similar to: very newbie question

> 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

Rafael J. Risco G.V. wrote: > > On Sat, 11 Dec 2004 16:49:12 +0000, Corvin <corvin.dun@wp.pl> wrote: >> Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisa?: >> > Hi. >> > >> > I need to set up H323 on an Asterisk box. I've succesfuly compiled the >> > asterisk oh323 (including of course all the dependencies: PWlib and >> >

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

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 all, I can see huge traffic here over 400 post in 4 days. My proposal is to create asterisk newsgrup proposal or phpBB forum what do think about it ? BR, Corvin btw. I'm admin of phpBB Forum (slackware forum - polish language), nearly 900 users. I think if someone will prepare it good it can be great project. (but I have 7 person team).

Soemthing goes wrong with this mail list: I am getitng something like it: >Sorry. Your message could not be delivered to: >Aster risk (Mailbox or Conference is full.) ?????????? Regards, Corvin

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

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

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 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

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 =>

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

Dnia sobota, 7 marca 2015 12:16:14 AM John R Pierce pisze: > I maintain a local mirror of the centos repository with a simple lftp > script, and configure my clients to get updates from this mirror via > the /etc/yum.repos.d files.... And why not rsync? -- Over And Out MoonWolf

Hi! I would like to use lcdproc and asterisk. Any hints or links? Maybe someone has experience in such matter. I am working on such solution. I've heard of SAPBX. Thanks for any help. Regards, Corvin

I'm helping a friend setup a small mailserver using dovecot, and I'm finding a strange problem with checkpasswd that I haven't had on my servers. How is the following debug output even possible? Jul 25 12:12:20 company2 dovecot: auth(default): master out: USER 5 joe home=/var/mail/joe.com/joe/Maildir/ uid=1005 gid=1005 Jul 25 12:12:20 company2 dovecot:

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, 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

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