similar to: QoS & VPN

Well, I thought I had the problem solved. Ported everything over to PJSip and build RDNS records for the phones and the server, but I am still experiencing the problem on incoming calls. ** On 6/7/2016 1:00 PM, Faheem Muhammad wrote: > I've faced the same issue. The issue was related to DNS, the reverse > lookup query failure caused the delay around(7-9 seconds). The purpose >

I am having an issue with a couple of phones where they ring, but there is a long delay after the phone is picked up before the audio starts. My setup: * Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC * Server is CentOS 7 * Quad core CPU with 16GB Ram * 2 Snom 300 phones. * NO NAT. Server and phone are on the same subnet with only a gigabit switch between them. * Digium

I'm having an issue with some Snom 300s on a server running Asterisk version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is _*NO NAT*_ involved. Phones and server are plugged into the same network switch, all on the same IP range. The server is running a Wildcard AEX410 analog card with 2 FXO modules receiving incoming analog lines. Occasionally, in the middle of a call, the

We're deploying an asterisk-based phone system at all of our branch offices in an effort to eliminate long-distance costs incurred from the constant branch to branch calls. We're using the Snom 300's at all offices for the desk phones and X100P cards to interface to 2 analog lines. I'm having a problem tuning all the echo out of the system. So far two branches are using the

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there any possibility of DAHDI supporting Automatic gain control on TDM ports? I'm having issues at a couple of offices where calls made to local numbers are fine but a when a calls from or goes to a large percentage of long-distance or 1-800 numbers the person at the remote end cannot hear the person in my office. Boosting the gains in zapata.conf (I'm still using 1.4.21) to 8

I have several branch offices, each with their own Asterisk server (version 1.4.22.1) handling their PBX functions. All of these offices need to talk to each other. In sip.conf I created a peer entry for each office with a username of branch-user and a friend entry for every branch-user with the username being just the branch, for example: [Office2] username=Office1-user host=10.10.80.253

Hi list, I am trying to create a VPN between two different locations. On the first location we have a cisco pix 525 Natting the internal 192.168.100.x network, while on the second location we have a Centos3 box Natting via iptables the internal 192.168.10.x netowrk. My goal is to connect this 2 over the internet via IPsec. I created the IPsec Net2Net via the network configuration graphic

I'm having a major problem at one of my branch offices with "Phantom Rings" on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and

IF you are not stuck to IPSec, you might want to take a look at OpenVPN (www.openvpn.org). I found OpenVPN easier to install than FreeSWAN (an IPSEC VPN) and have setup an OpenVPN solution between my German office and our mainoffice in a matter of hours. Thom van der Boon E-Mail: Thom.van.der.Boon at vdb.nl ===== Thom.H. van der Boon b.v. Havens 563 Jan Evertsenweg 2-4 NL-3115 JA Schiedam

If I have a global variable in my dialplan and I change it, does that change immediately take affect for all calls that are active? Here is my situation. The company I work for has two office groups that share a building. The two offices are separate companies but support one another and want to be able to transfer calls as if they were all on the same phone system. Each company has 4

Hey Everyone, We are in the works of planning a new * installation for our company. We have 20 users in our main office and 5 users in a remote office a couple of states away. Our call volume for the main office will be anywhere from 5-10 concurrent calls. The remote office will have about 3 heavy users with two users making calls occasionally. Right now we have an existing PBX. We have a

Hi List Just a little bit OT, but then again perhaps an information that could be of great value for a lot of administrators !! Does anyone have experience with how to setup VoIP QoS for outgoing data through a Cisco PIX (515) ? I believe that it should be possible to give higher priority to outgoing VoIP packets. This is due to the problem of ADSL as the UpStream data rate is 1/4 of the

Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra

Hey All, Sometimes we are running into issues with calls randomly being disconnected. We have a couple of phones is in a remote office. We have a PIX-to-PIX VPN setup between the offices. I know neither of these devices provide QOS but we usually don't run into any problems. I notice this problem when we are in a conference call with one of the employees at the remote office and a

Hi, I just wonder if I can replace PIX firewall with machine having IPTable as my firewall. I want to control our firewall but due to the lack of knowledge in configuring PIX, i want to replace it with Linux or BSD. I do not have the time now to learn the complexity of PIX and yet I want to know what happening in my firewall and control it as much as possible. Do you think it is justifiable for

Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a "call forward" option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them "try out" our dialtone

Hello, long time lurker, first time writer.... We have the following set up ITSP | | Internet | | Cisco 2600 | | Switch----Asterisk Server running 1.0.9(has public ip) | | Cisco 515e Pix Firewall running Pix OS 5.3(run's a class c 1-to-1 nat and pat) | | Grandstream GXP-2000(run latest fw from grandstream site 1.0.1.9) The grandstream registers with the public asterisk server fine. I even

Hi All Could a VPN be used to help with SIP Tunneling and QoS issues. State 1: Two IP Networks Connected via the Public Internet transmitting VoIP Traffic Say a VoIP User and VoIP Termination Provider. Each side can put QoS onto their part, but if QoS does NOT exist between them then call quality will be bad anyhow. State 2: Same as above except a VPN tunnel is setup between each side. Thus

Hello all, It would be a great feature if QOS could be implemented in Tinc. It's not to do any sort of comparison, but with OpenVPN, the traffic can be shapped. In the way that tinc works as mesh VPN, the great feature would be to fix a shapping value on each nodes, known and respected by each others. (like MPLS does) As an exemple, you've got 3 nodes A,B and C, you want Tinc flows on A