search for: lmoore

On Sat, Jun 18, 2011 at 6:40 AM, Larry Moore <lmoore at starwon.com.au> wrote: > On 18/06/2011 5:36 AM, Matteo Campana wrote: > >> >> Inviato da iPhone >> >> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling<EWieling at nyigc.com> >> ha scritto: >> >> We experience the same thing. The sol...

...;'m having trouble getting my virtual user setup to work. Can someone please tell me what''s wrong with this setup? # virtual.pp # class virtual_users::virtual { @group { "users": gid => "100", ensure => present } @user { "lmoore": password => "$1$WLx?v4oo#ajnxqrNG?ECT?00fh?P?H1", uid => "8050", comment => "Lucy Moore", gid => "100", home => "/home/lmoore", ensure => "present", shell =&gt...

...te but 2005comptrol does. adding user thodgkinson to group 2005comptrol getent group 2005comptrol:x:1005:Administrator,lchapman,hando,jcarroll,jchesnut,salbrecht ,shaddock,thodgkinson,sp2005,lpatterson,aladha,adore,ossis,cmikkelsen,cburto n,mhealy,scarscadden,lmccheyne,boardroom,bgamble,lmoore,wsmith,smorice,lkoza n,dfebbo,dpuff,gstevely,skihub02,kbuffel,krobertson,sp2006,sberry,lirvine,ti cket01,ticket02,ticket03,ticket04,ticket05,ticket06,ticket07,ticket08,ticket 09,pass01,tent02,pass03,pass04,tent01,pass02,skitech01,rental02,rental03,ren tal04,rental05,rental01,special01,special02,spe...

Hello everyone, I'm trying to send a received fax with mutt, when I try it from the Linux shel it works, but when trying with Asterisk's System command it doesn't. Successful Linux command: echo | mutt -s "New fax" earohuanca at gmail.com -a /tmp/faxes/201306191111.tif Unsuccessful Asterisk Command: same => n,System(mutt -s "New fax" elder.arohuanca at

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

Greetings everyone, I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream. In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information. I have noticed that when I do a MULTICAST page and send data

I am looking for a simple way to send occasional faxes via the FXO port on my SPA3102 -- without having to connect a fax modem to an ATA. In an ideal world, this would be some sort of "softfax" that runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with T.38. This is one of those things that I thought would be relatively straightforward, but a couple of hours of Googling

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

Hi all, I am running to the following problem, when using the below dialplan to receive fax, everything works perfect till this line exten => receive,n,ReceiveFAX(${FAXFILE}): and then the following line cannot be executed, it's like asterisk can't go back to dialplan and continue, the good news is when i check what is received in my fax folder i find that the file is a valid one (not

Hi! from 2 days I'm trying to run hylafax server and iaxmodem with Asterisk 1.8.5. I have 2 computers in the lan, one is the Asterisk PBX and the other is the server with hylafax and iaxmodem installed. In Asterisk I set up an IAX trunk in this way: ___________________________ iax.conf [iaxmodem] type=friend context=outgoing-fax disallow=all allow=ulaw username=iaxmodem secret=password

On 11/12/2014 4:52 PM, Recursive wrote: > Hello, > > at first, thanks for helping! > > In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was: > You may very well find getting T.38

On 2/04/2020 5:28 AM, Mark Boyce wrote: > On 1 Apr 2020, at 22:14, Greg Troxel <gdt at lexort.com > <mailto:gdt at lexort.com>> wrote: >> >> I think you need to use tcpdump and turn up firewall debugging. > > sngrep is your friend …My bet is UDP vs TCP on firewall rules :-) > > Mark Or the stateful entry still exists when the table entry is updated.

On 2/04/2020 6:35 AM, D'Arcy Cain wrote: > On 2020-04-01 16:28, Mark Boyce wrote: >> On 1 Apr 2020, at 22:14, Greg Troxel <gdt at lexort.com >> <mailto:gdt at lexort.com>> wrote: >>> I think you need to use tcpdump and turn up firewall debugging. >> sngrep is your friend …My bet is UDP vs TCP on firewall rules :-) > block drop in log quick on bge0

Just a thought regarding testing. Create a suitable TIFF file with more than 30 seconds worth of data and send it from Asterisk using SendFAX() to convince yourself whether Asterisk will work with your ITSP, you may still need to enable session timers. Have you considered setting up an extension on your Asterisk server which will receive the fax e.g. using ReceiveFAX() and see if your