [Asterisk-Users] RE: [Asterisk-biz] SS7 and Asterisk solution
To: "'Commercial and Business-Oriented Asterisk Discussion'"
	<asterisk-biz at lists.digium.com>, <rehan1 at rehan.com>
Cc: asterisk-users at lists.digium.com
Message-ID: <200501140616.j0E6GPag004475 at aphrodite.dbuzz.net>
Content-Type: text/plain;	charset="us-ascii"

Tracy, one example I can think of is here in South Africa, when VoIP is
deregulated on the 1st February the very first trick the incumbent monopoly
is going to pull out of its hat it saying that to interconnect with them
you're going to need SS7 - if there is a 'soft' way of doing this in * then
they'll come up with some excuse that its not approved by the regulator/it
not carrier grade....

Regards
Rob

> -----Original Message-----
> From: asterisk-biz-bounces at lists.digium.com 
> [mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of 
> Tracy R Reed
> Sent: 13 January 2005 23:23
> To: rehan1 at rehan.com; Commercial and Business-Oriented 
> Asterisk Discussion
> Cc: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-biz] SS7 and Asterisk solution
> 
> On Thu, Jan 13, 2005 at 01:44:16PM -0600, Rehan Ahmed spake thusly:
> > can u point us to where we can buy cheap ss7 solution
> 
> Can you tell me why you think you need one?
> 
> -- 
> Tracy Reed    http://copilotcom.com 
> This message is cryptographically signed for your protection.
> Info: http://copilotconsulting.com/sig
> 

------------------------------

Message: 6
Date: Fri, 14 Jan 2005 16:30:25 +1000
From: "Joshua McAdam" <josh at tlmtech.com>
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4
	server
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
	<asterisk-users at lists.digium.com>
Message-ID:
	<20050114063022.1077221B83 at mailsrv01-syd.hosting.mipt.com.au>
Content-Type: text/plain;	charset="us-ascii"

Has anyone logged a support issue with HP on this one?

I still haven't been able to get it working so far,
So I'm going to log a support issue here in australia to see what HP can do
about this and was wondering if anyone else has.

Josh

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexander
Lopez
Sent: Monday, 10 January 2005 4:22 PM
To: karlp at fortephones.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

Make sure you has a span defined for each port on the TE410P. With out
signaling it would not take interrupts.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Karl H.
Putz
Sent: Monday, January 10, 2005 12:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4
server

I have been having this exact problem with a Tatung dual EMT-64 server
as
well.

I have been trying to get a TE410P running and all looks great, driver
loads, runs ztcfg OK, etc. but no interrupts are ever processed.

One additional piece of info that I have not seen in this thread is that
I
am able to successfully start and run a T100P card in this system.  In
the
same PCI slot, wct1xxp driver built from the same CVS HEAD version as
the
wct4xxp.

Just hoping this might shed some light on the problem for any Digium
folks
monitoring the forum.

Karl Putz

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------------------------------

Message: 7
Date: Fri, 14 Jan 2005 01:37:14 -0500
From: "John Bittner" <john at simlab.net>
Subject: [Asterisk-Users] Polycom Shared Call Appearance
To: <asterisk-users at lists.digium.com>
Message-ID: <200501140137156.SM00436 at johnb2>
Content-Type: text/plain;	charset="us-ascii"

Has anyone got Polycom Shared Call Appearance working with
Asterisk ?

If Asterisk doesn't support this, I am willing to put up a
bounty of 1000 to get it to work. 

John Bittner
Simlab.net

Shared Call Appearance Signaling
A shared line is an address of record managed by a server.
The server allows multiple
endpoints to register locations against the address of
record.
SoundPointR IP supports shared call appearances (SCA) using
the SUBSCRIBENOTIFY
method in the "SIP Specific Event Notification" framework
(RFC 3265).
The events used are:
. "call-info" for call appearance state notification.
"line-seize for the phone to ask to seize the line

------------------------------

Message: 8
Date: Fri, 14 Jan 2005 00:43:05 -0600
From: "Julio Tejera" <jat at realityfirewall.net>
Subject: Re: [Asterisk-Users] SER vs Asterisk for SIP
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID: <015d01c4fa04$4d96d400$e101a8c0 at Aceituno>
Content-Type: text/plain;	charset="iso-8859-1"

* is a "middleware"

HTH

-------
Ing. Julio Alvarez Tejera
Unix Trends
*BSD, Solaris & Linux
VoIP & CT Solutions Finder
Asterisk PBX Consultant
Costa Rica Land +506-359-9753
USA Toll Free     +1-888-899-6269
---------------
"extremely stable systems"

----- Original Message -----
From: "Ashling O'Driscoll" <ashling.odriscoll at cit.ie>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, January 13, 2005 10:57 AM
Subject: RE: [Asterisk-Users] SER vs Asterisk for SIP

>From my (fairly limited) understanding, I think the fundamental
difference is that Asterisk is a pbx (offering all the features
associated with a pbx, voicemail, call transfer, call detail
recording etc) whereas SER is just a sip proxy (albeit a good one).

Therefore Asterisk deals in terms of phones extensions whereas if you
want a system that can contact clients with sip urls, ser will have
to be set up. Also the audio i.e. rtp stream, traverses asterisk i.e.
it acts as a middle man holding onto the call, and if you want the
audio to go peer to peer (which it ideally should with sip), ser is
also needed.

Aisling.
---- Original Message ----
From: vicky at freebsdcluster.net
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] SER vs Asterisk for SIP
Date: Thu, 13 Jan 2005 17:50:39 +0100

>Why is SER considered a better SIPserver than asterisk , why is it
>that SER
>can handle more clients than asterisk can. And if this is just cause
>of say
>poor SIP handling code in asterisk then is there anything being done
>to fix
>it. Just wanted to know why SER claims to be better than asterisk as
>a SIP
>server. ?
>
>--
>regards
>Vikram (http://www.vicramresearch.com)
>_______________________________________________
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------------------------------

Message: 9
Date: Fri, 14 Jan 2005 00:51:50 -0600
From: Steven Critchfield <critch at basesys.com>
Subject: Re: [Asterisk-Users] How to set asterisk NOT to answer
	incoming lines?
To: C F <shmaltz at gmail.com>,	Asterisk Users Mailing List -
	Non-Commercial Discussion	<asterisk-users at lists.digium.com>
Message-ID: <1105685510.13831.154.camel at critch>
Content-Type: text/plain

On Thu, 2005-01-13 at 21:09 -0500, C F wrote:
> The definition of normal in the case of PBX implementations is up to
> the customer.

You sure are acting like a 'tard lately. 

No a customer does not define normal, the market defines normal. A
customer defines an implementation. That implementation is either normal
or an exception/deviation of normal.  

> On Thu, 13 Jan 2005 10:44:51 -0600, Steven Critchfield
> <critch at basesys.com> wrote:
> > On Thu, 2005-01-13 at 16:08 +0000, Patrick Lidstone (Personal e-mail)
> > wrote:
> > 
> > > I don't think Kelly's response is correct, at least for TDM FXO
boards.
> > > I could not find a way of preventing the FXO board grabbing the line
> > > when it rang, and subsequent enquiries on this list at the time
> > > suggested that it wasn't actually possible - which is a pity, as it
> > > means it is impossible to piggy back Asterisk on a POTS line with
other
> > > auto-answering equipment (e.g. data collection terminals).
> > 
> > It isn't normal to put a PBX on a line shared with other equipment. It
> > is normal to route the other equipment through the PBX.
> > 
> > --
> > Steven Critchfield <critch at basesys.com>
> > 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Steven Critchfield <critch at basesys.com>

------------------------------

Message: 10
Date: Fri, 14 Jan 2005 18:00:26 +1100
From: "Mike Sander" <mike at corporatebankinginternational.com>
Subject: [Asterisk-Users] Limit outgoing trunk calls
To: <asterisk-users at lists.digium.com>
Message-ID: <20050114070028.9F8D9EE9AB at mail.tyneinternational.com>
Content-Type: text/plain; charset="windows-1250"

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Message: 11
Date: Fri, 14 Jan 2005 08:26:10 +0100
From: "Florian Overkamp" <florian at obsimref.com>
Subject: RE: [Asterisk-Users] Agentcallbackogin
	withoutanyuserinputafter	extension is dialed.
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
	<asterisk-users at lists.digium.com>
Message-ID: <E1CpLr1-0004Ao-00 at clio>
Content-Type: text/plain;	charset="us-ascii"

Hi, 

> -----Original Message-----
> Ok, maybe this is an ignorant question but......  where in memory does
> asterisk store the information and how do I access it?

It's not an ignorant question, but it is like I've stated a few times now:
The agent information asterisk has is in its own memory and cannot be
accessed easily (you could probably write an AGI script that executes 'show
agents' and parses the output though). That is exactly why you make your
dialplan so every time an agent logs on or off you store your own copy if
the info in the asterisk database where it is available to you for future
reference.

BTW, I know agent technology is a bit better in CVS-HEAD but for my
customers sake (where I have to run a stable branch) I kicked out usage of
agents and now emulate it all with a few AGI scripts.

Florian

------------------------------

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End of Asterisk-Users Digest, Vol 6, Issue 199
**********************************************