Niecly said.. hoeever, these list are not for astrix users, butt for bashing, didnot you realise this ?
It had where 4 years more , know that this is fluent in this site. Translated as in : this list is a bash fest since i can remember back in 2004, everyone is right, no one is wrong, everyone is a god, and so on. However you made a point that will get tossed back in the "pit of endless replies" however good a point it was. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno Sent: May-18-09 7:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Open source SIP client It seems like a few people including me DID understand what Dhaval meant, or maybe some people used they common sense and their intelligence to understand what somebody who's english is not the primary language wanted to say and put some effort to guide or help someone in the community getting to the right direction instead of trying to put him down. I think a few others need to consider investigating more deeply the basic mechanics of understanding written English, or should themselves research what some collections of syllables intend to convey. I also think if they were that good, why not provided some english tutoring instead of putting people down. Good luck in you research Dhaval! On Mon, May 18, 2009 at 9:46 AM, Scott Gifford <sgiff...@suspectclass.com> wrote: DHAVAL INDRODIYA <dhaval.it01...@gmail.com> writes: > can anybody help me to give Opensource SIP client information which > can be modified as per our requirment Hello Dhaval, We have tried several open-source SIP phones on Linux. We have had the best luck with Twinkle Phone: http://www.xs4all.nl/~mfnboer/twinkle/index.html It has lots of hooks where you can stick your own scripts to modify its behavior. We also had pretty good luck with SFLphone: http://www.sflphone.org/ There is a list of open source clients on voip-info that includes these two. It might be a good starting point: http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software Good luck! ----Scott. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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