search for: fccn

...rest of my script. Has anyone had this problem before? Or does anyone recommend a different GUI for intense number crunching? I have included details of my script below for further reference. Any advice is much appreciated. Thanks, Meagan Script objective/specific problem: Trying to read 'fCCN' values every second off of 'SMPS' data every 5 mins to calculate output of 'activation diameter' every second).Total time series is about 3 weeks, however, even when I try shortening the data to about half a day or so, I still have the same problem. Sometimes the GUI will close...

...3 channels) and Asterisk never sent the client a second call. How can I force Asterisk (or eyeBeam) just to send one call at each time. Is this a configuration I need to do in eyeBeam or Asterisk? Thanks Regards Joao Pereira -- ______________________________________________ Jo?o Gomes Pereira FCCN Av. do Brasil, n? 101 1700-066 Lisboa tel: +351 218 440 100 - fax: +351 218 472 167 email|SIP: joao.pereira at fccn.pt http://www.fccn.pt ______________________________________________ --- Aviso de Confidencialidade Esta mensagem ? exclusivamente destinada ao seu destinat?rio, podendo conter info...

Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten => s,1,Dial(SIP/74118@193.136.252.5,30,r) but this way all calls go to 74118@193.136.252.5 ..... Then I tried: exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r) but this way, the

...nter, the other shares area ok, and I can get files from this shares. Then I restart the deamon's (smbd and nmbd) and everything is working again. Someone know how to solve this ? Thank's Nuno Dias -- ____________________________________________________________________ Nuno Dias (ndias@fccn.pt) - WorkGroup LAN - Tel. +351 21 8440100 Funda??o para a Computa??o Cient?fica Nacional (http://www.fccn.pt) ?? Unix is user friendly, it simply chooses its friends wisely !! ??

I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close

..., and every ten minutes until it is answered. ------------------------------ Message: 3 Date: Thu, 2 Mar 2006 13:10:53 -0500 From: "Wojciech Tryc" <wojtek@VoIPMan.ORG> Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x) To: <joao.pereira@fccn.pt>, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <002501c63e24$a5f94780$4c45a8c0@kanatek.com> Content-Type: text/plain; format=flowed; charset="iso-8859-1"; reply-type=response Your pc has to able to s...

Hi all I have a pocketPC Toshiba e750 and I want to make SIP calls from it, but I didnt found any free softphones for my Toshiba. X lite's versions for pocketPC isnt free :( Did someone used before a free softphone for pocketPC? witch one? Thanks Joao Pereira www.fccn.pt

...always ask for the password on each share of the samba server, although the password is the same as the log on password. Is there any way to avoid this ? The first try always fails on the server, but I don't what user/passwd combination NT uses on that first time... Thanks, Joao Pagaime -- FCCN - Fundacao para a Computacao Cientifica Nacional - Tel: 351-1-8440100

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: Monday, September 17, 2007 12:45 PM To: joao.pereira at fccn.pt; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Center SoftPhone with Auto Answer Joao Pereira wrote: > But still, the user can choose not to answer the phone..... > I want to force the users to accept the calls. > You would also need a pho...

...calls. I'm now overwhelmed with the amount of SBCs that are suggested by the vendors to implement a solution. (http://www.juniper.net/solutions/literature/solutionbriefs/351085.pdf) Can anyone drop me some lines about this? I urgently need some feedback on this. Thanks! Joao Pereira www.fccn.pt

Switch is only tagging the vlan packets. Once the PC loads the vlan aware driver ("client") it will be able to read tagged packet for the vlan which PC has been configured to use. Nothing to be done on the switch. W -----Original Message----- From: Joao Pereira [mailto:joao.pereira@fccn.pt] Sent: Thursday, March 02, 2006 1:15 PM To: Wojciech Tryc; asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with802.1x) Ok, but the PC has an 802.1x client that configures the VLAN when he authenticates. Is this going to pass through t...

...ig adressbook with the friends emails, and we dont want to have this adressbook replicated (windows adressbook and Xlite phonebook). Thanks Joao ----- Original Message ----- From: "Walter Carter" <carterw@corp.earthlink.net> To: "'Joao Pereira'" <joao.pereira@fccn.pt>; "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>; <serusers@iptel.org> Sent: Friday, January 07, 2005 3:17 PM Subject: RE: [Serusers] softphones Try Xten: http://www.xten.com/index.php?menu=products&smenu=xlite...

Hello to all Im looking for a billing tool for Asterisk, that works with PostgreSQL. All the tools I found in www.asteriskbilling.com just work with MySQL :( Do you know a nice billing tool for Asterisk with PostgreSQL? Thanks Joao Pereira

Is it possible to implement a dial-out call queue in Asterisk? My idea is to give Asterisk a list of numbers, and then he makes the calls and delivers the calls to a call queue. Then, the agents will answer the calls. Is this possible? Thanks Regards Joao pereira

...emendously. I don't think it has nothing to do with the application because it was working fine on an older version of samba. Did anyone experience such problems ? What is the proper way to debug concurrency on samba - watching locks, measuring times, etc? Thanks in advance, Joao Pagaime -- FCCN - Fundacao para a Computacao Cientifica Nacional - Tel: 351-1-8440100 PS: I tried to eliminate locks - for test purposes - but smbstatus reports them anyway...

Hello I want to use Asterisk to implement a SIP Domain allowing my clients to do URI dialing and receive calls from the Internet through URIs and ENUM. My question is, should I put my Asterisk outside the firewall (in the DMZ) to allow connections to the Internet? Or should I have it inside my local network and put a SIP Proxy (like Openser) in the DMZ to implement the SIP domain? Thanks

Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao

...k switchport voice vlan 101 qos trust dscp qos trust extend spanning-tree portfast trunk -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Greg Oliver Sent: Thursday, March 02, 2006 6:24 PM To: joao.pereira@fccn.pt; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x) I have never used a switchport for .1x to a PC connected through a phone. I would say it probably will not work since it bypasses the idea of .1x ent...

I know this issue has been covered with at least 2 different patches, and probably a dozen different discussions, however I'm a bit unclear as to what my options are. I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall doing 1:1 NAT for machines behind the firewall. My asterisk box is one of these machines, and I'd like to allow foreign SIP clients

Hello to all When my SNOM (300 or 320) loses Internet connectivity, it loses its Asterisk registration (ok, thats normal). But when the phone is back online, he doesn't try to register in Asterisk. I believe this happens to avoid flooding the private LANs when the Internet link is lost.... but the problem is that the phones don't try to re-register in the future.... Sometimes it stays