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

Fri Jan 14 09:09:37 MST 2005

check out my bug post, I have yet to recieve a successful fax using rxfax.

http://www.opencall.org/mantis/bug_view_page.php?bug_id=0000019

and I'm using newest versions of everything.

-Matthew

----- Original Message ----- 
From: "Luis Mata" <mataluis at xtremenetworks.biz>
To: <asterisk-users at lists.digium.com>
Sent: Friday, January 14, 2005 9:14 AM
Subject: [Asterisk-Users] Spandsp....And garble incoming fax

> 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
>
> Send Asterisk-Users mailing list submissions to
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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!
>
>
>
>
>
>
>
>
>
> ---------------------------------
> Do you Yahoo!?
>  Yahoo! Mail - Easier than ever with enhanced search. Learn more.
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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)
> >_______________________________________________
> >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
> >
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> _______________________________________________
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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
>
>
>
>
> ------------------------------
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> End of Asterisk-Users Digest, Vol 6, Issue 199
> **********************************************
>
>
> _______________________________________________
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