similar to: ABCTI: first usable beta

Hallo everybody, today i've released the very first (and very alpha :-) version of ABCTI, a CTI solution for Asterisk written in python, using pyGTK and py-Asterisk. It talks directly to the Manager API, so no additional software is required on the Asterisk server. You can find it on Sourceforge as http://abcti.sourceforge.net Basic functionality is already there: Originate, Transfer,

we are using samba Version 1.9.18p8 and want to connect WIN-NT 3.5.1 to Samba- supportet Unix-Drives; but for any reason the password for this connection is always needed (we did the registry for enableplaintextpassword); the same connection is done well on WIN95 could you figure out, what our problem is??? thank you Klaus-Dieter Wurm (dieter.wurm@de.origini-it.com)

I'm toying with the idea of writing a Samba-oriented Win95/98/NT logon window. I know it's possible to replace Windows' logon mechanism with another. Novell's Client32 does this. I know there's also standards to make a network service available to Windows' change-password mechanism (so I can fire smbpasswd on the other side). That could be an alternative to establishing

Hello all, this may be slightly offtopic :-) I have some Cisco 7940 phones with SIP firmware, connected to an Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g (HorstBox Pro with custom extensions.conf). On some of the phones, two lines are configured, one for business, one for private calls. When forwarding a line to another destination (e.g. to voicemail), we can't use the phone's own

Hi all! I am trying to figure out how DEVICE_STATE is working, no luck so far. sip.conf [0317998975] type=friend regexten=0317998975 secret=???? username=0317998975 callerid="Magnus Benngard" mailbox=0317998975 at inputinterior.se host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes disallow=all allow=alaw extensions.conf exten => 0317998975,hint,SIP/0317998975 exten =>

hi, i @asterisk - svn-ed asterisk from digium 1.6 - make install >> its running and i can access the CLI @gui then i -svned asterisk-gui from digium - installed - repointes apache /var/www/1234 >> /var/lib/asterisk/static_html >> now, i see the login box, but i dont have any credentials. tutorials are suggestion manager.conf, BUT I DONT HAVE that file, in fact /etc/asterisk is

Hi people! I am not getting really smart. I get the SVN Edition of asterisk GUI interface, compiled and love to get it to run, what won't work. What am I doing wrong?! svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0 make make checkconfig make install and If I open one of the URLs: http://localhost:8088/asterisk/static/config/cfgbasic.html

Gentlemen, Forgive me if I am posting at the wrong place! I was going to test the "new" chan_ooh323 driver so I did install: debian: Linux sip2 2.6.26-2-686 #1 SMP dahdi-linux-complete-2.2.0.2+2.2.0 Asterisk SVN-trunk-r231692 Did enable chan_ooh323, everything compiled without any problems. Hardware setup: Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975) X-Lite can

Hi there, I just installed the GUI for Asterisk 1.4.4 and correctly set my settings but when I use my browser to access it, it gives me an error saying "Not Found. Nothing to see here, move along" with "asterisk" in the header and footer... anyone had this problemn before? greetz

Hi! Did play around with queues and need some help. I thought that MEMBERINTERFACE and MEMBERNAME should be set to the ?device? the call was queued to not the device that called the queue, or do i miss something? Running: Asterisk 1.8.2.3 built by root @ sip on a i686 running Linux on 2011-01-31 13:38:23 UTC 0317998985 calls Kinna (0320209030) Tomas Ekman (SIP/0317998972) receives the call but

I am trying to connect Asterisk and Openfire together, but it's not yet working completely... I don't know for sure if my manager.conf-file is set correctly. I use this manager.conf file just to let Openfire talk to Asterisk... [general] displaysystemname = yes enabled = yes webenabled = yes (is this necessary for Openfire ???) port = 5038 bindaddr = 0.0.0.0 [openfire] secret=XXXXXX

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Hi! Part of extensions.conf: exten => 985,1,Dial(SIP/0317998985&H323/00702221448 at Avaya,20) exten => 985,2,Goto(985-${DIALSTATUS},1) exten => 985-BUSY,1,VoiceMail(0317998985 at inputinterior.se,b) exten => 985-BUSY,2,PlayBack(vm-goodbye) exten => 985-BUSY,3,HangUp() exten => 985-NOANSWER,1,VoiceMail(0317998985 at inputinterior.se,u) exten =>

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

Hi! We have alot of users who are having 2 phones, 1 fixed and 1 DECT. I am looking for a way to log them into a queue and let both phone rings. Let me try to explain: 0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue. queue add member SIP/0317998975 to 0317998989 works ofc. sip*CLI> queue show 0317998989 0317998989 has 0 calls (max unlimited) in

On Fri, Dec 07, 2018 at 11:52:38 +0100, Milan Zamazal wrote: > Hi, some custom CPU models are reported from > virConnectGetDomainCapabilities as usable='yes' on a physical machine > while as usable='no' inside a VM running on the same machine. That's > not completely surprising. > > But what surprises me is that those models are still reported from >

On Tue, Dec 11, 2018 at 10:45:12 +0100, Milan Zamazal wrote: > Jiri Denemark <jdenemar@redhat.com> writes: > > > On Fri, Dec 07, 2018 at 11:52:38 +0100, Milan Zamazal wrote: > >> Hi, some custom CPU models are reported from > >> virConnectGetDomainCapabilities as usable='yes' on a physical machine > > > >> while as usable='no'

Hi, some custom CPU models are reported from virConnectGetDomainCapabilities as usable='yes' on a physical machine while as usable='no' inside a VM running on the same machine. That's not completely surprising. But what surprises me is that those models are still reported from virConnectCompareCPU as supported (VIR_CPU_COMPARE_SUPERSET) in the nested environment and VMs can

Jiri Denemark <jdenemar@redhat.com> writes: > On Fri, Dec 07, 2018 at 11:52:38 +0100, Milan Zamazal wrote: >> Hi, some custom CPU models are reported from >> virConnectGetDomainCapabilities as usable='yes' on a physical machine > >> while as usable='no' inside a VM running on the same machine. That's >> not completely surprising. >>

http://bugs.freedesktop.org/show_bug.cgi?id=16342 Summary: NV15: Randr12 not usable Product: xorg Version: unspecified Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org