I'm trying to setup QoS for my VoIP line on my debian router box. I have tested wondershaper and to me it doesn't seem to work at all, so I'm looking for a better solution. Ultimately I would like something in the lines of [URL] but I guess such nice things doesn't exist for linux. Currently I'm on an ADSL link switching to cable in a few months.
View 14 RepliesDoes anyone notice a degradation in sound quality with skype and pulseaudio? It seems that when I use pulseaudio there are bugs in the sound quality in skype. I also notice a lag when talking to someone with the VoIP feature.
View 2 Replies View RelatedI am encountering a strange problem on my VOIP setup Basically, I have a asterisk appliance IP04. I have setup all the extensions and everything. I use a Linksys PAP2T as an ATA remotely. Now, my problem is the ATA sometimes is okay can call SIP and PSTN but sometimes I just can't hear anything. I thought it was my ISP blocking the VOIP packets but I have tried both the SIP softphone and IAX2 softphone on my PC. For IAX2, it works perfectly however in the SIP, I can hear the other end but they cannot hear me.
These are the ports I have opened on my router
1.) UDP 5060 - SIP Port
2.) UDP 10000 - 20000 - RTP Port
3.) UDP 4569 - IAX2
Do I need to open both TCP/UDP for these ports or UDP should be enough? These are the test cases:
1.) Using my WiFi Connection and a analog phone connected to ATA --> Sometimes working sometimes not and sometimes you can call SIP but the other end cannot hear you
2.) Using IAX2 in WiFi connection --> This one works perfectly
3.) Using a mobile phone connected to WiFi Network --> The same...but you can call and go out on PSTN but the other end cannot hear you
4.) Using a mobile phone connected via 3G --> Works perfectly but as expected it is quite slow and voice quality is awful
I want to use SIP rather than IAX2 because it is widely used and since my ATA doesn't support IAX2. Are there other ports I need to open or configure?
I just got DHCP working, but I need to add a "Pre-Defined Option" for the Cisco VoIP network here. At least that's what it's called in Windows. This "Pre-defined Option 150" is necessary on Windows dhcp servers tied to Cisco VoIP networks, but I have no idea where to start getting that set up in Fedora.
View 3 Replies View RelatedMy actual mail is via pop3 from verizon. What I've done so far is just pull my pop3 mail onto my main machine at my house. But I'm finding I'd rather have a single imap server on my network so then all my machines could see the same mail.
I don't really want the imap server to poll verizon for mail, I just want the imap server to check pop3 mail at verizon only when one of my mail apps checks into it to look for mail.
I am familiar with windows 2008 cluster servers, and I just started testing with centos cluster. I am creating a simple 2-node cluster, for a simple ping test.
Node 1: 10.0.0.1
Node 2: 10.0.0.1
Virtual ip: 10.0.0.10
So far, I can ping a virtual ip, and manually relocate it between the nodes, but I didn't figure out, how to do this automatically. So this is my question: How can I setup the cluster, to it automatically failover the a service to another node case one node fails?
My home networking consists of a slackware box, running iptables with a dual NIC. That's the firewall. I have a Netgear FSM 7352S, which is a level 3 switch, which I am currently just using as a switch. Clients are numerous PCs and a couple of networked printers.
The "firewall" machine is also a file server.
Here are the issues I could use some pointers on:
I'd like to assure that the VOIP adapters get priority, assuring QOS, particularly voice quality.
I'd like to provide reasonable priority for video streaming, such as hulu and other sources, that the kids use.
I'd like bulk data transfers (like backing up partitions) to the file server that runs iptables, and acts as the firewall/gateway for a cable internet connection. It would be good to be able to do this without impacting VOIP and video streaming.
I just setup a linux machine that act as a gateway along with squid running in transparent mode. Now I have one asterisk server which is behind that gateway I mean on my local subnet which pass through my linux gateway. Voip server having 4mb up n 4 mb down limit. Clients having 512kbps and upload 2mb.
Linux gateway : controlling band width of each clients
Squid acl forNAT issue with voip sites
Now my question is regarding skype calling. Since skype uses port 80, does it mean that it passes its request via proxy or direct and for safe side I've changed skype incoming port to 443 which squid does not see it. How much and width does skype use for calling in that case. Some one told me that it using squid to pass its request which I don't agree.
I was given a project of installing a new Jive VOIP PBX and will be migrating it from an older Avaya PBX. I need to perform in order to migrate the DID's and extensions and etc from the old system to the new? It is something that I have never done and have been ask to perform a miracle. I have never used JIVE VOIP PBX's and am familiar with Trixbox stuff but for smaller business and nothing of this size.
[URL]
I have Ubuntu 10.10 server gateway:
Code:
_______________________________
| ISP1 |<---->|ADSL modem, internal IP 192.168.1.1 |<------->|eth0 IP 192.168.1.10 |
|ubuntu server |
| ISP2 |<--------------------------------------------------->|wimax0,
[code]....
My goal is LAN must use ISP 2 to go to Internet and VOiP server must use ISP1. So, I write some iptables rules:
Code:
#!/bin/sh
#
IPT="/sbin/iptables"
# Internet Interface
[code]....
But there is problem: packets from DMZ network are not natting or may be something else wrong.Also my routing table:
Code:
# route -n
Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
192.168.1.0 0.0.0.0 255.255.255.0 U 199 0 0 eth0
[code]....
how do i send email in SSH but rather this:
How do i setup an email service? I have my server configured and everything (ip tables, fail2ban, able to access it, open ports, blah blah blah) and now i want to get me some emailing service on it!
I'm using tcpdump and tcptrace to track all incoming and outgoing data packets through my network interfaces. But I fail to monitor the voip data for skype that way, although it works well with http port 80, for example.
I want to track the ip address of the data packets for skype, i.e. know the ip address of the other one speaking at the other end of skype. How can I achieve this?
I've checked the port setting in my skype and I'm sure I'm listening on the right port. But nothing is showing up while I'm in connection with skype.
I have a complex network. ADSL broadband comes into the house and connects to an Orange Livebox. An Ethernet cable then connects the Livebox to a more powerful router, a DrayTek Vigor 2710Vn. The reason for this is that the Livebox has a second line capability using Voip, but it is not powerful enough to get around my stone house. The DrayTek router has Voip capability, but as yet Orange will not connect the Voip line to it. I connect to this system with Ubuntu. Android, Windows and I-phone. I can connect to either of the routers, though I usually use the DrayTek.
Voip on the Livebox does not require a computer to run it, you just plug a normal phone into it and use it to get free calls. I actually take this line into a Panasonic telephone switch to give me a 2 line system around the house. The problem with this set-up is that after a short time something happens to the network which prevents Ubuntu computers connecting to it. Windows machines, I-phones and Android phones connect, but Ubuntu does not. If I re-boot the Livebox, or in an extreme case take it back to it's factory settings, the Ubuntu machines can connect again, but it's only temporary.
The fact that fixing Livebox sorts the problem definitely points to Ubuntu being innocent, but at the moment I can't do without the Livebox. That means, for the moment, having to stay with Windows. If I post the output log after a failed connection attempt, all it would show is the connection timing out. Why is Ubuntu so sensitive to network problems that are not of it's making. Is there anything I can do about it other than changing my ISP. I am considering that, but other factors make that difficult.
I have two asterisk servers each one behind a linux firewall/gw. Linux is Centos 5.4, kernel 2.6.18-164.el5, iptables v1.3.5. Routes on the fws are ok and when iptables is stoped the servers are see each other, all good. But when I run iptables script in any fw, one server (not always the same) goes unreachable. I verify this with asterisk -r, then show sip trunk, and status becomes UNREACHABLE.
Iptables scripts is generated by fwbuilder. The weird part is I put only one rule to de script and it looks like Source=any, Destination=any, Service=any, Interface=any, Direction (Inbound,Outbound)=any, Time=Any, Action=ACCEPT. So as you can see I tried something like "Do not do anything at all". But anyway I run the script in any fw and one server becomes UNREACHABLE. I think the script does something wrong after all or maybe I have some missconfiguration in my asterisk conf files. The point is I am not so expert in iptables or shell scripting so I can't see anything in the iptables script. I have look for some issues like iptables blocking because of ip_conntrack table full, or "dont fragment" bit set in kernel problem, but nothing seems to be the right problem at all.
I have recently bought a IP/PABX system with one FXO and one FXS port. I intend to install this on a remote site with a public but dynamic IP (can be resolved via dyndns though) and make calls via clients that are NATTed (inside a home router). I would like to seek advice on the port opening and the recommended settings. I have been reading a lot on VOIP and I am getting feedback that SIP calls are difficult to establish on a NATTed environment.
1.) SIP port 5060 UDP?
2.) RTP ports - what range should I open for this? I see some use 10000-20000 UDP
3.) STUN server - Is this something that needs to be configured?
How can I ensure that the other party can hear the audio just like a regular telephone? Is it really impossible to do if the client is behind a router in which it is using a Private IP Address? What other network configurations needs to be done?
I am trying to accomplish an objective, I need to get 389 Directory server installed in the process however is it possible to just have email addresses of users in the LDAP server? I have a primary domain, example 'abcefg.com' with about 3,000 users and several hosted domains like 'xyz.com' and 'mno.com'. The end goal is to speed up email processing, so instead of the Barracuda asking the email server(s) if a user exist it queries the LDAP.
Is there a way to configure 389 Directory Server to contain just the email addresses of the end users to allow the Barracuda to poll the LDAP server to verify the email address is valid. The rest of the authentication like password is handled by the email server. So in the end the LDAP server would just contain the email addresses (live) and the Barracuda would query the LDAP asking if joebob@abcefg.com exist if it does carry on. If user xxeedd@abcefg.com does not exist drop it and go on.
I`m using Fedora 14 and i`ve one problem, i use x-lite phone on windows and what is x-lite alternative for linux ? i`ve found x-lite phone for linux but it dont work fine . It has problems with sound card etc. What do you recommend?
View 8 Replies View RelatedI am complete new to the technical side of VOIP. I know above diagram is not technically correct. I want a setup that works like that and oh the cheaper yet not compromising the better, even ekiga or skype can do that.
View 3 Replies View Relatedi have added a tomcat startup service using chkconfig. I have set up JAVA_HOME and JRE_HOME as well ass alternatives to point to the correct location.I have an init script in /etc/init.d. When i start the tomcat service using [ /etc/init.d/<x> start ] everything works fine and the environment variables for JAVA_HOME and JRE_HOME are used correctly.Now i added the service <x> using chkconfig --add so that it can start on boot up. But now the service does not use the system environment variable for JAVA and JRE setup by me.Also after complete bootup if i try to start the service using [ service <x> start] the environment variable in the shell are not used. don't want to exclusively declare variables for JAVA_HOME and JRE_HOME in the init script.
View 2 Replies View RelatedI've 7 PC's on my network. I've to setup an internal mail service, to send mails inside the network. I've installed Fedora 12 on one machine and the rest run's on Windows XP. What all packages do I need? How can I configure it?
View 2 Replies View Relatedi am looking for ideas to set up a calender/ scheduling system (or online service) for a doctor's office to set up patient appointments. i would like to set it up so that only the doctor, technician and receptionist can access it and set up appointments from 3 different computers.
View 2 Replies View RelatedIm running a virtual linux server on a low-end system, and i wold like to configure the server to boot in terminal window and not to the Gnome log in window, this is because of the low specs on the hardware.
View 5 Replies View RelatedI have a friend who is using Ubuntu 10.10 Netbook Edition on their netbook and well they love it. However, I notice that at times (if not all the time) their internet is ridiculously slow. Yes, it's a 1 Mbps download connection, but I still think it's a little slow and at times I can't even get it to load a website. I'm thinking if I adjusted the MTU within Ubuntu, it might improve, if even only slightly. I believe the router is set to 1490 or 1500, I haven't really looked yet.
View 3 Replies View RelatedWhat does the Quality, Signal Level and Noise Level actually mean? Which should I be looking for when looking for good APs?Is Quality the same as RXQ in airodump?
View 1 Replies View RelatedSo im pleased that my mobile broadband connection works but unfortunately the gnome/ubuntu way of displaying the connection strength is poor to say the least. Shown in this screenshot is my connection, 4 out of 4 bars.
BUT, does this mean i have a decent signal but only gprs? How do i tell if i have connected with 3G hspda gprs etc....?
Im trying to setup multiple domU through the default bridge setup. I am able to access only one of them through the network at a time. If you ping one of the domU it works perfectly but you cannot ping any of the others until you stop pinging the one and even then it takes a bit before you can. Ive looked around for a while and seen similar problems but nothing ever seems quite the same. Im probably missing something really stupid. Or is this the way the bridge is supposed to behave? Do i need to use a routed virtual network?
[Code]....
I have a problem during BOOT. During BOOT Fedora seems to stop at the following line:
Code: [20.245743 systemd[1]: unit systemd-tmpfiles-setup service entered failed state for more than a minute... and then it resumes.
I am running Server 10. I have a requirement to perform an action before the MySQL service starts, and perform another action after MySQL service stops.
I found the init script for MySQL under /etc/init/mysql.conf. I added my thing to the pre-start script there and works fine.
I am having trouble finding the script that stops the server so I can modify.
will the command service httpd restart the apache service if i enable httpsd service in graphical mode
View 12 Replies View RelatedI've been using Kubuntu Lucid on a Compaq tc4200 for nearly a month now. Suddenly today I logged into my machine and found that I couldn't connect to anything, Internet, LAN, router, nothing!Upon further investigation, I found that networking services weren't turned on at all!I tried manually starting the network service using
Code:
sudo service networking start
sudo /etc/init.d/networking start
[code]....