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" <[EMAIL PROTECTED]> To: <[email protected]> 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: Friday, January 14, 2005 2:28 AM > To: [email protected] > Subject: Asterisk-Users Digest, Vol 6, Issue 199 > > Send Asterisk-Users mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > 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 <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Problem patching asterisk CVS with > SpanDSP > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > 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" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] DIAX 0.9.9g more features and higher > stability > To: <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > 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 <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test > chan_unicall > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > 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. > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20050113/87c34f > 80/attachment-0001.htm > > ------------------------------ > > Message: 4 > Date: Fri, 14 Jan 2005 17:14:26 +1100 > From: Howard Lowndes <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Updated kphone 4.0.5, asterisk v1.0.3 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > 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" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] RE: [Asterisk-biz] SS7 and Asterisk solution > To: "'Commercial and Business-Oriented Asterisk Discussion'" > <[EMAIL PROTECTED]>, <[EMAIL PROTECTED]> > Cc: [email protected] > Message-ID: <[EMAIL PROTECTED]> > 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: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Tracy R Reed > > Sent: 13 January 2005 23:23 > > To: [EMAIL PROTECTED]; Commercial and Business-Oriented > > Asterisk Discussion > > Cc: [email protected] > > 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" <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 > server > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <[email protected]> > Message-ID: > <[EMAIL PROTECTED]> > 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Alexander > Lopez > Sent: Monday, 10 January 2005 4:22 PM > To: [EMAIL PROTECTED]; 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 > > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > 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 > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 7 > Date: Fri, 14 Jan 2005 01:37:14 -0500 > From: "John Bittner" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Polycom Shared Call Appearance > To: <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > 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" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] SER vs Asterisk for SIP > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > 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" <[EMAIL PROTECTED]> > To: <[email protected]> > 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: [EMAIL PROTECTED] > To: [email protected] > 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 > >[email protected] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >-------------------Legal > >Disclaimer--------------------------------------- > > > >The above electronic mail transmission is confidential and intended > >only for the person to whom it is addressed. Its contents may be > >protected by legal and/or professional privilege. Should it be > >received by you in error please contact the sender at the above > >quoted email address. Any unauthorised form of reproduction of this > >message is strictly prohibited. The Institute does not guarantee the > >security of any information electronically transmitted and is not > >liable if the information contained in this communication is not a > >proper and complete record of the message as transmitted by the > >sender nor for any delay in its receipt. > > > > > > -------------------Legal Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended only for > the person to whom it is addressed. Its contents may be protected by legal > and/or professional privilege. Should it be received by you in error please > contact the sender at the above quoted email address. Any unauthorised form > of reproduction of this message is strictly prohibited. The Institute does > not guarantee the security of any information electronically transmitted and > is not liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the sender nor > for any delay in its receipt. > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ------------------------------ > > Message: 9 > Date: Fri, 14 Jan 2005 00:51:50 -0600 > From: Steven Critchfield <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] How to set asterisk NOT to answer > incoming lines? > To: C F <[EMAIL PROTECTED]>, Asterisk Users Mailing List - > Non-Commercial Discussion <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > 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 > > <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > [email protected] > > > 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 > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Steven Critchfield <[EMAIL PROTECTED]> > > > > ------------------------------ > > Message: 10 > Date: Fri, 14 Jan 2005 18:00:26 +1100 > From: "Mike Sander" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Limit outgoing trunk calls > To: <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="windows-1250" > > Skipped content of type multipart/alternative-------------- next part > -------------- > A non-text attachment was scrubbed... > Name: not available > Type: image/jpeg > Size: 3649 bytes > Desc: not available > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20050114/b60b16 > 94/attachment-0001.jpeg > > ------------------------------ > > Message: 11 > Date: Fri, 14 Jan 2005 08:26:10 +0100 > From: "Florian Overkamp" <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] Agentcallbackogin > withoutanyuserinputafter extension is dialed. > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > 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 > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest, Vol 6, Issue 199 > ********************************************** > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > 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 [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
