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L.S. I've some problems on a couple of my samba servers. The servers running CentOS release 5 with samba-3.0.25b-1.el5_1.4 The config is pretty standard, not much special in it. But the log files are filling up with error messages like this one: [2008/03/12 17:08:47, 0] smbd/service.c:set_current_service(184) chdir (/var/lib/samba/profiles) failed On the clients (all windows XP) are

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on

Hi folks! Our company are going to buy an E1 line with Euro ISDN and 30 lines (channels). This is how it will be configured: 3 Lines, of the total of 30, is going to be for the company phones, and share one phonenumber (eg. 555-12340). 1 Line will be dedicated to a specific unique phonenumber (Fax) (eg. 555-54321). The rest of the lines/channels (26) will be used by (by, not for) our customers,

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming

Dear R users, I am using lme and nlme to account for spatially correlated errors as random effects. My basic question is about being able to correct F, p, R2 and parameters of models that do not take into account the nature of such errors using gls, glm or nlm and replace them for new F, p, R2 and parameters using lme and nlme as random effects. I am studying distribution patterns of 50 tree

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd

Hi all, Firstly I need to apologise for some comments I made regarding GalaxyVoice and their service/abilities. Having opened another dialog with them they were more than willing to help out and tollerate abuse of their system whilst testing. Secondly I'd like to thank my anonymous friend without whom I would have never been able to get this going. Thirdly, I GOT IT TO WORK!! Below is the

database authentication table, to include per user quota's. Only then will the query I specified really work. In your present database, add the quota field and insert a KB value. This value will then be used when the authentication query is executed. In the example SQL query I posted, I used a tabled called quota_kb. One way to test the whole setup is to enable all the debug and verbose

>I went back to the example system direct from CVS with small >additions to sip.conf and extnsion.conf needed to make one >xten X-Lite phone work. I can dail in and hear the anouncements, >call the demo server at Digium. The audio quality I hear >comming from Asterisk back to X-Lite is good (9 on a 10 scale) >but the sound volume comming from the X-Lite extension is very low

modern technology, shorter durations could work. Most phones of all types don't make a standardized tone burst but produce tones only while the button is pressed. Fast "punching" will produce short tones. On the other hand, a redialed number will be very well formatted. Reliability of TT data transfer for audio applications (over the phone voice mail, credit card, IVR, etc) would

I got one email from eric asked me to Lower the rxgain and txgain on your Zap channels. But actually it is already the voice volume is low and I was looking to increase the gain (currently it is 0.0), so I do not know if eric was mean to reduce it less than 0.0, but I can not do that due to the low volume that is already existed, so any more reduce will make the voice not hearable well, even if

http://procontrol.fi/test/ - Crashfix for multipart messages - More optimized mbox rewriting, should be "good enough" for most uses I think. I've been using it for a while now, seems to work. Still some problems: imap(cras): Apr 15 01:48:48 Warning: Our dotlock file /home/cras/mail//full-disclosure.lock was overridden And I'm sure it wasn't. Or at least shouldn't have

Hi Dovecot: 2.2.19.rc2 OS: CentOS Linux release 7.1.1503 (Core) FS: XFS Problem: Dovecot it does not include line breaks in the letter. Bug or feautre? Message size in fs: 1125 bytes Message: From cras at irccrew.org Tue Jul 23 19:39:23 2002 Received: with ECARTIS (v1.0.0; list dovecot); Tue, 23 Jul 2002 19:39:23 +0300 (EEST) Return-Path: <cras at irccrew.org> Delivered-To: dovecot

Hi everyone, I'd like to use Asterisk to build a phonetree (www.phonetree.com) type of application, like this: 1. Read a text-based name/phonenumber file. 2. Call every number and play a recorded message. 3. If a beep is detected, replay the message from scratch (to leave messages on an answering machine). 4. Write results to a log file. Does anything like this exist already? Can this be

FEI: the CVS changelog is no longer updating. cras's commits look like this: date: YYYY-MM-DD HH:MM:SS +TZDD; author cras; state Exp; lines +1 -0 but cvs2cl.pl (2.50) demands: date: YYYY/MM/DD HH:MM:SS; author cras; state: Exp; lines +1 -0 cvs2cl.pl version 2.55 still can't parse these lines. I patched my cvs2cl.pl so I could still follow dovecot, but I'm not certain these

Hi All, After searching and didnt found it, im just sending my situation here, maybe someone knows where i should look. Im using Asterisk 1.6.1.10 Internally the user with a sip phone dials a number for instance 0623456789 It goes fine to the specific dial rule: which is: exten => _0[6].,2,Dial(SIP/0${EXTEN:1}#@xs4all-out,60,tTwWkK) This works fine without a charm, but the situation is that

Hello! My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on Fedora Core 4. I am making automatic call-out campaign with this setup on 4 PRI. The scripts for this: ==================================================================== caller php script write this to outgoung folder: fwrite($outfile,"Channel: Zap/g1/$phonenumber\n"); fwrite($outfile,"MaxRetries:

Readin : https://wiki.samba.org/index.php/Samba_AD_schema_extensions Is it an option to make an ldiff for the msNPAllowDialin and others on that Dail-in Tab. Im looking at the automount example. Hints tips? Greetz, Louis > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens > L.P.H. van Belle via samba > Verzonden: donderdag 14

device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote: Hi, We have few systems with asterisk 1.4.22.1 and we use sip

Hi list, I'm in central-europe and signed yesterday a broadvoice account. My Asterisk box is CVS 2005-08-25. Problem I face is: "Failed to authenticate on REGISTER to 'phonenumber@sip.broadvoice.com' (Tries 2)" then "Registration for 'phonenumber@sip.broadvoice.com' timed out" and finaly "Giving up forever to register