similar to: Multiple tennants, two DIDs, One IAX provider

Hi Dmitriy and others and thanks for your help so far. The option "match_auth_username=yes" seems to have had no effect. From my reading, this option will try to match the username of the incoming SIP account to a section heading. If that is how it must work then i can see a big problem. I'm trying to present the receptionist with a nice display of which line the call came in on.

Hello all, I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts with the same service provides. We have 8 phone numbers in total. Incoming calls from the public are all correctly directed to appropriate office handsets. However, the display on the reception phone (the only one i care about) is always showing the same "SIP/Account1_0843214321" rather than the account

Hi everyone! I have very simple question, how to limit SIP phone user making calls to for example longdistant calls? Maybe: Put in his context in sip.conf context which don't provide possibility to make such calls? Is it correct? thanks for any help, regards, Corvin

Dear All, Sorry to trouble again..... After go through www.lartc.org I have implemented the HTB instead of CBQ for the same scenario. Now following files are under /etc/sysconfig/htb directory. eth0 DEFAULT=30 R2Q=10 eth0-2.root RATE=256kbps BURST=25k eth0-2:10.comp1 RATE=120kbps BURST=12k PRIO=0 LEAF=sfq RULE=192.168.200.0/24 eth0-2:20.comp2

I have the Active Directory domain with Windows 2008 R2 domain controller and Samba domain controller on CentOS 7. Samba is 4.3.5 (self-compiled). Forest and domain levels are Windows 2008 R2. After joining Samba to the domain as the domain controller there were no DC=ForestDnsZones and DC=DomainDnsZones records on "OUTBOUND NEIGHBORS". I fixed it with ntdsutil, as it's written here

I have the following in sip.conf. It was my understanding that this configuration (ie with deny/permit) would only allow connections from hosts 192.168.10.4 and 192.168.10.5. That doesn't seem to be the case. Asterisk is accepting INVITE's from other addresses. [a00090101] type=friend context=Company1 username=a00090101 ;secret=180 ;insecure=very host=dynamic mailbox=company1@vmusers

Hi, Guus The use case the shared default gateway for multi-tenant, if that the case the node who own the default gateway will have problem to route with different tenant who has overlapped address scope? Is it true when no any other tools like the namespaces? (tenant1)\ (tenant2)——common node—— shared gw node—— Internet (tenant3)/ But if the each tenant have it’s dedicate default gateway, but

Good day all I want to put the openline4 card into a box that will support 3 different companies I read the caller ID id fixed but now HOW DO I: If a call come in for 12345 it plays company 1's welcome message If a call come in for 98765 it plays company 2's welcome message ens.. Does This make sense Thanks Altus

Hypothetical question.. Lets say there is a situation where you are using the highest compression codecs for all extensions (I guess that would be G.729) and the load on a single server is overpowering the most powerful single processor(lets say SMP is not an option).. So two or more servers are required.. Or The situation is that you need fault tolerance so want to have two Asterisk

Hi All, I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4 PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?, other detail is: this card have 4 card green. I need to know what is the best card for the following scenario: I need a IVR for my comapny and a PBX, but i want that my extension not use FXS I want IP phone . Thanks ins advanced,

Hi all, Below is what I did to run Asterisk in pass-thru mode: sip.conf: [general] disallow=all allow=ulaw canreinvite=yes For each channel, canreinvite=yes is enabled. No dial command has 't' option. However, it seems that Asterisk still stay in the media path and bridge the 2 end points. Am I missing something??? sip*CLI> show channels Channel (Context Extension

I have an * system running version 1.0.8 and it works mostly fine. I am using it as a virtual PBX and we share the box among companies. I have the calls all staying separate, we well as the companies' extensions, voicemail, etc. The only problem I'm having is with two accounts that use the same SIP termination provider. * seems to be confusing the sip contexts for the incoming calls.

On Wed, May 03, 2017 at 02:35:08PM +0800, Bright Zhao wrote: > The use case the shared default gateway for multi-tenant, if that the case the node who own the default gateway will have problem to route with different tenant who has overlapped address scope? Is it true when no any other tools like the namespaces? > > (tenant1)\ > (tenant2)——common node—— shared gw node—— Internet >

Hello all, One of the buildings I have an asterisk box deployed in is used by two small companies on two floors. They have an agreement between them whereby they'll answer each other's incoming calls and take messages if the office is empty / everyone is on the phone. Each of them has an ISDN BRI delivered to asterisk via zaphfc, then dropped into a context as follows: exten =>

Hello List, I'm dealing with the following issue here: https://forum.zentyal.org/index.php?topic=25300.0 Although it starts with OpenChange, it ends with Samba4 so I very much hope that somebody on this list can help me out. Basically I try to authenticate users through the Outlook autoconfigurator using RPC-OVER-HTTP to a samba server. The problem is that in Samba4/LDAP I cannot have

I am trying to get a ROR app running on a linux box running Apache 1.3 at my hosting comapny, but cannot get it running. There is a fresh/clean ROR app in the base directory / .htaccess, dispatch.cgi, dispatch.fcgi and dispatch.rb are all in / All dispatch files have been set to 755. ROR docs say to point doc root at /public/ and to put .htaccess, dispatch.cgi, dispatch.fcgi and dispatch.rb

oops, I hit the return and it got sent. All those questions depends on the application you setting up. if your worried about security, isolate the icecast server from your network. We designed a user interface on the company web site fronts the stream servers. The customers get a simple URL for the stream. We use several stream servers some on site and some off, our web server setup to server

Hi, please see the inline comments. On 02/21/2015 08:21 PM, Dean Sauer wrote: > I have a VPS on OpenVZ 2GB RAM / 2GB Burst, and Ubuntu 12.04 64b server > > I use Icecast 2.3.2 <snip /> > > As posted in previous threads I am seeing a situation reoccur where the > sources and clients ALL GET DUMPED...ALL AT ONCE. We'd need error logs from such an incident, perferably

I have a table of companies and a join table forming the relationship between a list of possible services they can provide. Working from a basic understanding of OOP I assumed that there should be a series of Serviceprovider child objects of the company class. Company < ActiveRecord::Base has_belongs_to_many :services end Serviceprovidertype1 < Company Serviceprovidertype2 <

I am new user of WINE and need a little help. 1. From the best I can tell I have everything configured correctly. However I can't figure out how to get a Windows screen to show up. 2. How do I get a WINE icon over to my desktop so as start it with a mouse click(s) Yes as you can tell I am very new to linux systems as I just installed this system this afternoon, so please be genle with