similar to: Fax for Asterisk SIP-TDM

Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500

Can I install my free fax for asterisk license on more than one machine? I.e using my digiun account to download the free FFA module, am I restricted to just the first machine I put it on, or can I put the free FFA on multiple servers? Thanks, --Warren Selby

In what appears to be the most current documentation of FFA I'm directed to http://www.digium.com/en/docs/FAX/faa-download.php However the download selector utility found there "ain't working". Specifically, none of the drop lists appear to be populated (in any browser, on any platform), thereby preventing any actual 'selection'. Has anyone else noticed this? Is there

I've been having problems segfaulting when receiving faxes. So I tried FFA. I used the register utility and my license is in /var/lib/asterisk : ls /var/lib/asterisk/licenses/ -l total 4 -rw-r--r--. 1 root root 325 Jun 5 22:41 FFA-DNCXXXXXXX.lic but: Executing [17772822954 at FaxIncoming:4] ReceiveFAX("SIP/ec2faxcall18-00000002",

Hello, We succeed to send faxes using FFA, when the files are converted to tif from PDF using gs, but it doesn't work with tif files we copy/upload directly from our PCs. We saw in the manual that the size is important, since we got the error "FAX handle 0: failed to queue document 'filename.tif'", so we set it to 1680x2285, but it's still rejected. Is there a way

Just finished with the instructions from digium website/ net on how to compile FFA: After restart, modules did not get loaded so tried to load manually: [Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module: Error loadin ile: No such file or directory [Dec 18 14:31:26] WARNING[11002]: loader.c:653 load_resource: Module 'res_fax.so Verified the files exist: astbh00*CLI>

Dears , I am running asterisk version 1.6.0.6 with asterisk Fax Version 1.6.0_1.0.10, I installed on it two channels (one Fax for Asterisk channel and one Free Fax for Asterisk) My license files located at /var/lib/asterisk/licenses/ The two channels was working fine until I restart my PC,I did not change anyof my network cards or mac Address , How can I reload the channel license and

Has anyone experimented with increasing the DAHDI chunk size in improve fax reliability? If so, did it help, hurt, or not make any difference?

Hi all, I've got a perl agi script that exec()'s the FFA version of receivefax to... receive a fax. However, after the fax is received, the script seems to die. This is what I have: $main::agi->exec("receivefax","/tmp/${$}.tiff|fs"); $main::agi->verbose("FAX COMPLETE",1); I never see the "FAX COMPLETE" message on the console, I've set

Hi All, I seem to have stumbled on a bit of a problem. When trying to send a fax with Fax For Asterisk on 1.6.2.x (have tried 1.6.2.5, 1.6.2.7 and the current svn version, with FFA 1.2 I get a core dump each time. Here is an extract form the console: [May 12 22:47:09] DEBUG[22584]: app_queue.c:1084 handle_statechange: Device 'SIP/vltb-sbc01' changed to state '1' (Not in use)

All; I'm running Asterisk 1.8.15-cert3 with the newest version of spandsp. I've even tried unloading that and using Digium's FFA module but I receive the same error on an outbound transmission: [2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open_type: UDPTL (SIP/XXXXXXXXXXX_outbound-00000000): Buffer overflow detected (59 + 127 > 175) I only get this with one

Hi, what is the state at this time for 64bits applications and compatibility with 1.6.2 Mainly speaking about FFA, SFA, G729. Thanks for any information -- Daniel

Hi, I have a 1.6.2.6 fax installation with a FFA license which seems to be installed correctly (in fax show stats, I see that I have 1 Digium G.711 licensed channel, and 1 Digium T.38 licensed channel). When trying to call my business line with a fax machine, it looks like it's ringing to my asterisk box, then transfer the call to my extension. In the logs, I see (after the line where

Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in my extensions.conf: exten => 306,1,NoOp(Fax transmission) same => n,Set(FAXOPT(gateway)=yes) same => n,Dial(DAHDI/3) ----->FXS port to fax machine same => n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -->

Hello, We installed AsteriskNOW (Asterisk version 1.6.2.19) and purchased FFA and G.729 licenses and installed them successfully. When we tried to install asterisk addons (we need the CDRs), we get license conflict error messages. Searching google, we see that there's a way to force install the addons, but that "way" isn't properly detailed, so we didn't manage to do it.

Hi Guys, i am using the latest version of asterisk 1.4 (1.4.33.1), dahdi (2.3.0.1) and FFA (Applications: 1.4_1.2.0, Digium FAX Driver: 1.4_1.2.0). the issue i'm having is that i'm able to receive faxes from a website (that offer this service) but not able to receive from a regular fax machine (that is working perfect). [fax-rx] exten => receive,1,NoOp(**** FAX RECEIVE ****) exten

Hello, I have a stupid problem with scp and I can't find where it comes from. I am running 3 machines here with RedHat 7.3, RedHat 8 and RedHat 9. The 3 of them have openssh-3.7.1p2 (built from the SRPMS found on the openssh web site). I have though a problem to run a command of the type 'scp host1:file1 host2:file2' (my ssh being correctly setup): [ted at papeete ~]$ scp

After an extensive conversation with Mediatrx 's sales department , I stand corrected and so does the salesman who spoke to me. My apologies to Voip Supply. I understand now you never knew about the CD. Garrett Smith wrote: > I though I would post an update for everyone on what DOES and DOES NOT > come with every Mediatrix product. > > > > Every Mediatrix product,

I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan

This message is pointed directly to Steve Underwood. I tought it would not be nice to directly email him with a question that interests a good part of the Asterisk community, so here it is. :) Steve, remember a few days ago when we discussed about issues on Asterisk 1.6.1.13 and T.38 fax reception? Well I opened an issue on Mantis (https://issues.asterisk.org/view.php?id=16756) and turns out it