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pls kindly respond to this email thx ! _________________________________________________________________ Connect and share in new ways with Windows Live. http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008

speech rec what works ? anything out there with established dictionary, eg medical ? don't want to pay $3-4K for Nuance API thx in advance _________________________________________________________________ Is your PC infected? Get a FREE online computer virus scan from McAfee® Security. http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963

i have a customer that needs to plug the phones into the pc's using the pass-through rj45 available on most sip phones the question they are asking me is how to keep the data network separate from / secure from the voip network i understand they can set up vlans but i am hazy on a few details 1 since the phones are plugged into the pc's how will the phones be segmented into their own

make[2]: *** [obj_linux_x86_r/simph323] Error 1 make[2]: Leaving directory `/usr/src/openh323/samples/simple' make[1]: *** [opt] Error 2 make[1]: Leaving directory `/usr/src/openh323' make: *** [optshared] Error 2 ???? any idea ? Thx in Advance _________________________________________________________________ Is your PC infected? Get a FREE online computer virus scan from McAfee®

ver 1.9.1 -I. -shared sound_oss.cxx -o ../pwlib/device/sound/oss_pwplugin.so sound_oss.cxx: In member function ‘virtual BOOL PSoundChannelOSS::Read(void*, PINDEX)’: sound_oss.cxx:766: error: cast from ‘void*’ to ‘unsigned int’ loses precision make[3]: *** [../pwlib/device/sound/oss_pwplugin.so] Error 1 make[3]: Leaving directory `/usr/src/pwlib_v1_9_1/plugins/sound_oss' make[2]: *** [opt]

ver 1.17.2 [root@71 openh323_v1_17_2]# make opt /usr/src/openh323_v1_17_2/openh323u.mak:192: usr/src/pwlib_v1_9_1/make/ptlib.mak: No such file or directory make: *** No rule to make target `usr/src/pwlib_v1_9_1/make/ptlib.mak'. Stop. [root@71 openh323_v1_17_2]# thx in advance _________________________________________________________________ Express yourself instantly with MSN Messenger!

we are experiencing 30 second delay before voice is heard after answer when we ran wireshark it showed the problem between frames 634 (where the softphone answers) and 1366. Between those frames, asterisk receives RTP packets from both the softphone and the sip carrier, but doesn't forward them to each other.... Then we see a bunch of packets sent at once from the asterisk server, until

apologies for cross posting to the biz list and the user list but the biz list has been slow of late the big players...cisco and avaya and alcatel are all tendering you must be based in the new york area; if you are based outside new york or overseas and you have a new york branch office, it will not work; also you must have a reference of an install of at least 1000 extensions i received this