similar to: Asterisk in pass-thru mode

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

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.

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

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

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

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

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

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 using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18 Spectralink wireless IP phones. Most of the Spectralink phones have entries in 'sip show channels' that do not go away. None of the other phones do this. Is there anyway to remove these entries without restarting Asterisk? Any ideas on what could be done to prevent this? Example output: xxx.xxx.xxx.xxx 541

I would like to setup a system with two tennants with two seperate DIDs through one IAX provider account. Is it possible to route the calls into different contexts based on the DID dialed? I have searched and found nothing. I do not see anywhere in the console that says what DID was dialed so I am thinking two seperate accounts are needed to make this work. Can anyone confirm? Thanks

Hi, Andrew. You are trying to solve two tasks: definition through what line the call came and a beautiful display of this information. 1. definition through what line the call came. If the username and password for inbound and outbound registration the same, then try the following: a) delete "register" lines. b) add option "callbackextension=Company1" to Company1 friend

Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but it does work. For prosperity, the SIP service is through Internode. Here is my "extensions.conf" file: exten => s,1,Set(thedid="${SIP_HEADER(TO)}"); ignore this one exten => s,2,Set(pseudodid=${SIP_HEADER(To)}) exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)}) exten =>

On 21/03/16 15:44, Landau Daniil wrote: > 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

I'm experiencing one-way voice paths, followed by a hangup on one softphoine and not the other. Also, caller ID has odd outputs -- and I wonder if the problems are related. My configuration has Asterisk and a Linphone softphone running on the same box. I have a PC, and on that PC I use X-Lite or SJPhone to connect to the Linphone instance. When I call from the PC to Linphone: * I call

Hello I'm about to set up a voicemail system for multiple wholesale customers. So I use a realtime mysql config for the mailboxes. All single mailboxes have their information about the number, emailaddress, password in the database. This works fine. Now the notification emails of course should be customized per wholesale customer. I added a 'mandate' table to the database and

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

> On Sat, 27 Nov 2004 19:37:54 +0000, Corvin <corvin.dun@wp.pl> wrote: > > I have very simple question, how to limit SIP phone user making > > calls to for example longdistant calls? > > This is how I do it - Thank you very much to all of you. I have one more question which troubles me. We have scenario: (only SIP is considered now) Subscriber A registered in Asterisk

All, when I do show channels: Channel Location State Application(Data) SIP/201-e478 (None) Up Bridged Call(IAX2/muncie_to_ge IAX2/muncie_to_georg s@default:7 Up Dial(SIP/201|20) I am getting TRUNCATED call information.... the IAX2/muncie_to_ge is truncated. How do I get the need call information to transfer the call. Jerry

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.

I have some stuck channels that I think I'm going to have to bounce Asterisk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels: