similar to: limit sip sessions

Hi People, what files is processed first?, balcklist or rules, i want to globally filter imesh, but at the same time allow managers to connect, i.e. , imesh work on port 1214, i have this: /etc/shorewall/blacklist #ADDRESS/SUBNET PROTOCOL PORT 192.168.0.0/16 tcp 1214 192.168.0.0/16 udp 1214

Hi, im running shorewall 2.0.16 with centos 3 (iptables v1.2.8), everything is working fine for several days, i have configured a masq lan and all the outgoing traffic is ok, but now i want to redirect (port forward) the external web traffic to an internal machine, somethig like this INTERNET ---------> SHOREWALL -------------------> INTERNAL_MACHINE [public

Hi, i cant make a three way call using grandstream phones (BT-100) and asterisk using sip, is this supported or i need a zap interface? thanks

Hi people, is there any good document on setup asterisk using only its voip functions , i mean without e1/t1 hardware, i have a cisco 5400 gateway and a cisco 7000 working as gatekeeper, can asterisk register with this GK?, im thinking in using cisco voip phones Any pointer will be apreciated, and sorry for my bad english Miguel

Hi people, what is the current status of send/receive fax on asterisk extensions, i dont want to receive the fax and send an email or viceversa, i want to connect a standard fax machine to a Linksys' ATA (FXS RJ11 port) . Webdoc?, pointers? Thanks

Hi, i have just migrated from courier-imap to dovecot. im using it with vpopmail, the pop service is working ok but the imap service for imp webmail is not writing the quota information to the maildirsize file, as recommended in the docs, i have this in dovecot.conf : plugin { # Quota plugin. Multiple backends are supported: # dirsize: Find and sum all the files found from mail

Hi list, I am a happy user of shorewall, i have followed the instructions in the shorewall''s web site relative to the squid transparent proxy configuration,all works ok, but i have been instructed to let adicional specified ports (aplications) to be snated (allowed to run) together to the web browsing service,i mean if i snat the network (i have a static ip from my isp), all

I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH

Hi all, Is there anyway to have more than 2 dummy interfaces? I need more of them but the system refuses to create it :P How can avoid this issue? Thanks. Luis Miguel Cruz. _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

Hi, this was my fault, you are right, i tried with a X-lite Professional and the conference (3-way call) is working now, i guess the phone BT-100 doesnt support it, i dont have a BT102D, so i can tell if it works too, bye -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of mmiranda@americatel.com.sv Sent: Friday,

I am trying to get the presence/hints/BLF working along with Realtime SIP but I never get any "busy" notification. core show hints always shows the realtime sip user as idle. I have tried setting call-limit to various values, including 1 but nothing seems to help. I have tried limitonpeers both yes and no. Anybody got any other ideas? I do know the hinting is working as I can

Hello Fach, I have used openvpn for a while and in the new release thereis a feature called "server mode" that makes posible to have a full network of vpn links besides a single TUN/TAP adaptor (a pure software NIC) in the server. I haven't used that feature, but I think this is what you need. Also openvpn runs on linux, *bsd, solaris, windows, and maybe in other OS. Miguel >

Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call

Hi, i have followed the idealx tutorial to the letter, however i get this error when i try to start slapd: ambepdc# /usr/local/etc/rc.d/slapd start Starting slapd. /usr/local/etc/openldap/slapd.conf: line 74: index type "uidNumber" undefined this is my slapd.conf ambepdc# cat /usr/local/etc/openldap/slapd.conf # # See slapd.conf(5) for details on configuration options. # This file

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

Sorry for the repetition, unless I've got bad deja vu this questions been asked before but I couldn't turn up an answer on CRAN. Is there already any code in existence for local dependence measures such as Moran's I or Getis and Ord G? Also, S-Plus has a number of interstingly named Cluster Algorithms based on some previous stand-alone fortran algorithms (agnes, daisy etc.) which

I just installed samba on a new server, 3.4.5-42, 64 bit version from Sernet, over CentOS 5.4. When running testparm, I get the following warning: rlimit_max: rlimit_max (8192) below minimum Windows limit (16384) I searched Google for some answer but I couldn't find a satisfactory one. What should I do to solve this? Can someone from the Samba team enlighten me on this? Thank you!

OK. So check out the Wiki here.... http://www.voip-info.org/tiki-index.php?page=Asterisk The archive of this list can be search via google by entering... site:lists.digium.com <some parameter> www.digium.com has a link to all the materials for getting started in the Documentation section of the website. Those are really quite good so I would start there. Most were written prior to

How can I copy all the contenent of the asterisk database to another machine? I want copy all the active sessions from one asterisk@home to another one and running on the second(this I can do using vrrp protocol, it isn't a problem), I want copy only all the active sessions and softphone registrations to another asterisk@home and then run on it. -------------- next part -------------- An HTML

I'm using Dovecot 1.1.7 on CentOS 5.2. I've changed my passdb from passwd to pam, it works fine, but I've found this messages on /var/log/secure: dovecot-auth: PAM adding faulty module: /lib64/security/pam_limits.so dovecot-auth: PAM unable to dlopen(/lib64/security/pam_limits.so) dovecot-auth: PAM [error: /lib64/security/pam_limits.so: failed to map segment from shared object: