similar to: incoming call should ring on several dahdi channels

I have several branch offices all running Asterisk PBX's that register to each other via SIP so that calls can be transferred from office to office. Everything is working great on the office to office transfers, but I'd like to somehow make the CallerID more useful. Currently if an extension at Office1 dials an extension at Office2 the CID on the phone at Office2 says

Hello, I have 2 offices connected through VPN (all ipv4 and ipv6 traffic allowed), every office with it's own subnet. I built a DC in office1 for mydomain.local, built a second one in same office and joined mydomain.local with no problem. Then i built a DC in office2, but when i try it to join mydomain.local, the join process blocks at "Setting account password for OFFICE2-DC$" and

Hi Folks, First of all, I've been reading a lot and a lot about asterisk and ISDN-BRI, but is really a complicated world, not like PRI or Analog. I have 2 offices with 2 NT boxes and I want to connect them with Asterisk to route outgoing from one office two the other. At the end I want them to look like this Office2 will have only VoIP phones (not ISDN phones connected to a card in asterisk)

Hi, I was thinking about a common (I hope!) use case of Glusterfs geo-replication. Imagine 3 different facility having their own glusterfs deployment: * central-office * remote-office1 * remote-office2 Every client mount their local glusterfs deployment and write files (i.e.: user A deposit a PDF document on remote-office2), and it get replicated to the central-office glusterfs volume as soon

>> the fact that ssh insists on tap* and tun* tun/tap-device-names is a >> real nag and prevents from nice and easy solutions in some cases. > > Could you offer some examples? some client: ssh -o "Tunnel Ethernet" -w any office next client: ssh -o "Tunnel Ethernet" -w any office ...and so forth. interface configuration on the hub for all clients:

Hi, I need to connect 2 asterisks in 2 different countries (A and B) for one company so it's possible to make connections between the 2 offices. For connectivity reasons (NAT traversal) i want to connect the 2 asterisk with IAX so that when a user on office A connects via SIP to user on office B the call is going trought IAX channel. Can anyone give me an ideia how to accomplish

I have several branch offices, each with their own Asterisk server (version 1.4.22.1) handling their PBX functions. All of these offices need to talk to each other. In sip.conf I created a peer entry for each office with a username of branch-user and a friend entry for every branch-user with the username being just the branch, for example: [Office2] username=Office1-user host=10.10.80.253

Hi, I run into a problem and I'm not shure what do I misconfigure. I've a B410P ISDN card with bri_cpe signalling and two Openvox (A1200, A800) cards with fxo_ks signalling, all with dahdi drivers. I can receive fax from a public number, but I can't send fax. The CLI says it picks up the line but no dialing. I tried the extension with an analog phone, it works fine, I can dial

Dear Mr Petr Pikal I am extremely sorry for the manner I have raised the query. Actually that was my first post to this R forum and in fact even I was also bit confused while drafting the query, for which I really owe sorry to all for consuming the precious time. Perhaps I will try to redraft my query in a better way as follows. I have two datasets "A" and "B" containing the

Hello all/Timo, Up until now, my main Clients office has consisted of a single location, and I have never had to deal with the situation of multiple locations for a single company. They have just told me that they are acquiring an additional floor at a building that is about 4 minutes away - but obviously far enough away that I now have to deal with supporting users in the same domain but at

Hai, If the primary domain is set in windows, which is after domain join, it used that. Ipconfig /all and see primary DNS suffix. The dns suffix and first dns search list should be the same. Yes, other settings are possible, but stick to this for now. The Primay DNS suffix is used for the register of the IP in the DNS. The DHCP Service User MUST be a member of the DNSAdmins. The DHCP

Hey all, I hope I'm missing something simple. This is my second PDC install and I'm having some problems getting my win2k machine to join the domain. First I made the machine account: useradd -g 100 -d /dev/null -c "bob's computer" -s /bin/false office1$ Then I lock the password: passwd -l office1$ Next I make the smbpasswd -am office1 name. Finally, I added a

Mandi! Rowland Penny via samba In chel di` si favelave... > This is probably where you are going wrong. AD lives and dies on DNS, > your DC MUST be authoritative for the AD domain. ...but *is* authoritative! Simply DHCP server assign the ''old'' DNS, where all resolution fr the AD (sub)domain are forwarded to AD DNS... > Your AD clients should be using the DC as

I'm new to asterisk and need some help with getting a SIP connection working. I am trying to establish a termination point/DID number in another country. I am currently running Asterisk CVS-HEAD. My foreign provider uses SIP and authenticates via IP address. I am not required to register my SIP connection in order to send or receive calls. Can someone help me with how to understand the

Hi, I have a B400P BRI card with point-to-point connection (signalling: bri_cpe) with this dmesg: http://pastebin.com/sXrRt1yM When i restart asterisk server, the card cannot connect to the telco, the control led flashes red. If I unplug the cable between the ISDN nt and the card and wait 40 sec, the card can connect and works properly. The telco says the asterisk crashes the connection

Ok So when i get into a context and dial a number that does not exist the 'i' exten works fine. I have a test exten that runs my menu system if i dial that then dial an exten that does not exist ' i ' works. BUT when i dial a number directly that does not exist the 'i' exten does not get called i did try exten => ${INVALID_EXTEN}, Do summit ( no good ) What i need

In responce of Mourik Jan in this thing.. - The pc's boot that fast these days that the network stack is not loaded yet, which explains the "currently no logon servers available to service the logon request" which can be solved by setting the "Always wait for network " Setting in GPO. For Mathias, If your using sites and different dns domains. Like Office1.domain.tld

the fact that ssh insists on tap* and tun* tun/tap-device-names is a real nag and prevents from nice and easy solutions in some cases. the channel type used for tun/tap forwarding is tun at openssh.com, and it defines the remote tunnel id to be integer. so from what i understood these packet definitions are static per channel type, so for compatiblity the tun at openssh.com type has to stay,

Hai Santiago,   Your welkom, i hope i can help you out.   Post my your bind9 configuration, you can anonimize it if needed, but dont remove any lines from it.   And i need a snap of the log when bind is starting up. Like this one : Mar 29 16:42:58 dc1 named[21921]: starting BIND 9.9.5-9+deb8u10-Debian -f -u bind Mar 29 16:42:58 dc1 named[21921]: built with '?pr.... etc. .  Mar

Hi, I have been doing some tests using hping2 and TCP SYN pings targeting local and remote hosts from two FreeBSD 4.7-RELEASE-p10 and one Linux 2.4.18 host. The three machines have the same hardware configuration and have been running for 6 months now. The average load isnt too high (usually 0.01 to 0.15) on the FreeBSD machines. Here is the output from hping2 (excuse me the line wrap),