similar to: open source sip softphone (Window OS version )

Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_ W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewIte m Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid =27 But looking for real world feedback. Cheers,

Hello, Has anyone Knowledge about softphone IAX for pocket PC totally free? Tkanks for all. -- Sandra Salmer?n Ntutumu <makevuy@ehas.org> Tlf. Analog: +34 914888405 / M?vil: 653574298 Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundaci?n EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de pa?ses en desarrollo

Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

Hi, I have two softphones connected to an Asterisk "stable". I have two extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on extension 2000 will ring, but as soon as the call is picked up, extension 2000 will hang up the call. The softphone on 1000 (SIP, SJphone, e.g.) will give a "403

Has anyone got any information on bulk provisioning of Linksys SPA-941/94s? There is an overview in the admin guide but it refers to a different provisioning guide that I haven't found anywhere. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com

I am ready to pull my hair out. I cannot seem to get the Polycoms to read the time properly. Regardless of the server they are pointed to our the offset, i am getting the correct time, but 24 hours ahead. So for today it is showing Friday April 28 but with the correct time. Any clues? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider

Is there any setting in the voicemail that will send the voicemail file in a type that is recognized on a Blackberry? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com --------------

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com

On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's

We are trying to setup a sip connection behind a Watchguard Firebox 1000 and it is simply put...not working. The ports are all forwarded but the packets are not going out. It is as if the firewall simply ignores SIP packets. Has anyone seen this or have any idea what the issue could be? Watchguard so far has been of zero help. Kerry Garrison Director of Technical Services Tech Data Pros -

Has anyone had any experience connecting Asterisk to Cbeyond's SIPconnect service (http://www.sipconnect.info)? Any opinions? Thanks, -Brian

Hi, What can it be when I can establish a connection between two Softphones but no voice is transfered ? thnx Hugo,

Hey guys, I have just developed this problem with my Windows XP box. I think it started since I installed XP SP2. Both SJPhone and Xlite does some kind of bridging with the speaker out port. When ever I make a sip call to where ever, the other party hears a lot of echoing. Well I noticed just now when I was playing mp3's via Winamp, the music was being played through my sip calls that I

Guys.. which free softphone is the best,grandest,most recommended one out there? based on your own experiences..

i have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com>

As Promised, I've released a new version of Firefly (ver 1.8) with IAX & SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user -> software -> firefly), then delete tree from your registry. If that fixes it, send

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten =>