I 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 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 RelatedI 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 RelatedUsing 9.10 on dual boot P4. Try to use a USR 9602 voip phone plugged into USB port (this device works on WinXP)It does not show in Sound Preferences. I have tried on a different computer with U9.10 and same problem
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I recently set up a Debian-based gateway+router on a remote site. I've installed OpenVPN and made a VPN bridge to another network (that server is also Debian). The main network has all the resources and also a VoIP server (asterisk). Bridge seems to work fine, except that every time a Linksys phone is used to call - after a few seconds I get:
Jul 20 12:16:05 sklad kernel: [403987.817695] eth0: link down
Jul 20 12:16:05 sklad kernel: [403987.817939] br0: port 1(eth0) entering disabled state
Jul 20 12:16:07 sklad kernel: [403990.113701] eth0: link up, 100Mbps, full-duplex, lpa 0x4DE1
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So whenever the phone is used, network falls down for a few seconds. And of course this breaks the call. There is also another Linksys box (analog-to-voip) and it works fine most of the time. However sometimes this "eth0 link down" happens too. There is nothing more in syslog to analyze, so I don't know where to dig. Physical device of eth0 is D-link DFE520.
Today my boss come to me and ask me to get a cote to upgrade our old "Cisco call manager" (2004) now "Cisco Unified Communications Manager". So I was wondering, instead of doing a costly upgrade (over 35 000�), maybe it's time to change... Does anyone of you got some insight with Asterisk in an enterprise environment? Is it reliable? Following you own judgement, what are the + and - ? If Asterix worth it, what argument (apart of the price) could I use to help the management turning on my side? Will the Cisco 7921 VOIP phone will be able to connect to it? (as we do have over 35 of them)
Enterprise environment:
- 3 sites (VPN interconnected)
- ~35 VOIP phones and ~10 landlines phones
I am looking for a way to use my computer as a regular landline telephone not by VoIP but by actually dialing on my landline to talk. I don't know what software I need. My searches only bring up VoIP stuff leaving me stuck without answers. I am using a new installation of debian 5.0. I think that's all, if you need more details I can add them.
View 1 Replies View RelatedI`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 RelatedMost of the BT stuff I see involves connecting a headset to the PC...I'm not seeing much for what I want to do. I have a motorola Droid, and it's pairing via bluetooth to my thinkpad just fine. I'm running Ubuntu 9.10 64bit.What I want to do is pair the PC to my phone and have the PC act as the headset, i.e., the audio from my phone comes out my PC speakers. Ultimately I'm trying to record audio from my phone, voicemails from Verizon that I'd like to save to audio files. A couple years ago I was able to do something like this with a Thinkpad in WinXP, and record audio using Audacity. I'm not seeing how to make the PC work as a headset.
View 4 Replies View RelatedI'm using it on a small IBM machine in my closed ( ) which is my multimedia center, file sharing, router, http server & others. To escape the cable paradox, lately I came up with an idea to stream music from my mobile phone (LG GD880) via bluetooth to my SUSE box (which has it's audio output linked to my HIFI Sound system). This means pairing the devices and of course set the SUSE box as reciever. I found how to accomplish this here and here .
The problem: While trying to complete the above tutorial(s), I found that in /etc/bluetooth there is no ' audio.conf ' file, only ' main.conf '. If I insert the option: Enable=Source in main.conf, it has no effect what so ever. This means that org.bluez.AudioSource will not show up in D-Feet, thus I cannot take the next steps. What can I do? Is there another way to accomplish the audio streaming?
It's installed as a mobile broadband connection. Only works if my phone is plugged in during boot, otherwise plugging in my phone does nothing. For example, I booted my netbook earlier today but my phone wasn't plugged in. Tethering did not work, it just acts as if it isn't present. I rebooted (with my phone still attached) and now tethering magically works.
View 5 Replies View RelatedI have a mobile phone (T68i) attached to my server and everything works fine. But sometimes the phone craches and needs to be rebooted wich is quite annoying.So I want to use a newer phone that is not 10+ years old :-)But when using a K610i or C702 the phone does not appear on /dev/ttyUSB0 as expected.When inserting the usb cable a dmesg gives this:
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[5924924.451033] usb 1-3: new high speed USB device using ehci_hcd and address 19
[5924924.576728] usb 1-3: New USB device found, idVendor=0fce, idProduct=d0d9
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I have a dell precision m4300 laptop with a 360 wireless bluetooth dell adapter On my system there is a debian lenny with kde3 and backported enable(everythings is p to date except bluez-utils and bluetooth holded at version 3.36-3)
Nowadays bluetooth more less works fine, I can send and receive single file to/from my phone (nokia n70)
The hell begin when I try to browse my phone files from konqueror...with bluetooth:/ I can see the list of all the device near me with sdp://[address]/ i can see two icons (obex file transfer & obex object push)
but I cant see or access to any file or folder into the phone.
I also try to update my bluez-utils to 4.60-1~bpo50+1 but in this case kbluetooth totally fail and a see the contextual menu of the system tray icon all disaled.
I'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.
[URL]
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 Relatedget 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 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.
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.
Can anyone recommend a simple open source voip software in linux for pc-to-pc voice calls.
One requirement i that it should be simple with bare minimal functionality since I will be using as a part of a university project and so I don't want any additional features such as buddylist, GUI, etc.
Even a command line interface will also be fine since I am only interested to test different associated codecs/network characteristics.
The code should be simple with bare minimal libraries so that can be compiled easily.
I am aware of some open-source software such as WengoPhone but it seems that they have lot of associated features supporting different input devices with sophisticated GUI.
I require only PC-to-PC call functionality and I will be using within a LAN.
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..
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.
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Is 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?
The Opportonity to share Infos about upgrading and installing Stuff on Ubuntu Ok so i wasn't sure where to put this!!As i was a Windows User i used to call Landlines for free using Softwares such as :[URL].. Now i dont know How to use these Softwares in Ubuntu , I tried to Configure it through Wine Microsoft Windows Compatibility Layer . In case of Using Ekiga , I really don't know How to use it (as a Matter of Fact I tried to use it but i assume using it require an SIP-Account )
View 6 Replies View RelatedI have a Compaq Presario V2000 running Ubuntu 10.04. I use voip to call international mobile numbers. I'm currently using a Netgear router (at a friend's place) and now, my calls cut in 25 seconds. The call time keeps running, but neither can hear each other after 25 seconds. I've tried using service providers like Actionvoip and Jumblo (as they are cheaper than Skype) and clients like Ekiga and Twinkle. Same It happens on both Wi-Fi and ethernet. There is enough credit I tried using my friends Dell Latitude running Windows 7 with my Jumblo account. Surprisingly, it worked perfect!Previously I used a D-Link router and it worked flawlessly with my system.I fail to understand if it is some settings issue or a bug with Netgear's firmware. If anyone else has come across this issue and has been successful in getting around it
View 1 Replies View RelatedI am trying to setup a mumble server on Ubuntu Server 10.10. I have downloaded/installed all of the necessary files and configured my router to open up the necessary port (64738 ). However users outside of my network are unable to connect to the server (get Timeout error). Also to add, I am fairly new to using linux/ubuntu.I made a post on reddit where you will find more information regarding my problem: [URL]
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