search for: niasoff

Displaying 8 results from an estimated 8 matches for "niasoff".

2004 Jul 30
1
FW: Limit incoming calls to SIP Channels
Hi All, Can someone please tell me how to limit incoming calls to SIP channels using the SetGroup & Checkgroup command. I don't want any call waiting on SIP channels and you are somehow meant to be able to do it with these commands. Many Thanks Daniel Niasoff
2004 Aug 01
1
Does anyone know how to use the DND feature oc Cisco 7940/7960
...says DND is set by pressing the services button and choosing DND. Does anyone know how to configure DND in the services.xml file. I've googled around and not found anything. When you enable it in SIPDefault.cnf it just allows you to use it once. Many thanks for all your amazing work. Daniel Niasoff
2004 Sep 12
1
SetGroup Limitation!!!
Hi all, I am just scratching my head trying to work out a way to use SetGroup to check busy status on a sip to sip call. The complication is that one call can't be in two groups so I have got no way of setting busy status on both the calling and called party. Has anyone got a way around this. Thanks Daniel -------------- next part -------------- An HTML attachment was
2005 Jan 10
1
SetGroup
Hi All, I use the SetGroup command to identify if a specific extension is in use. I create a group for each extension and check against that group name when putting through any further calls. A problem I am finding is that with internal calls I want to increment both the called and calling extension and SetGroup only appears to allow a call to be in a single group. Ideally I would like to
2005 Jan 11
1
Analogue RAS Server
Hi, Does anyone have any idea how to set up Asterisk so that it can act as an Analogue Remote Access Server. I've looked around and as far as I can see it will only act as an ISDN Ras server. Thanks Daniel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050111/696bfcce/attachment.htm
2003 Jul 27
20
g729 Codec
Hi, Do the g729 codec licenses for Asterisk work on a SIP environment (only SIP UAs running g729 + Asterisk)? I would like to buy a couple for a SIP test lab but I have not found any documentation on wether it works for SIP UAs or not. The Digium page only mentions: "The G.729 codec works with all Digium cards." Can somebody tell me please? Thanks, Ricardo Villa
2002 Jun 25
0
Samba & Redhat - Oplocks
...ferently as it's too much of a coinicidence. This is what I get. [2002/08/10 12:21:01, 3] smbd/oplock.c:oplock_break(890) oplock_break: returning success for dev = 303, inode = 2589371, file_id = 205 Current exclusive_oplocks_open = 0 [2002/08/10 12:21:01, 2] smbd/open.c:open_file(230) dniasoff opened file usr/package/etc/xinetd.d/swat read=Yes write=No (numopen=3) [2002/08/10 12:21:01, 3] smbd/process.c:process_smb(877) Transaction 768 of length 76 [2002/08/10 12:21:01, 3] smbd/process.c:switch_message(684) switch message SMBtrans2 (pid 1259) [2002/08/10 12:21:01, 3] smbd/trans2.c:ca...
2005 Jan 11
0
RE: Asterisk-Users Digest, Vol 6, Issue 142
...n <asterisk-users@lists.digium.com> Message-ID: <41E3D58A.1070801@lumiss.hr> Content-Type: text/plain; charset="iso-8859-1" I don't think it's possible. Asterisk would have to emulate analog modem, and I believe that feature is not (at least yet) implemented. Daniel Niasoff wrote: > Hi, > > > > Does anyone have any idea how to set up Asterisk so that it can act as > an Analogue Remote Access Server. I've looked around and as far as I > can see it will only act as an ISDN Ras server. > > > > Thanks > > > > Daniel...