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.
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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 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]....
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 Replies View RelatedI'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`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'm currently working for a company that requires me to have an IAX-compatible VoIP client. So far, the only one that works, and the one that was suggested to me, is Zoiper.
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I can download Zoiper classic (which runs literally as an executable) or Zoiper communicator which has a .deb, but both of them are unreliable for everyday use.
I can run Zoiper in Windows XP no problem, but I'm a programmer and I work much faster with my native workflow in Ubuntu.
I tried installing IAX libraries in Synaptic and tried to see if Empathy, Pidgin or Ekiga would pick it up, but no luck.
I need suggestions for a good program (light and easy) that can handle both there protocols (voip (for skype) and SIP)
View 2 Replies View RelatedThe right direction with a linux Voip client that can make calls to a Microsoft MSN user.
I have looked at:
Sip Communicator
Ekiga
Empathy
GiZmo
Twinkle
and they dont seem to work well with MSN.
Alternate for Nymgo. As i use fedora 14 and only for nymgo, i have to log on to windows.
Nymgo is a software through which I can call home phone or mobile... it is just VOIP and not Video phone...
Also it is giving me much cheaper rate than Skype..
get cheap local and long distance phone service with a cordless handset around the house. I'm willing to invest some money into hardware. I have a small form factor Ubuntu computer with Atheros pci card that I use as my wireless access point. The only idea I have right now is buying a BELKIN F1PP000GN-SK Wi-Fi, which runs Skype with only needing a wifi connection. I to hear other ideas even if they are pretty technical
View 1 Replies View RelatedI have installed 2 VoIP programs, skype and xten softphone (X-Lite), skype works well and no problem but on the xten softphone not so lucky, I get to hear the tone before dialing like any phone but can't receive calls and if I call, there's no sound, other side can hear me, but I can't hear a thing.
I need both, skype it's my personal account and xten softphone is for work.
I am wondering if there is free VoIP software which will allow me to make and receive calls to regular phone numbers, such as landlines or cell phones, at no additional charge.
View 3 Replies View RelatedIs there another SIPS/VoIP program that can be used to speak to someone on Skype?
View 1 Replies View RelatedI need to build a small VPN server to connect one of my end VOIP devices out of the country, and am especially interested in a free or a limited Solution.
View 2 Replies View RelatedI'd like VOIP that works. VOIP is essential for my work, video would be very, very nice for keeping in touch with loved ones (keeping in touch with little nieces as they grow up, that kind of thing). It would be nice to get Skype working, but a more open alternative would be good too. Skype worked fine with Mandriva 2008.0, I couldn't get it working with acceptable sound quality in 2008. 1 in spite of following instructions. Now it seems broken in 2009.0 too, with no clear instructions on how to make it work - I just don't want to get into another vortex. Especially since, as I now discover, Skype does not support the open "SIP" VOIP protocol.
I gather that MSN & Yahoo both do support it, and Google Talk supports other protocols for text (not sure about voice) so I could connect through Ekiga, for example. I just set up Ekiga Softphone - don't know if it will work, but it's asked me to set up port forwarding without giving me the first clue of how to do it or which ports to forward. I'm also thinking about changing to Debian - I want a distro that lets me configure it at install, setting up the system to be very light and fast. (I'm not really a Linux Geek, but at least I figure Debian should be well documented, and stable and fast enough to make it worth the trouble.) Does that affect my choice of VOIP?
I migrated from 9.10 to 10.4 during the package installation I have the error: E: resolvconf: il sottoprocesso vecchio script di post-installation ha restituito lo stato di errore 1 ( In italian because I install the Italian version). The problem is: the file /etc/resol.conf I can't modify or delete from all user, root also.
View 1 Replies View RelatedI have a Windows 2k server running the AD PDC. 60 desktops users and 6 Windows servers use it as a single sign on server to login. As I plan to install a Centos 5.5 server with Samba for NAS and print sharing, is it possible to migrate the PDC services to the Linux server? I want desktop users and windows servers to authenticate on the Linux server. Is this possible? I have never worked with LDAP services. Worse, should I attempt this migration, it must be completed in a very short time frame.
View 5 Replies View RelatedI am trying to set up linphone for testing voip performance. Essentially I am a beginner with linux. I picked Fedora based on the recommendation of a friend. I am running Fedora 12 and I need to get linphone up and running with a working registrar. I have no idea what to do so I turned to hours of research which yielded little results but I did thing to try using partysip. I would like to know if this is a wise choice and if it is what do I do to set it up?
View 2 Replies View RelatedI have this HP 5610 all-in-one fax machine and have always had the issue of faxes failing either receiving or sending ever since I began using VOIP at home about 5 years ago. I have played with the setting of the fax machine and turned off error correction and etc and it has improved but mostly still fail. Voice quality if fine and my network has no problems that I can tell. I have read on the internet that this is a common issues with VOIP and fax machines. This issue is also happening at a friends house as well.
View 3 Replies View RelatedI've bought a Conceptronics CPHONELU (C01-200) VoIP USB phone. I would like to take full advantage of my VoIP phone in Linux. Currently I'm using Sip Communicator as the softphone, but I tried x-lite 3.0 on Windows too. The phone works has like a regular headset, I can hear and speak without a problem. However I cannot take full advantage of the phone, since the phone keyboard only work in Skype through the Skype Portal software that come bundle with the phone (Windows only). Is there a driver that I can use in order to be possible to dial using the phone and display the called ID with a regular softphone?
View 2 Replies View RelatedRecently a hav installed fedora and ns-2.34(after much effort).Now i want to generate VoIP traffic for wifi. give m url where i can find gud examples for generating VoIP traffic.
View 1 Replies View RelatedI need a voip tool for linux (fedora12). Any voip tool available for linux, if possible please tell me the step by step install/config.
My os details : Linux maniannam-server 2.6.31.9-174.fc12.i686.PAE #1 SMP Mon Dec 21 06:04:56 UTC 2009 i686 i686 i386 GNU/Linux
I would like to experiment with voip, is there any application which works successfully to make call from PC to phone, other than skype. If its beta also let me know, so that I can contribute my ideas to it. Any resource?
View 2 Replies View RelatedCan I migrate from fc12 to fc13 or do I have to do a full re-install? If I can migrate, can you point me to the instructions/working paper/etc.?
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