similar to: permit syntax question

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 all! I have trouble with connection to AMI 1.1 wich enabled on Elastix "*Asterisk Call Manager/1.1* *Action: Login Username: admin Secret: qweasd123* *Response: Error* *Message: Missing action in request*" Elastix versions: "* Kernel* * Linux(x86_64)-2.6.18-348.1.1.el5* * Elastix* * elastix-2.4.0-1* * elastix-portknock-0.0.1-0* * elastix-agenda-2.4.0-1* *

Hello, I have tried to install this phone for hours now and I can't get it working. Maybe someone can help me :) I have searched for more info from everywhere but there isn't much about 7910 :( >From the CLI I get this: NAME ADDRESS MAC Reg. State ================ =============== ================ ========== telefon --

Hi all, I tested with few Asterisk versions from 1.4.18 to 1.4.21, same result. Here is the problem: I have a peer -which is peer AND user- setted up like this [MyPeer] ; type=peer host=xxx.xxx.xxx.139 deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142 permit=yyy.yyy.yyy.yyy/255.255.255.255 context=from-MyPeer dtfmode=auto disallow=all allow=ulaw,alaw

Hi all, can somebody explain me how exactly the "type", "host", "permit" and "deny" option in sip.conf play together? Where is the difference between "user" and "peer" ? I want configure SIP so that it is only from specified net section possible to make a call. I have tried following: [test] type=peer callerid=testaccount

I see in my log file this: Jun 30 21:44:26] NOTICE[42192][C-000002f3] chan_sip.c: Call from '' ( 5.189.144.120:5076) to extension '011972592675431' rejected because extension not found in context 'default'. which is great its rejected - however in my sip.conf file I have deny=0.0.0.0 permit=x.y.z.z/255.255.255.255 permit=a.b.c.d/255.255.255.255 So I'm expecting to

Hi I'm very new to Asterisk and this is my first posting to this mailing list. I got a Asterisk@home V2.8 working, and now I'm trying to use Asterisk.NET (http://sourceforge.net/projects/asterisk-dotnet) to get call events to my C# program. Asterisk.NET comes with a sample program called Asterisk.NET.Test and it uses the following default constants for login: const

Hello all, I'm wondering if there's any interest in trying to bond em1 and eth0 (respectively wired and wireless interfaces here), and if any, how to do it in CentOS6. I've found this, which could possibly help (but I'm failing yet): http://r.outlyer.net/linux:bonding https://help.ubuntu.com/community/UbuntuBonding Both are dealing w/ ifenslave. Any experience or point of

Valued Colleagues, I am trying to configure and use asterisk manager API. The /etc/asterisk/manager.conf and the output of "netstat -nl" are appended below. When I restart asterisk, I believe I should be able to see the asterisk listening on port 5038 using netstat. But when I type netstat, I don't see any applications listening on port 5038. When I telnet to port

Hi, Thank you Guus, I think the /16 solution is the easier to apply so I modified my tinc-up in host A to be like #!/bin/sh ifconfig $INTERFACE 192.168.10.1 netmask 255.255.0.0 a route -a from HOST A shows: Kernel IP routing table Destination Gateway Genmask Flags Metric Ref Use Iface 0.0.0.0 178.62.128.1 0.0.0.0 UG 0 0 0 eth0 10.129.0.0

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know

Hi all again, I've got a problem with sporadic one way audio calls, which means sometimes I can't hear the calling party (call is established, but audio is missing). Today I received ~90 calls, one of them got this problem. I've got two networks involved, without NAT: - 192.168.1.X, in there one nic of my server and all the phones - a private net to my provider, in there a nic of my

Dear Tinc Experts, I have been struggling for some time now, with Tinc pre3, and firewall rulesets and routing. I did once manage to get Tinc to work okay in a test-bed environment. I then tried to set it up for a 'real-life' setup and cannot get it to work properly. My real-life setup looks like this: Network A: 192.168.1.0 / 255.255.255.0 192.168.1.7 tap1 device gateway >

Hi all, The firmware I used is pa168s_iax2_us_151011.bin. My problem is the handset dial before I finished key in all the numbers, no matter how fast I managed to press the keys. It appeared it always dialed immediately, for example "011862", when I actually ment to dial 0118620xxxxxxxx. Thus left the remaining numbers "0xxxxxxxx" unsent. The handset had its dial plan

Hello. I'm new with asterisk. Can you help me in this: I have cisco sip phone (601) connected to asterisk server, and 1 client number (500). I want to dial from 601 to 500. But get error in cli console: [Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite: Call from '601' to extension '500' rejected because extension not found. What's wrong?

El 12 de febrero de 2016 16:51:59 CET, Eric Yau <ericyaukhy at hotmail.com> escribi?: >Hi All, > > > >I am trying to setup the site-to-site VPN with TINC for connect my home >network to company network. Here is the IP allocation and configuration >for >your reference. > > > >Home PC (192.168.1.2) ?-----? Home (OPENWRT Router, 192.168.1.1, >10.0.0.1)

Hi, after trying for days I ended up with a working tinc configurazion of 2 subnets, now my goal is to add 2 more subnets and comunicate. I might seem dumb at this point but honestly I don't work in IT or Networking stuff, and so I dont have that deep knowledge. A little explanation of my configuration is HOST A (VPN server) Public IP: 1.2.3.4 tun0 Subnet = 192.168.10.0/24 tun0 IP =

Hi All, I am trying to configure asterisk to receive an inbound SIP connection and send it to a specified context. Instead of sending the call the specified context, asterisk is using the context default from [general]? Any thoughts? I am sure that it is something simple I am missing. To recap, it is sending calls to the context default, not thecontext... [general] context=default

I''m trying to connect to two ISPs the connections are as follows: Eth0 IP: 192.168.1.1 Mask: 255.255.255.0 Eth1 IP: 216.167.217.241 Mask: 255.255.255.0 Eth2 IP: 216.167.208.40 Mask: 255.255.255.0 Here are the commands to set things up: #Load modules insmod /lib/modules/2.6.19.2- pyramid.metrix.net/kernel/net/ipv4/multipath_drr.ko insmod /lib/modules/2.6.19.2-

I''m trying to connect to two ISPs the connections are as follows: Eth0 IP: 192.168.1.1 Mask: 255.255.255.0 Eth1 IP: 216.167.217.241 Mask: 255.255.255.0 Eth2 IP: 216.167.208.40 Mask: 255.255.255.0 Here are the commands to set things up: #Load modules insmod /lib/modules/2.6.19.2- pyramid.metrix.net/kernel/net/ipv4/multipath_drr.ko insmod