Hi all http://www.asterisk.org/ http://www.voip-info.org/wiki/view/Asterisk/
I think the above sites can help you as they say's Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI <http://www.voip-info.org/wiki/view/ADSI>, IAX <http://www.voip-info.org/wiki/view/IAX>, SIP<http://www.voip-info.org/wiki/view/SIP>, H.323 <http://www.voip-info.org/wiki/view/H.323> (as both client and gateway), MGCP <http://www.voip-info.org/wiki/view/MGCP> (call manager only) and SCCP <http://www.voip-info.org/wiki/view/SCCP>/Skinny. Regards Santosh On 8/9/07, Donald Lee <[EMAIL PROTECTED]> wrote: > > Hi amit: > > I think the following sites can help you: > > http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-13.txt > http://www.ietf.org/rfc/rfc4579.txt > > > 2007/8/9, amit <[EMAIL PROTECTED]>: > > > > Hi All, > > > > > > How we can done conference in SIP ? > > > > We was already see in Tech-invite site but it not > > > > helpfully for us........ > > > > Please tell what we do for conference...... > > > > > > Thanks in advance, > > > > Amit Vijayvargiya > > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, please visit > > http://sipforum.org/mailman/listinfo/discussion > > Post to the list at [EMAIL PROTECTED] > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at [EMAIL PROTECTED] > >
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