similar to: Asterisk Inbound calls, multiple SIP accounts, calledID

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.

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 =>

This is one of the chronic problems. Try this option in sip.conf: match_auth_username=yes Carefully read the description, it is better to test in "after hours". 02.04.2015 2:50, Andrew Galdes ?????: > 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. > >

Surely there must be someone here who can help me with this problem. I have spent weeks trying to get this damned service to work with no luck. I have incoming calls working, but no outgoing. If get outgoing working then incoming don't work. I have sent this problem to this list a couple of times with little or no response, and I _really_ need some help to sort it out. I have an asterisk

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 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

On Thu, Oct 05, 2017 at 02:57:18PM +1300, Clint Dilks (clintd at scms.waikato.ac.nz) wrote: > On Thu, Oct 5, 2017 at 2:41 PM, Jobst Schmalenbach <jobst at barrett.com.au> > wrote: > [snip] > Hi, > > Are you sure that your issue isn't related to the mirror that your > systems are selecting ? If they are using different mirrors I would try > using the

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. I have a company gateway that is connected to a 30/30 Fiber connection, network termination point is a MRV OS-904. It acts as a firewall/router for the DMZ/hosts/lans behind. Software: CentOS 6.9, bare minimum install, all latest patches. Hardware: Xeon CPU, Intel server MB with two Intel PRO 1000 (e1000, e1000e) network cards, adaptec RAID, 8GB RAM On the hosts/lan behind I can happily

http://bugzilla.mindrot.org/show_bug.cgi?id=1208 Summary: ssh fails to remove control socket when using with subversion Product: Portable OpenSSH Version: 4.3p2 Platform: ix86 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo: bitbucket at

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

https://bugzilla.mindrot.org/show_bug.cgi?id=2103 Bug ID: 2103 Summary: remote command as an option in ssh_config Classification: Unclassified Product: Portable OpenSSH Version: 6.2p1 Hardware: Other OS: Linux Status: NEW Severity: enhancement Priority: P5 Component: ssh

after install libguestfs_xfs, all results are: [using guestfish] guestfish -N fs:xfs -m /dev/sda1 statvfs / bsize: 4096 frsize: 4096 blocks: 24713 bfree: 23391 bavail: 23391 files: 51136 ffree: 51133 favail: 51133 fsid: 2049 flag: 4096 namemax: 255 [using virt-rescure] virt-rescue -a test1.img ><rescue> mount /dev/sda1 /sysroot ><rescue> stat -f /sysroot File:

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 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

I have an updated asterisk 1.8 server running on Freebsd 8.1, and connecting through a Freebsd 8.1 pf firewall with a dumb modem adsl connection (in other words FreeBSD is doing all the hard work). I am trying to connect with Internode nodephone, but they aren't really willing to spend the time to work it out (depending on who you get to talk to), and they reckon its all working as it

Hello, In order to overcome a known performance bug in perl-5.8.8-10 in centos 5 (see https://bugzilla.redhat.com/show_bug.cgi?id=196836) I downloaded the perl package from fedora 8 (http://mirror.internode.on.net/pub/fedora/linux/releases/8/Fedora/source/SRPMS/perl-5.8.8-30.fc8.src.rpm) and mod_perl

https://bugzilla.mindrot.org/show_bug.cgi?id=2308 Bug ID: 2308 Summary: Forwarded Unix domain sockets not removed on logout Product: Portable OpenSSH Version: 6.7p1 Hardware: All OS: Linux Status: NEW Severity: normal Priority: P5 Component: sshd Assignee: unassigned-bugs at