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?
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.
I'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?
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 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
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.
Using 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
I 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.
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?
Is there an app out there that will at least mute my speakers or pause my music when I receive a phone call? This would come in handy since I listen to music a bit loud and have a phone that vibrates like a kitten.
Maybe I'm not using the right search terms, but I can't find anything on this. I have a setup of Slackware with Asterisk and FreePBX. I am recording calls on demand and can get to the recordings from the Call Monitor (web interface). I would like to be able to access the recording from the phone (Aastra 57i), much like a voice mail is accessed.
In practice I have a script that call a java program that call a linux system command. The script if I run it, from a shell functions well,so it is not a java problem. The problem come out when i put this script in a crontab schedulation. The result in this case is that java do not execute the system command. I think it depends on crontab
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.
I 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:
Code: [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
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.
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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.
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
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.
I 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.
I'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 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?