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I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

...=> ProcessFax,n,NoOp(Result = ${FAXOPT(statusstr)}) I can't figure out how to turn on any debuging for this nor can I understand why this should be failing. I am using the spandsp version that Steve recommended (0.0.6pre18) Any ideas or feedback would be appreciated. Thanks Bryant Zimmerman (ZK Tech Inc.) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110124/6893ced8/attachment.htm>

...What could be causing this and how can I get it out of the system. Is there a patch I can apply to the system or is there a know fix for this issue. Right now I am stuck on this version because of some bugs in the current release that are show stoppers. I am on 1.8.2.3 build. Thanks Bryant Zimmerman (ZK Tech Inc.) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110614/306857a1/attachment.htm>

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use

--- Matt Zimmerman <mdz@debian.org> wrote: > On Fri, Jul 20, 2001 at 05:15:21PM -0700, Josh Coalson wrote: > > > I rethought it and it seemed like a bad idea to post > > a big file, so you can get it here: > > > > > http://cvs.sourceforge.net/cgi-bin/viewcvs.cgi/flac/junk/flac...

On Fri, Jul 20, 2001 at 10:51:11PM -0400, Matt Zimmerman wrote: > This version seems to work at least partially on ia64. I am able to encode my > usual test WAV file now, but I still get a segfault during the self-tests. Interestingly enough, when I recompiled with --enable-debug to get a stack trace, it worked. Any ideas how to track down the...

On Fri, Jul 20, 2001 at 08:14:55PM -0700, Josh Coalson wrote: > --- Matt Zimmerman <mdz@debian.org> wrote: > > On Fri, Jul 20, 2001 at 10:51:11PM -0400, Matt Zimmerman wrote: > > > > > This version seems to work at least partially on ia64. I am able > > to encode my > > > usual test WAV file now, but I still get a segfault during the &g...

...ic ----- Forwarded message from Chris Lattner <sabre at nondot.org> ----- Date: Wed, 14 Apr 2004 19:25:45 -0500 (CDT) From: Chris Lattner <sabre at nondot.org> To: llvmdev at cs.uiuc.edu Subject: Re: [LLVMdev] Linking strncpy Reply-To: llvmdev at cs.uiuc.edu On Wed, 14 Apr 2004, Eric Zimmerman wrote: > I'm working on a CS326 compiler project, and I'm having some problems > using string functions. Some LLVM programs produced are either > aborting or giving incorrect results; however, if I disassemble the > LLVM bytecode and recompile with GCC, everything works fine. &...

...to approach something like this. Do I just ignore the implementation details? Do I push the actual posting of data to a private method and mock that method so that I can verify that it is being called? Would you recommend I actually hit the web service? Any help is greatly appreciated. -- Jimmy Zimmerman http://jimmyzimmerman.com/blog/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://rubyforge.org/pipermail/rspec-users/attachments/20071221/4610b75e/attachment.html

Richard Zimmerman wrote: > hw wrote: >> Next question: you want RAID, how much storage do you need? Will 4 or 8 3.5" drives be enough (DO NOT GET crappy 2.5" drives - they're *much* more expensive than the 3.5" drives, and >smaller disk space. For the price of a 1TB 2.5", I can ge...

---------------------------------------- From: "Joshua Colp" <jcolp at digium.com> Sent: Sunday, October 4, 2015 12:12 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:09 PM, Bryant Zimmerman wrote: > -- > Joshua > Thanks for your reply. It thought the same thing, but when I change the > line in the corcery.conf to: > registration=realtime,px1_ps_registrations > Asterisk crashes and won't start. Here is what the log loop. > [Oct 4 16:04:18] WARNING[64823] config...

...ust get a beep no greeting played. We checked and the greeting files are there and play back from the voicemail ivr. If no greeting is there it just plays "The Pers.. beep" Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue goes away. Any Ideas? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131125/f2dde401/attachment.html>

I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a "core restart now" cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2

The only thing I can think of is that string.h is being #included and has different signatures for memcpy and strncpy. Possibly "char" is not signed on your machine (very unusual) or some of the parameters are declared as "const". Reid. On Wed, 2004-04-14 at 18:19, Eric Zimmerman wrote: > Chris, > > I'm fine with using JIT, but I'm trying to understand this problem: > 1. My LLVM program does not produce correct results > 2. Using llvm-dis, I disassemble the bytecode to C > 3. I recompile using GCC and the program _works correctly_. > > The...

Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant

It's been a while since I touched CVS, and things seem to have changed: > Cannot access /cvsroot/flac/CVSROOT The CVSROOT I'm using is :ext:<username>@cvs.flac.sourceforge.net:/cvsroot/flac Has the repository moved elsewhere? -- - mdz

Is there a pass that will transform this: %cc = seteq ushort %val, 0 into this: %cc = cast ushort %val to bool Would instcombine be the logical place to do this? Thanks.

Is there a way to drop a ip connection to asterisk after a number of register attempts. I have been having issues with hackers doing registration scanning against our server. We block their address at the fire wall but since asterisk does not force a drop of the connect after so many bad reg attempts I can't enforce the block until they drop and try again. This allows them to run the box

On Sat, May 19, 2001 at 06:42:22PM -0400, Matt Zimmerman wrote: > I think this could be fixed by changing the (data_len > 0) test to be > (data_len > 0 && total_error_X > 0). Yes, this appears to have worked. I can now correctly encode and decode both my 8kHz/8-bit/mono sample, and a 44.1kHz/16-bit/stereo sample on Debian/alpha....

On Sat, Nov 17, 2001 at 04:26:48PM -0500, Matt Zimmerman wrote: > On Thu, Nov 15, 2001 at 01:57:16PM -0800, Josh Coalson wrote: > > > > yes, the ones a month ago. it's not clear that this is even related to > > 3dnow but since it was happening on Matt's new AMD box and I don't have > > enough info I turned it off...