similar to: uplink call quality issues

Hi there, Could you please remove me from your mailing list? Thank you Daniel _________________________________________________________________ Need a new place to rent, share or buy? Let ninemsn property help.

Check your slapd.conf ACL options. ____________________________________________________________________ --Forwarded Message Attachment-- From: fcd.listas@gmail.com Subject: [Samba] Samba + Ldap: password sync Date: Thu, 4 Oct 2007 15:19:42 -0300 To: samba@lists.samba.org Gentleman, While using Openldap 2.2.x, password syncronization was working fine. It means that when a user changed his

Hi, I am running dovecot 1.0.rc7 on a Suse Linux server. The server has approx 200+ mailboxes. Last week the filesystem (/dev/mapper/datavg/dat2lv) ran out of space - causing it to do into read-only mode. When I realised this I allocated some more space and re-booted the machine... Strangely it seems that dovecot is still having problems... It's like dovecot doesn't realise that

Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond....) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line

Hey, I am having trouble graphing the following function √2Γ(n/2)/[√n − 1Γ((n − 1)/2 for the values of n between 2 and 50. i know that Γ(n) = (n-1)!, which in R is factorial(n-1) When i type that into R, using y <- function(n)..... and then plot(y,2,50), it doesnt give me anything meaningful, in fact, it comes up with a message saying something like "in gamma(n+1) ploted" or

Does anyone know how to set the distinctive ring on the Linksys SPA941? I want to be able to dial one extension and have the phone ring with a certain tone and then dial another and have the phone ring with a different tone. I have tried the following ------------------------------------------------------------------- exten => 802,1,SIPAddHeader(call_info=Classic-4) exten =>

I am running * 1.0.5 and have been having lots of problems with outgoing calls and their sound quality. I am using ULAW for the codec and sixtel for termination. Basically the problem is that portions of the call seem to be lost and replaced with silence. Sometimes I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth

Hi All, Im just wondering if there are currently any plans to support UDF ISO images in PXE linux anytime soon ? Thanks. _________________________________________________________________ Advertisement: Are you paid what you're worth? Find out: SEEK Salary Centre

This is the last one, hopefully. _________________________________________________________________ Meet Sexy Singles today @ Lavalife - Click here http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Flavalife9%2Eninemsn%2Ecom%2Eau%2Fclickthru%2Fclickthru%2Eact%3Fid%3Dninemsn%26context%3Dan99%26locale%3Den%5FAU%26a%3D22740&_t=751140432&_r=emailtagline_meetsexy_june&_m=EXT

Helo List, I'm having an issue using the AND (&) operator in the code of my dialplan. The dial plan is coded to detect inbound DTMF digits from callers. key "1" is equivalent to "yes" and key "2" is equivalent to "no" in my dial plan. When a caller presses 1, yes is passed as a varialble and same when 2 is pressed a "no" is passed.

Hi I am making an extraordinary claim: rysnc seems to copy all my files, not just ones that have changed or new files. rsync version 2.6.0 protocol version 27 Debian 3.0 Woody I have tested this with one simple file, my example is shown below. Does anyone have any suggestions to rectify this? Regards Gareth example: I create a file called 'testfile' in /tmp/play containing the

I recently posted a question RE the Sipura 941 and using different ring tones, Thanks to hads I managed to use SET(_ALERT_INFO=Classic-1) to achieve this but trying this on the GXP 2000's didn't seem to do the trick?? Has anyone one had any luck on this topic? Also haven't been able to find any info on an auto-answer for the GXP 2000, again, I have succeeded in doing so with the

Hi List, I am experiencing an issue in a server that I have installed asterisk; configured an AVM FRITZ card to work with the capi module. Once istalled the card works perfect; however every time I reboot the machine I found that I have to re install the capi4k-utils before I can load asterisk otherwise the capi module will not loadup. Can anyone direct me in the right direction in order to

Hi all, I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. On an incoming call the following is produced in the Asterisk console with verbose 4 -- Starting simple switch on 'Zap/2-1' Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Mar 22

In the real world (or at least in my world) we use undersubscribed internet connections that come with a service level agreement (SLA) that guarantees that the jitter, delay, and packet loss with be within defined parameters in the service agreement. With most DSL and Cable you will not get a SLA, with the cheapest T1s you might get one, but the only penalty to the ISP if they do not meet is a

I have written up an guide on how to do bulk provisioning of the Linksys phones and ATAs. http://voipspeak.net/index.php?option=com_content <http://voipspeak.net/index.php?option=com_content&task=view&id=73> &task=view&id=73 Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 -

Ok, I must be really stupid here - I'm playing with ael and svn trunk. given the following in ael: context isdn10 { 444601 => { Answer(); NoOp(${CALLERIDNUM}); Hangup(); }; }; isdn10 is the incoming isdn context. why do I get this on the console: -- Accepting call from '01702xxxxxx' to 'yyyyyy' on

hey all, I'm getting this warning on the console when I leave a voicemail on my test server: [Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6552 vm_exec: Prefixing the mailbox with an option is deprecated ('u250@local-vm-users'). [Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6553 vm_exec: Please move all leading options to the second argument. This is what voicemail.conf

I was planning on using a TDM400P with 3 FXO & 1 FXS, with the 1 FXS being used for a fax machine. It now appears that Digium doesn't support this, are there other manufacturers anyone can recommend that will support it? Has anyone used a TDM400P in this setup and had it work without much issue? Thanks for the help, Ken -------------- next part -------------- An HTML attachment was

Hi all, I was wondering whether anybody here would help me clarify this minor issue please. If I have the following setup; Asterisk ------ Sipura SPA 3000 (fxo) --------- Pstn Line Would a call coming in on the pstn line be answered by the ATA or just get passed through to the * server (depending on dialplan) to handle? So basically, the caller does