similar to: Outbound registration expirey

Does anybory knows any work comparing R with other (charged) statistical softwares (like Minitab, SPSS, SAS)? I work in a brasilian government bureau and I intend to use R as our preferable statistical software, but I have to show it's as good as the others. I also intend to use Weka, and for this one I have the same problem. Can anyone help me? Thanks René M. Raupp e-mail: rener@mpdft.gov.br

Hello people, i'm brasilian. I'm sorry for the erros... ok?! :-) Let's go! My company uses openldap-2.1.21. What is the version samba compatible with openldap-2.1.21?! I'm trying install samba-3.0.11, but it requires openldap-2.2.23. I don't want install the release openldap-2.2.23 for it. Thank you.

Lilantha, the tones are supposed to be switched using the loadzone and defaultzone lines in /etc/zaptel.conf , and, progzone in /etc/asterisk/zapata.conf. The information about countries and frequencies/times are at zonedata.c located in the sourcecode of zaptel. As you may know, changing zonedata.c information requires a re-compilation of the zaptel module. Hope it helps, Ricardo

Is there anyway to expire password for windows clients with a Samba (in my case recent alpha version)? I found a post the TNG parameter "password expire time = 10" but a "strings smbd | grep expire" didn't turn in up in the binary. Anyone have any ideas if this is possble (I see it posted frequently to the archives, but no answers). Systems and Network Administrator

Hi list! Do anybody has success histories about using the Beronet app_saynumber application? For those that are hearing about for the first time, it?s a piece of software that enables the use of another language in say_number commands in asterisk dialplan or AGI scripts. Link to download: http://www.beronet.com/download/app_saynumber-beta-rc1.tar.gz I?m trying to compile it in

Hi list. I have a fine running Ser+Asterisk environment and have just installed ASTCC. It?s working fine either, including its caller-id authentication feature (the one we pass the card-number as CALLERID variable and number-to-dial as EXTEN variable). The issue, a great one, is that when the credit is about one minute to end, the ASTCC prompt gets into the call, says that "you have one

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know

Guys, when registering I am getting this error message, my question is that if this could be the reason whay I am able to make calls but not to recieve call ? [Apr 3 11:24:31] WARNING[19578]: chan_sip.c:15104 handle_response_register: Got 423 Interval too brief for service +506phonenumber at domain.co.cr@host.ip.addr, minimum is 3600 seconds Thanks -- http://celord.blogspot.com/

Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable

hi ppl, i get this error in my asterisk CLI: Aug 31 02:45:31 WARNING[26091]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register Aug 31 10:22:24 WARNING[26091]: chan_sip.c:9835 handle_response_register: Got 200 OK on REGISTER that isn't a register I guess it is related to my problem i have with one of my voip providers: i'm using asterisk 1.2

I have sent an e-mail to this list (awaiting moderator approval by the size) talking about some difficult to make calls with a SIP Provider in Brazil. I'm new at this list and have no sure if I have posted my question in the right place. If this is not the channel to make this kind of question about this issue, I'm sorry but want to ask if anyone can indicate the correct place . Thanks

Guten Tag, bonjour, sauda??es, hola, ahoj, welkom and of course hello. Ladies and gentlemen, the CentOS mailing lists are going international! Yes, now you can talk about your favourite linux distribution in your favourite language. If - and that is a big if - you are able to talk in - german - french - czech - dutch - the brasilian variant of portuguese - spanish Which should be about,

Guten Tag, bonjour, sauda??es, hola, ahoj, welkom and of course hello. Ladies and gentlemen, the CentOS mailing lists are going international! Yes, now you can talk about your favourite linux distribution in your favourite language. If - and that is a big if - you are able to talk in - german - french - czech - dutch - the brasilian variant of portuguese - spanish Which should be about,

What does that mean? Where can I get more info about it? Mar 18 07:19:51 WARNING[9563]: chan_sip.c:7549 handle_response: Got 200 OK on REGISTER that isn't a register

>All, >I have a TDm04b card an 8 SIP extensions. Calls come into the TDM and >are answerd by the auto attendant. When an extensions is entered I see >the Dial(SIP/100) on the console but the phone never rings... >I can pick up any extension and call 100 and it rings just fin>e and they >answer and everthing is fine. All 8 extensions can dial 9 for an analog line >and call

Hi! Here is my isolinux.cfg. It's slackware based. ====================== default Kernels/bare.i/bzImage initrd=initrd.img load_ramdisk=1 prompt_ramdisk=0 ramdisk_size=8192 rw root=/dev/ram ZILIX_KERNEL=bare.i prompt 1 timeout 1200 font lat1-16.psfu display mensagem.msg F1 mensagem.msg F2 opcoes.msg F3 help.msg label linux kernel Kernels/bare.i/bzImage append - label adaptec.s kernel

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I'm running Asterisk CVS-v1-0-06/06/05-17:29:02 built by root@............... on a i686 running Linux. I just spent some time in testing this. I tested the local and IAX2 trunks. Both worked flawlessly. Any comments? Darren Wiebe darren@aleph-com.net

Hi all. Just to update list and increase the "souls-save" database. The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working fine with ASTCC and "inuse" flag. The link of the patch is: http://bugs.digium.com/view.php?id=5400 Best regards to all you in the list. Ricardo Poppi.

I lost my internet connection today for a short time. During that time 1.2.12.1 stopped talking to my phones. Asterisk was still working as I got 2 voicemails. I have TDM analog cards for incoming calls. Anyway my cisco phones had X's (lost registration) and my uniden phones said "Registration error". Why would phones loose registration to asterisk when the internet connection