[Asterisk-Users] Re: SoftFAX/spandsp
Reynaldo Simbulan
rsimbulan at hotmail.com
Thu Mar 25 20:22:22 MST 2004
Hi,
I've been testing the soft fax but I am getting this segmentation fault
whenever I receive a fax. Can somebody help me please?
I've loaded asterisk from CVS and running Redhat 9.0 and downloaded the
latest spandsp. I am sending a fax from Windows XP fax software thru a
ringmaster then to X100P.
WindowsXP --> modem ---> ringmaster (CO simulator) ---> X100P ---> *
== Spawn extension (prepaid, fax, 0) exited non-zero on 'Zap/1-1'
-- Executing RxFAX("Zap/1-1", "/root/test.tiff") in new stack
Changed from phase 0 to 1
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Start receiving document
Changed from phase 1 to 4
Sending ident
>>> CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and V.29
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85
R8x15.4lines/mm OK
Minimum scan line time for higher resolutions: T15.4 = T7.7
>>> DIS: 80 00 ce f0 80 80 01
HDLC underflow in state 9
Changed from phase 4 to 3
Slow carrier up
<<< TSI: 43 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 78 61 46
TSI without final frame tag
Remote fax gave TSI as: "Fax "
<<< DCS: 83 00 c6 70
DCS with final frame tag
In state 9
DCS:
Can receive fax
Selected data signalling rate: V.29, 9600bps
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Minimum scan line time: 0ms
Get at 9600
Changed from phase 3 to 5
Fast carrier up
Coarse carrier frequency 1683.64 (18)
Fast carrier down
Trainability test failed - longest run of zeros test was 0
>>> FTT: 44
Fast carrier up
Coarse carrier frequency 1699.71 (64)
Training error 9.552597
Training succeeded (constellation mismatch 11.776280)
Fast carrier trained
Fast carrier down
Changed from phase 5 to 4
Start rx document - compression 2
Start rx page
Segmentation fault
----- Original Message -----
From: <asterisk-users-request at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Friday, March 26, 2004 1:51 PM
Subject: Asterisk-Users digest, Vol 1 #3229 - 15 msgs
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>
> Today's Topics:
>
> 1. Re: IAX2 International Termination (Robert Sprockeels)
> 2. RE: SoftFAX/spandsp (Wade J. Weppler)
> 3. G.729 variants and Asterisk (Carlos Chavez)
> 4. Re: G.729 variants and Asterisk (Miguel Cavazos)
> 5. Re: G.729 variants and Asterisk (Adam Hart)
> 6. Re: ATA 182 and * (Leo Ann Boon)
> 7. Re: SoftFAX/spandsp (Eric Wieling)
> 8. Error on * startup (Simon Brown)
> 9. Asterisk (simprix)
> 10. Call & Drop / Call & Tranfer - tranfering a call to a different
number. (johnc at cleburne.com)
> 11. New minor release of Firefly (now with Speex) (Adam Hart)
> 12. Codec Voodoo (Hadar Pedhazur)
> 13. RE: New minor release of Firefly (now with Speex) (Simon Brown)
> 14. Re: SoftFAX/spandsp (Nicolas Gudino)
> 15. IAX drops calls exactly 5 secs into the call ( John Brown (CV))
>
> --__--__--
>
> Message: 1
> Subject: Re: [Asterisk-Users] IAX2 International Termination
> From: Robert Sprockeels <rsp+asterisk at boat.be>
> To: asterisk-users at lists.digium.com
> Date: Thu, 25 Mar 2004 23:42:20 +0100
> Reply-To: asterisk-users at lists.digium.com
>
> Tested from Belgium
>
> Very good quality, sometimes breaking up a little.
>
> The phone I used is a Snom200 behind *, gsm codec.
> Ping times are 110 - 115 ms.
> Did not try dtmf sending.
>
> Robert Sprockeels
>
>
> --__--__--
>
> Message: 2
> Subject: RE: [Asterisk-Users] SoftFAX/spandsp
> Date: Thu, 25 Mar 2004 17:51:31 -0500
> From: "Wade J. Weppler" <weppler at wwworks-inc.com>
> To: <asterisk-users at lists.digium.com>
> Reply-To: asterisk-users at lists.digium.com
>
> Excellent work Steve.
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Steve
> Underwood
> Sent: Thursday, March 25, 2004 10:34 AM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] SoftFAX/spandsp
>
> Hi all,
>
> My SoftFAX is now up to spandsp-0.0.1h.tar.gz. This version has a number
>
> of changes in the way the V.29 modem works. It also has some missing=20
> functionality in the T.30 implementation filled in - it was not handling
>
> EOM messages.
>
> The previous version failed for several reasons with a Dialogic=20
> VFX/40ESC. This version succeeds, although it still seems to get a few=20
> bit errors, giving some flaws on the received image. I do not see these=20
> errors with the other FAX machines I have tried. It seems like a fairly=20
> big improvement though, and work will continue to make it better.
>
> app_rxfax.c and app_txfax.c have gained a new feature. Previously they=20
> always started in answering party mode. Now this is the default=20
> behaviour, but something like:
>
> exten =3D> 5678,1,txfax(/tmp/testfax.tif|caller)
>
> will make them start in calling party mode. So far, these two apps have=20
> been little more that testbeds for spandsp. It seems some people are=20
> trying to use them for real work, so it seems like they should be=20
> gaining more features. The caller mode option was asked for.
>
> Regards,
> Steve
>
>
>
> _______________________________________________
> 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
>
>
>
> --__--__--
>
> Message: 3
> From: "Carlos Chavez" <cursor at corplacer.com>
> To: "Asterisk" <asterisk-users at lists.digium.com>
> Date: Thu, 25 Mar 2004 16:47:55 -0600
> Subject: [Asterisk-Users] G.729 variants and Asterisk
> Reply-To: asterisk-users at lists.digium.com
>
> I see that I can purchase G.729 licenses for my Asterisk server, but
I
> have seen that many phones support a G.729 variant like A or B. Are these
> suppoted by the same G.729 codec in Asterisk?
>
> --
> Carlos Chavez
> Computer Engineer, CCNA
> Corporativo Lacer S.A. de C.V.
>
>
> --__--__--
>
> Message: 4
> Subject: Re: [Asterisk-Users] G.729 variants and Asterisk
> From: Miguel Cavazos <miguel at cavazos.com.mx>
> To: Asterisk <asterisk-users at lists.digium.com>
> Date: Thu, 25 Mar 2004 17:00:00 +0000
> Reply-To: asterisk-users at lists.digium.com
>
> si funciona con el A y B
>
> Miguel Cavazos
> On Thu, 2004-03-25 at 22:47, Carlos Chavez wrote:
> > I see that I can purchase G.729 licenses for my Asterisk server,
but I
> > have seen that many phones support a G.729 variant like A or B. Are
these
> > suppoted by the same G.729 codec in Asterisk?
> >
> > --
> > Carlos Chavez
> > Computer Engineer, CCNA
> > Corporativo Lacer S.A. de C.V.
> >
> > _______________________________________________
> > 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
>
> --__--__--
>
> Message: 5
> Date: Fri, 26 Mar 2004 10:05:29 +1100
> From: Adam Hart <adam at teragen.com.au>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] G.729 variants and Asterisk
> Reply-To: asterisk-users at lists.digium.com
>
> Carlos Chavez wrote:
>
> > I see that I can purchase G.729 licenses for my Asterisk server, but
I
> >have seen that many phones support a G.729 variant like A or B. Are
these
> >suppoted by the same G.729 codec in Asterisk?
> >
> >
> B is just the fixed point version of A (from memory) - so it works the
> same as A.
>
> A is a reduced complexity version of G.729 - although they both work
> with each other. A is just slack when looking for the best
> representation of your voice.
>
> FYI, Digium's codec is G.729A, although it makes little difference
>
> --__--__--
>
> Message: 6
> Date: Fri, 26 Mar 2004 07:15:08 +0800
> From: Leo Ann Boon <leo at innovax.com.sg>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] ATA 182 and *
> Reply-To: asterisk-users at lists.digium.com
>
> No. The FXO on the 182 is only usable from the box itself. It's for
> calling local numbers.
>
> Erick Weber V. wrote:
>
> >Hi to everyone:
> >
> >Does someone know if the ATA 182 works OK with asterisk or should I get a
> >HandyTone 486 instade or an ATA 186 and a FXS to FXO converter
> >
> >Thanks
> >
> >Erick
> >
> >
> >_______________________________________________
> >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
> >
> >
> >
>
> --__--__--
>
> Message: 7
> Subject: Re: [Asterisk-Users] SoftFAX/spandsp
> From: Eric Wieling <eric at fnords.org>
> To: asterisk-users at lists.digium.com
> Organization: BTEL Consulting
> Date: Thu, 25 Mar 2004 18:19:10 -0600
> Reply-To: asterisk-users at lists.digium.com
>
> On Thu, 2004-03-25 at 09:33, Steve Underwood wrote:
> > exten => 5678,1,txfax(/tmp/testfax.tif|caller)
>
> There are a zillion fax and tiff formats. I'm trying to figure out what
> output format I should tell GhostScript to use. Any suggestions on
> which format to try?
>
> These are the formats GhostScript can output:
>
> faxg3 faxg32d faxg4 tiff12nc tiff24nc tiffcrle tiffg3 tiffg32d tiffg4
> tifflzw tiffpack
>
> --
> Eric Wieling <eric at fnords.org>
> BTEL Consulting
>
>
> --__--__--
>
> Message: 8
> Date: Fri, 26 Mar 2004 11:19:25 +1100
> From: "Simon Brown" <Simon.Brown at otterson.com.au>
> To: <asterisk-users at lists.digium.com>
> Subject: [Asterisk-Users] Error on * startup
> Reply-To: asterisk-users at lists.digium.com
>
> When I start or reload * I always get this error (once).
> Can someone point me in the right direction to fix this.
>
> WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries =
> exceeded on
> call 3c52718c594f43736b88cb4b5ec8af65 at 10.2.2.20 for seqno 102 (request)
>
> Simon
>
> -----
> This mail was content checked for malicious code and viruses
> by GFI MailSecurity.
>
>
> --__--__--
>
> Message: 9
> Date: Thu, 25 Mar 2004 19:29:12 -0500
> From: simprix <simprix at simprix.net>
> To: <asterisk-users at lists.digium.com>
> Subject: [Asterisk-Users] Asterisk
> Reply-To: asterisk-users at lists.digium.com
>
> What kind of specs do I need for a asterisk box that will have a pri for
> pstn and about 65 sip phones
>
> I was thinking a Xeon 3.05
>
>
>
> --__--__--
>
> Message: 10
> From: johnc at cleburne.com
> To: asterisk-users at lists.digium.com
> Date: Thu, 25 Mar 2004 18:38:47 -0600
> Subject: [Asterisk-Users] Call & Drop / Call & Tranfer - tranfering a call
to a different number.
> Reply-To: asterisk-users at lists.digium.com
>
> First, I am very new to this software. If I missed a searchable archive,
please point
> me in the right direction.
>
> I am wishing to know if Asterisk can be used to do a Call & Drop scenario.
>
> This is where someone calls, Asterisk answers, ask for the number that the
person
> wishes to dial, gets the PIN, and then completes the call to the number
they desired.
> Once the connection is completed, this software/service is no longer in
the call loop.
>
> Typically this scenario is used to offer a wider calling area. Called
Metro or Extended
> Metro in our area. There are many people in area that this feature is not
available
> from their phone company, or they don't want to pay much for it, as they
don't make
> many calls.
>
> I am certainly willing to provide more information, but I wanted to find
out if Asterisk
> was even something that could do it- or be modified to do so.
>
> Thanks,
>
> John Chapman
>
> --__--__--
>
> Message: 11
> Date: Fri, 26 Mar 2004 11:47:40 +1100
> From: Adam Hart <adam at teragen.com.au>
> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com>
> Subject: [Asterisk-Users] New minor release of Firefly (now with Speex)
> Reply-To: asterisk-users at lists.digium.com
>
> I've put up a new dev version of Firefly
> (http://www.virbiage.com/firefly/download/firefly-dev.exe)
>
> Notable Changes:
> DTMF now works with SIP
> Speex codec has been added
> 1 crash bug fixed - 2 more to go (if you can crash Firefly, send me the
> Hex address - probably stored in event viewer under control panel)
>
> Sorry for the delay but I've completely rewritten how contacts work
> internally (although it looks exactly the same as it did before). This
> now allows me to do some sexy things with contacts. Stay tuned
>
> I'm aiming for a stable release in two weeks so help me find the bugs.
> Many thanks to thoses who have
>
> -Adam
>
> --__--__--
>
> Message: 12
> Date: Thu, 25 Mar 2004 20:08:38 -0500
> From: Hadar Pedhazur <hadar at unorthodox.com>
> To: Asterisk-Users at lists.digium.com
> Subject: [Asterisk-Users] Codec Voodoo
> Reply-To: asterisk-users at lists.digium.com
>
> I have three * servers that all talk to each just fine, and
> all talk to other * servers (like NuFone, VoicePulse, etc.).
> I have hard-phones connected to Sipura SPA-2000s on two of
> the * servers via a local network connection. The third *
> server only gets connected to remotely, both with IAX and
> SIP softphones, and with a "roaming" Sipura with
> hard-phones.
>
> The setup works well. All of the * servers communicate
> exclusively with GSM between themselves (and also to NuFone
> and VoicePulse). The quality is pretty good. The "local"
> hard phones are using g711 uLaw (since I think that the X100P
> cards I believe use uLaw by default as well, but I could be
> way off on that assumption). Codec transcoding from uLaw to
> GSM seems to work just fine.
>
> From a couple of people who post regularly on this list, I
> have heard that they have great success with iLBC (and some
> with Speex as well). I think that NuFone prefers iLBC as
> well, though it works remarkably well for me with GSM.
>
> I did some experiments in forcing my * servers to
> communicate with each other only with iLBC. When I do that,
> and can see that they are indeed using iLBC, the quality is
> horrible. There is long stutter, like every sound is being
> "stretched" out.
>
> I purchased g729 licenses from Digium for all three servers
> as well. Using g729 on the Sipura devices yielded no better
> quality than the built-in g726. However, when I made two *
> servers communicate only with g729, the quality was
> marginally better than iLBC, and ridiculously worse than
> GSM. This was surprising to me.
>
> All of this is with a very recent cvs checkout of *, done
> this past Monday the 22nd I believe.
>
> Last point is that if I turn jitterbuffer on (with =yes),
> then I never hear _any sound_ whatsoever, but there are _no
> errors_ on either side of the channel. I can see on the CLI
> that voicemail prompts are being played (for example), but I
> can't hear anything on either side. Turning jitterbuffer=no
> immediately restores sound, but the quality only sounds good
> with GSM.
>
> What I don't understand is how some people have success with
> iLBC, and I don't. I also noticed one or two posts from
> people that claim that GSM isn't working for them, yet it
> works really well for me. Are there any settings that I am
> unaware of (other than the standard "allow/disallow"
> directives) that I should be tweaking to make these other
> codecs work as I understand they should?
>
> P.S. One last piece of voodoo, just if anyone knows the
> answer to this. On occasion, I use DIAX to connect to the
> remote * server. It works very well, and is the best of the
> IAX softphones (IMHO). Yesterday, it was working just fine.
> Today, from a different location (both yesterday and today
> behind NAT, just from different networks), it connects fine,
> but I have zero sounds and zero errors. There were _no_
> changes to the server or the software setup in between.
>
> In the past, I have had trouble using X-Lite to this
> particular * server. Today, when DIAX wasn't working
> (neither was iaxcomm, it's not a specific DIAX problem), I
> tried X-Lite again, and it worked flawlessly...
>
> The last bit of info on this is that one of the other *
> servers is on the same lan as the DIAX client, but on
> different machines. Both are coming from the same NAT
> router though. The * machine is in the DMZ, so all packets
> that are sent to the public side are routed directly to *,
> and that part works perfectly. I don't know if DIAX is
> clashing with * packets, but I know this has worked in the
> past (though it's been 2 weeks since I've tried, and I did
> cvs up the * server since it last worked...).
>
> Thanks in advance to any brave soul who tackles some or all
> of these questions/issues! :-)
>
>
>
> --__--__--
>
> Message: 13
> Subject: RE: [Asterisk-Users] New minor release of Firefly (now with
Speex)
> Date: Fri, 26 Mar 2004 12:14:07 +1100
> From: "Simon Brown" <Simon.Brown at otterson.com.au>
> To: <asterisk-users at lists.digium.com>
> Reply-To: asterisk-users at lists.digium.com
>
> When you use firefly in SIP mode it does not un-register with * on =
> exiting
> the software
>
> Simon
> =20
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Adam Hart
> Sent: Friday, 26 March 2004 11:48
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] New minor release of Firefly (now with Speex)
>
> I've put up a new dev version of Firefly
> (http://www.virbiage.com/firefly/download/firefly-dev.exe)
>
> Notable Changes:
> DTMF now works with SIP
> Speex codec has been added
> 1 crash bug fixed - 2 more to go (if you can crash Firefly, send me the =
> Hex
> address - probably stored in event viewer under control panel)
>
> Sorry for the delay but I've completely rewritten how contacts work
> internally (although it looks exactly the same as it did before). This =
> now
> allows me to do some sexy things with contacts. Stay tuned
>
> I'm aiming for a stable release in two weeks so help me find the bugs.=20
> Many thanks to thoses who have
>
> -Adam
> _______________________________________________
> 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
>
>
>
> -----
> This mail was content checked for malicious code and viruses
> by GFI MailSecurity.
>
>
> --__--__--
>
> Message: 14
> Subject: Re: [Asterisk-Users] SoftFAX/spandsp
> From: Nicolas Gudino <nicolas at house.com.ar>
> To: asterisk-users at lists.digium.com
> Organization: House Internet S.R.L.
> Date: Thu, 25 Mar 2004 22:14:39 -0300
> Reply-To: asterisk-users at lists.digium.com
>
> Hi Eric,
>
> I was all day trying and came up with this:
>
> gs -q -sDEVICE=tiffg3 -sPAPERSIZE=a4 -r204x196 \
> -dNOPAUSE -sOutputFile=$TIFFILE -- $PSFILE
>
> I'm using a modified version of "salsafax/sambafax" to enable a
> print2fax option for windows/linux clients.
>
> You add a printer to cups and share it via Samba. Then, you append a
> line with the fax number in the file you want to be faxed "Fax-Nr
> 34333333" and print it to the network printer from any application.
>
> The scripts extracts the number and then generates a call file for
> asterisk.
>
> Some ps files cannot be extracted, so I used an OCR application (gocr)
> to extract the text, maybe its overkill, but it works most of the time
> (here we send less than ten faxes a day, so its no problem for us). I
> will clean up the scripts and post them for others to use.
>
> Good luck,
>
>
> On Thu, 2004-03-25 at 21:19, Eric Wieling wrote:
> > On Thu, 2004-03-25 at 09:33, Steve Underwood wrote:
> > > exten => 5678,1,txfax(/tmp/testfax.tif|caller)
> >
> > There are a zillion fax and tiff formats. I'm trying to figure out what
> > output format I should tell GhostScript to use. Any suggestions on
> > which format to try?
> >
> > These are the formats GhostScript can output:
> >
> > faxg3 faxg32d faxg4 tiff12nc tiff24nc tiffcrle tiffg3 tiffg32d tiffg4
> > tifflzw tiffpack
> --
> Nicolas Gudino <nicolas at house.com.ar>
> House Internet S.R.L.
>
>
> --__--__--
>
> Message: 15
> Date: Thu, 25 Mar 2004 18:40:41 -0700
> From: " John Brown (CV)" <jmbrown at chagresventures.com>
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] IAX drops calls exactly 5 secs into the call
> Reply-To: asterisk-users at lists.digium.com
>
> Hi List,
>
> Two boxes
>
> A has a PRI
>
> B terminates SIP devices
>
>
> A <--IAX--> B
>
> Both on the same switch, same IP network.
>
> Call from PSTN to A gets pushed via IAX to B -> Sip device
> with no problems.
>
> Call from Sip device -> B via IAX -> A -> PSTN
> will drop exactly 5 seconds after the call is answered.
>
> I've built with 0.7.2, 1_0_Stable devel etc
>
>
> Any clue / hints ??
>
> thanks
>
>
>
>
> --__--__--
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> End of Asterisk-Users Digest
>
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