similar to: Voicemail prompts garbled

Hi there: I know this has come up before. I'm having trouble finding the fix for it. When I call into Asterisk to check voice mail, the prompt gets cut off. Example -- on the Polycom IP 501, I press "messages"; All I hear is the last part of the initial prompt, like so: "assword" So I enter my assword and listen to my messages. The prompts for incoming calls are

Hello, I am trying to connect an asterisk 1.6 to a trunking plate forme. With asterisk 1.4.x I added to sip.conf a line asking for registration in the form of: register = XXXXXXXXX at domain.com:Password:XXXXXXXXX at domain.com<assword%3AXXXXXXXXX at domain.com> @domain.com Unfortunately, as you can see, my usernames have to be of the form XXXXXX at domain.com which means that I had to

Hello Community, (sorry to be more busy, hence more running questions in parallel :) ) As mentioned in another post, I am testing mail_crypt plugin. I was wondering how to really secure the process sothat even the admin cannot have any access to the other users mail content. My current config is simple: - using per-folder keys (hence the per-user spaces are preserved) - put the

List, I purchased an SPA-1001 from voipsupply.com (which was great.. no hassle), but now anytime there is silence and then Playback() sound files the first half second gets cut off. For example, when I login to the VoiceMailMain() function I don't hear "Password", I hear "-assword". Then after I login I hear "- have no messages..." instead of "YOU have no

Hi, I have no problems whatsoever accessing and using my GNU Linux user directory from Windows Vista, but the reverse is not true. My /etc/samba/smb.conf has this (via testparm -s): ---<---------------cut here---------------start-------------->--- ~$ testparm -s /etc/samba/smb.conf Load smb config files from /etc/samba/smb.conf Processing section "[homes]" Processing section

!!! Regards For example, I tried the following command: samba-tool user create jhon p at assword -U mike Then, the user is created without authenticate the user mike Another command that I need execute with authentication is "samba-tool fsmo transfer". That's all the point of an AD domain : ) If any user could make change into AD database, the product would not be too much secure.

On Mon, Nov 28, 2016 at 01:56:04AM +0000, Gonglei (Arei) wrote: > Hi Michael, > > Thanks for your feedback firstly! > > > -----Original Message----- > > From: virtio-dev at lists.oasis-open.org [mailto:virtio-dev at lists.oasis-open.org] > > On Behalf Of Michael S. Tsirkin > > Sent: Sunday, November 27, 2016 11:33 AM > > To: Gonglei (Arei) > >

On Mon, Nov 28, 2016 at 01:56:04AM +0000, Gonglei (Arei) wrote: > Hi Michael, > > Thanks for your feedback firstly! > > > -----Original Message----- > > From: virtio-dev at lists.oasis-open.org [mailto:virtio-dev at lists.oasis-open.org] > > On Behalf Of Michael S. Tsirkin > > Sent: Sunday, November 27, 2016 11:33 AM > > To: Gonglei (Arei) > >

On Tue, Nov 22, 2016 at 04:10:22PM +0800, Gonglei wrote: > Virtio modern devices are always little edian, let's introduce > the LE functions for read/write configuration space for > virtio modern devices, which avoid complaint by Sparse when > we use the virtio_creaed/virtio_cwrite in VIRTIO_1 devices. > > Signed-off-by: Gonglei <arei.gonglei at huawei.com> > ---

On Tue, Nov 22, 2016 at 04:10:22PM +0800, Gonglei wrote: > Virtio modern devices are always little edian, let's introduce > the LE functions for read/write configuration space for > virtio modern devices, which avoid complaint by Sparse when > we use the virtio_creaed/virtio_cwrite in VIRTIO_1 devices. > > Signed-off-by: Gonglei <arei.gonglei at huawei.com> > ---

Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to

Hi all, This is my first post to the list. I have searched the net far and wide but can't find an answer to this problem. When I have call forward working or use the voicemail from a SIP phone, the first part of the message is always cut off. So instead of hearing "call forward cancelled" I hear "l forward cancelled". Or in voicemail I hear "edian mail"

Hi all, I''m struggling with the following: I have a defined type wrapped in a class: class myclass { define mytype(bla) { file { "blabla": .... } exec { "blablabla": ... } } } And I access the defined type in my other class by doing this: class otherclass { include myclass myclass::mytype { "alb": ... } # and here I want to

Would anyone know why Voicemail in * doesn't get the DTML keypresses from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do with "dtmf_avt_payload: 101" setting in SIPDefault.cnf in the tftp server? Thanks for any help! Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823

I had a compile problem with the CVS I downloaded on 21 June. I have a Debian box with 2.4.18 kernel (version needed for support of Conexant ADSL). There is a difficulty with Zaptel build regarding HDLC detection. It tries to build it in and then results in unresolved kernel symbols and fails to load. I have had to comment out the entire HDLC defines in zconfig.h to get a driver to install at

Samba 2.2.3a on OS-X 10.2.6 Trying to get a standalone server up and running with user level security. All the users have valid accounts and passwords, but we've enabled password sync and password chat and chat debug for the accounts as we don't do authentication against the active directory. unix password sync = yes passwd program = /usr/bin/passwd %u passwd chat debug = yes passwd chat

All, I'm having a heck of a time getting hdlc to work on kernel 2.6.11.9 .. I compiled hdlc, hdlc_gen, hdlc_cisco, hdlc_raw, into the kernel (note into, and not 'modules'). System comes up, I configured zaptel.conf span=1,0,0,esf,b8zs nethdlc=1-24 modprobe wct4xxp ztcfg sethdlc hdlc0 cisco ifconfig hdlc0 up All of this works fine, believe it or not. I have a T1 cross plugged

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 [To: flac-dev@lists.sourceforge.net] On Sun, 10 Nov 2002, Josh Coalson wrote: > Hmm, that's how the plugins work. They are using the file > decoder layer but that's a pretty trivial wrapper around > the seekable stream decoder. Without seeing your code there's > not much else I can say. Do you want to see my code ;-) >

Hello all, My set up is a fractionated t1, with 1-6 voice channels, 21-24 data channels. I have a t100P installed in a amd 1500 with 512meg. Question: In order to set up the data channels, is the following correct? 1) compile new kernel with generic hdlc support 2) compile hdlc from hq.pm.waw.pl/hdlc 3) uncomment KFLAGS+=DConfig_ZAPATA_NET in zaptel makefile 4) compile zaptel, 5 compile rest

All, I am trying to configure hdlc support on T100P Digium card - everything seems ok... but it just does not work at all. I was able to compile all drivers, the light on the card becomes green when I plug T1 link.... but I even can't ping default router IP... there is no data coming back to me, so I am confused completely now. Also I tried both protocols - raw hdlc and cisco hdlc. The