similar to: Sip Trunking

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 need a feature from asterisk with dahdi channels, if there is an incoming call, it should ring on several dahdi channels. My channels look like: OFFICE1=DAHDI/13,,rtT OFFICE2=DAHDI/14,,rtT If I add this line: exten => 12345678,1,Dial(${OFFICE1}&{OFFICE2}) only OFFICE1 rings. If I change it to exten => 12345678,1,Dial(DAHDI/13&DAHDI/14) DAHDI/13 and 14 rings together,

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

Well, I thought I had the problem solved. Ported everything over to PJSip and build RDNS records for the phones and the server, but I am still experiencing the problem on incoming calls. ** On 6/7/2016 1:00 PM, Faheem Muhammad wrote: > I've faced the same issue. The issue was related to DNS, the reverse > lookup query failure caused the delay around(7-9 seconds). The purpose >

I am having an issue with a couple of phones where they ring, but there is a long delay after the phone is picked up before the audio starts. My setup: * Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC * Server is CentOS 7 * Quad core CPU with 16GB Ram * 2 Snom 300 phones. * NO NAT. Server and phone are on the same subnet with only a gigabit switch between them. * Digium

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

I'm having an issue with some Snom 300s on a server running Asterisk version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is _*NO NAT*_ involved. Phones and server are plugged into the same network switch, all on the same IP range. The server is running a Wildcard AEX410 analog card with 2 FXO modules receiving incoming analog lines. Occasionally, in the middle of a call, the

We're deploying an asterisk-based phone system at all of our branch offices in an effort to eliminate long-distance costs incurred from the constant branch to branch calls. We're using the Snom 300's at all offices for the desk phones and X100P cards to interface to 2 analog lines. I'm having a problem tuning all the echo out of the system. So far two branches are using the

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that

>> 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:

Is there any possibility of DAHDI supporting Automatic gain control on TDM ports? I'm having issues at a couple of offices where calls made to local numbers are fine but a when a calls from or goes to a large percentage of long-distance or 1-800 numbers the person at the remote end cannot hear the person in my office. Boosting the gains in zapata.conf (I'm still using 1.4.21) to 8

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

I'm having a major problem at one of my branch offices with "Phantom Rings" on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and

If I have a global variable in my dialplan and I change it, does that change immediately take affect for all calls that are active? Here is my situation. The company I work for has two office groups that share a building. The two offices are separate companies but support one another and want to be able to transfer calls as if they were all on the same phone system. Each company has 4

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

I think I have my echo problem solved, now i need to tackle the MWI. I can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a section from my sip.conf for my test phone: [general] context=internal allowguest=no allowoverlap=no allowtransfer=yes notifyhold=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes vmexten=9998 at internal ;vmexten=*97

I run VMWare server 1.07 on Centos 5. Last night I left a Windows 2000 virtual machine doing a ClamWin scan of drive M: when I went to bed around midnight. Drive M: is actually a volume on the Centos 5 host, mounted via Samba. It has about 40Gb of photos on it, plus a few other things. I had the VM up visible in the VMWare Server Console running under KDE on display 8 (X session #1), my