similar to: feeling n00b again

Hi, I have a share like this: [Accounts] comments = Accounting Dep common files browseable = yes writable = yes locking = no path = /mnt/accounts guest ok = no hide ureadable = yes create mask = 0777 directory mask = 0777 host allowed = 192.168.0.20 192.168.0.25 192.168.0.27 192.168.0.28 I wan't 'Accounts' to only be visable on the client's WORKGROUP listed

Hi all, Please excuse my newbie status I need help in configuring a mediatrix 1204 PSTN gateway with asterisk. Basically each FXO port is configured with a SIP username and automatic transfer extension, which should transfer incoming calls to an asterisk extension. I created extensions corresponding to the FXO port SIP usernames. Port 1 - SIP username - 21383396 - call forward to - 300

I recently installed CentOS 6.0 and Dovecot 2.0. My problem is that, when I download emails to my laptop (running Thunderbird), the emails are NOT deleted from my server. They stay there and are downloaded again and again. 8-( The "Leave messages on server" option in Thunderbird is not checked. I have several laptops that used to work when I ran CentOS 5.6 / Dovecot 1.x. I

Hi Friends, I have configured Postfix mail server on Centos for relaying mails from 5 linux servers (including itself) within the same LAN. The postfix mail server should relay mails from these 5 linux servers for specific domains only. For example hosts 192.168.0.23/24/25/26/27 and the postfix mail server should only be able to receive and send mails from and to example.com,example2.com and

i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX trunking between the 2 servers so that i dial -say from a sip extension 2000 on fedora server to a sip extension 3000 on CentOS server the call seems to be established but hangup automatically after very short time and here is the iax2 set debug command result on centos server and also my iax.conf and extension.conf and

Dear All, I use sipml5 to register two users from browser and the two clients are successfully connected. But when I made a call from one of the users, the other user doen'st have call notification and for a while the calling process ended. I check the /var/log/asterisk/messages got the following log: [Dec 17 14:54:11] WARNING[11471][C-00000000] chan_sip.c: Received SAVPF profle in audio

Dear tinc community, I am using tinc in switch mode. I have three nodes. Two nodes reside on routers, vpn-eth is bridged with internal lan, each router has several machines connected to it's internal lan. Third node is the roadwarrior - "endpoint" linux PC. When the roadwarrior is off - everything works perfectly, machines on both sides can communicate without a problem in any

The Asterisk Development Team would like to announce security releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3. These releases are available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk/releases

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

It?s been awhile since I set up my dovecot instance (like several years) and my transition from 1.x to 2 seems to have not gone well: all I can see is that authentication is banjaxed and I?m not sure what needs to be done to fix it. # 2.2.32 (dfbe293d4): /usr/local/etc/dovecot/dovecot.conf # OS: FreeBSD 10.3-STABLE i386 auth_debug = yes auth_mechanisms = plain login listen = *,[::] log_path =

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Greetings. I updated and old server to run Samba 4.9, It was running a distribution that still supported Samba 3.x. That Samba server has always been standalone, there is no interest in joining it to the Windows AD domain already in place. When it was running on Samba 3, users from a Windows domain joined machine, users were able to use the defined user on the Samba server to access the share.

Hi All, I have configured WebRTC according to the install document. The clients register correctly. I'm use SIPjs. The clients are able to send messages to the server. The SIP debug shows the messages being received. However I'm stumped for directions on how to route the messages between the clients. Asterisk 11.11.0 Here is my client sip config: [1060] type=friend username=1060 ; The

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

Second test after big upgrade.. -- http://www.PowerDNS.com Open source, database driven DNS Software http://netherlabs.nl Open and Closed source services

Hi all, Someone mentioned today to me, that the "competing virtualisation product" is capable of doing PKCS-forwarding towards a virtual client. So, my question here, does XEN supports PKCS-passthrough? As i also need my smartcard locally (on the hypervisor), i can not use neither pci nor usb-forwarding.... Defensie/CDC/IVENT/Research en Innovation Centrum Ing J. (Hans) Witvliet

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

im in Germany atm, that why your missing me atm tuesday im back. try this because its not needed to open a /16 subnet. edit /etc/default/ufw enable netbios conntrack in the modules below now run ufw reset now add the new rules. that should work before you do that, safe the output off iptables -S after the reset, run it again, and mail me both output's. Greetz for Germany

On Sun, Sep 29, 2019 at 3:33 PM Rowland penny via samba < samba at lists.samba.org> wrote: > >... > This is interesting, from a Unix domain member using smbclient it works: > > rowland at devstation:~/tests$ smbclient //192.168.0.27/rowland > Enter rowland at SAMDOM.EXAMPLE.COM's password: > tree connect failed: NT_STATUS_ACCESS_DENIED > rowland at

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >