similar to: Fwd: Re: very newbie question

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

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

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 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

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 >> >

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

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 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'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:

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

Howdy: I'm new to this samba stuff, but I've run across a problem that doesn't seem to be documented anywhere. It seems that OSR/2 clients (with all the M$ system updates; I already added the EnableClearTextPassword key) that have had their TCP/IP settings tweaked in the registry (eg, MaxMTU=576, TTL=32, RWIN added, etc) for DUN connections have mucho problems with a samba

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 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

I posted a similar issue, but this relates to how to keep a function call from being lowered to an ISD:FABS. In this case, I don't want FABS (floating point absolute) to be converted into an operation). In this case, what happens is that clang generates the code for floating point absolute (fabs) as a subroutine call: %5 = load douuble %x, align 8 %call = call double @fabs(double %5)

All, What i'm trying to do is setup a windows DUN connection via my asterisk box and over PSTN or VOIP to my work. What I hoped i'd find was a vitual modem driver for windows 2000 that wouldtalk over sip to my asterisk box and then act like a normal modem so I can dial out from that to our RAS service at work. Any One got any ideas as VPN is not an option for security reasons the DUN

When the world was young, Adam Snodgrass carved some runes like this: > Date: Mon, 10 Aug 1998 19:06:07 -0400 > Subject: Failure to execute programs from a mapped drive letter on a samba > I have a strange situation. I have a Linux machine running samba > 1.9.18p8, serving a mix of NT Workstation 4.0 and Windows 95 clients. In > general, things have worked flawlessly, with one

I am blessed with this user forum and able to set my Dev-PCI Digium card working fine with the Asterisk server of mine (i)But today i just wanted to know if someone can help me to set X-Ten Lite to call PSTN line using my FX0 Currently , I am able to use X Lite to call another X lite user locally (LAN) I also has attached my setting together Thanking you all in advance --------------