similar to: "Say YES to continue" prompts

We've got a Toshiba DK system w/ analog ports that went to a voicemail server. I swapped in an Asterisk box with a Digium 4-port fxo card. It /almost/ worked perfectly. The problem is that Zap channels never hang up. They have to time out. I set up MeetMe, but all Zap channels hung forever. Very annoying. Same thing for FXO-to-FXO bridges. I figured out today why and fixed it.

We've done a direct swap of an old Amanda voicemail system with a shiney new Asterisk system (Asterisk 1.0.9). The system consists of 4 FXO ports on the * box (TDM400P), and three old Wildcards we aren't using (too buggy we found). CO lines-> Toshiba -> FXO ports on * We want to branch out a little more and use it as an auto-attendant. The first problem seems to be an asterisk

The CentOS drive for the HP NC4010 is built. I just needed to use external instead of USB power for my USB CDrom. Go figure... I did this in my hotel room, where they only have wireless. I went down to the 'business center' and used an ethernet drop, but I don't think the registration worked. I didn't know that even for the ehternet drops you have to do that portal thing

I'm posting here in case anyone else runs into this and needs some help. I'll probably update the voip-info Wiki pages on Toshiba integration in a bit. Asterisk 1.6 makes things a bit easier than what is on that page. I'm integrating an Asterisk server with a Toshiba Strata system at my office. Right now, it is driving some VoIP phones (Cisco ATAs with analog phones plugged into

Hello, community. Several months ago Toshiba Inc. released a new communicator - "Toshiba Portege G900". All device's features are greate, but in practice things are going another way: communicator is really SLOW. Toshiba users have found that G900 is using nVidia GoForce 5500 Chip in it and it seems like it's working partially, because of an incomplete driver. We have written a

Added a new function, "ask_yes_or_no", to ovirt-functions. It contains a default prompt if none is provided. Changed all prompts that ask for yes, no and/or abort to use this new method. Signed-off-by: Darryl L. Pierce <dpierce at redhat.com> --- scripts/ovirt-config-boot-wrapper | 8 ++--- scripts/ovirt-config-collectd | 39 +++++++++++++++-----------

Hello. I have some problems, with a new samba configuration. I want to setup a SAMBA PDC with ldap authentification, following an HOWTO found on the web. I used this howto with success on OPENSUSE 10.1 with a X86 processor and I have used it a lot of time (windows 2000 clients). Now I use OPENSUSE 11.3. And I test this new configuration with two new PC running X64 processor running linux. PDC

I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make calls from the Toshiba to Asterisk and internal calls from Asterisk to the Toshiba. What I can't do is make an call with an outside destination from Asterisk to the Toshiba. The Toshiba is looking for 9 to grab an outside line then it expects to see the 10 digits. In the FreePBX dial plan I use 9|. which sends 9 plus the 10

I have googled, and wiki'ed until blue. Is it possible to put T1---*----Toshiba CTX ? I have a TE405P, with one interface programmed for the T1, I am not sure how to program the 2nd port to mimick the T1 to the Toshiba. The Zapata.conf [channels] switchtype=national context=from-pstn signalling=pri_cpe usecallerid=asreceived echocancel=yes echocancelwhenbridged=no echotraining=400

I found some postings from Google (notably from Mark Spencer) about successful integration of a legacy Toshiba Strata system and Asterisk. I am also facing that current dilemma. The general legacy solutions that I can come up with is very easy -- either making Asterisk a "proxy" (or frontdoor) to the Toshiba system, or have it operate behind the Toshiba via regular extensions. I'm

I set up a TE405P to go T1---*---Toshiba. I have the channels configured, and can place calls from the Toshiba, through * to the t1. Incoming calls work great to *, but if they go to the Toshiba, I get a hangup. I think the * is sending the call to the wrong span. I have 2 spans, span 1 from the T1, span 2 to the Toshiba. The bchannels show as 0/1 through 0/23 on both spans in * when it starts.

I have recently changed jobs and I now have to administer a Toshiba DK424. I would like to integrate Asterisk with this Toshiba PBX. I am quite familiar with Asterisk but I have absolutely no idea about this Toshiba. To make matters worse the manuals for this system are missing. What I would like to do is place Asterisk in such a way that we can extend the Voicemail boxes and possibly add a few

Hi, I have a wierd problem regarding printing. I use the CUPS-backend on a debian-box to share printers with samba. Sometimes for no apperently reason, a win-box (XP-home) just can't print after a reboot... and no matter how many reboots the win-box makes it just stops working, and dosen't work until samba is restartet.... and its not a samba crash, because other win-boxes prints

Hi, I use Samba and cups on my debian-box to share printers, but sometimes, for no apperently reason, windows machines (XP-pro and XP-home), sudently won't print after reboot and this is only solved by a samba restart. Other times the first print after win-reboot, just gets lost, and never apper in the cups-printing log. It's not domain controled, but everyone is mounting their

Deat all, I am in middle of integrate Asterisk with Toshiba astrata legacy pbx. Following is my setup *Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx <-> Toshiba PBX* A =============================================> B C <============================================ D Asterisk PBX and strata PBX connected using back to back E1 cross cable. Physicall connectivity

Before studying your configs, what have you tried so far? Did you change this? Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. Here is the documentation on voip-info for why it may be the cause of your issues http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax span definition format:

As these problems are very time sensitive and frustrating, I suggest you document each change you make and do them one at a time so you can actually know what the problem was and not introduce new problems in the process. Find someone who is on the phone quite a bit and will give you an honest evaluation of the call dropping situation (unless you yourself are experiencing this issue too).

Hi, I am currently integrating our company's Toshiba PBX with the Asterisk version 1.2.1. I bought Quad T1 card, and making the port 1 to connect to PSTN PRI (use "pri_cpe" in zaptel.conf") and making the port 3 to connect to Toshiba PBX (using "pri_net" in zaptel.conf). The first stage goal is to just adding the Asterisk relay between PSTN and Toshiba system. The

I am new to winehq. I want to use it to run a program called Bestcase but it is not working. Do I just have to wait for something to happen to fix it or for someone to care enough about Bestcase to change wine so that it runs Bestcase? Is there something that I can do to possibly make it work? What are the steps required to get this program working under wine? I don't know what the

> > > Hi, > > > I checked by changing to from-zaptel, but no luck yet. Pls guide me on > this. > > Regards, > vudura senadeera > > > ------------------------------ > > > > Message: 9 > > Date: Fri, 19 Jan 2007 16:47:18 -0000 > > From: "Robert Jenkins" < raj@jrw.co.uk> > > Subject: RE: [asterisk-users]