not sure if this is a biz question or tech question
its a tech question insofar as ms lync desktop client has integration with
email and screen sharing and wondering how to do that in the asterisk world
perhaps with a softfone like zoiper or an IM client like pidgin.when i
search for
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
if you are not local, pls do not reply
local is an area defined by syracuse, rochester, ithaca, and binghamton new york
its an ongoing consulting gig
thx
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need asterisk tech to relocate to riyadh
this is a permanent position
arabic speaking not necessaryurdu and tagalog ok
ENGLISH required
shukran, shukria, salamat
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pls kindly respond to this email address
thx !
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pls kindly respond to this email
thx !
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-- Bandwidth and
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
PROBLEM...NO DTMF ON OUTBOUND CALLS
1
ASTERISK FORWARDS THE DIGITS
Got rfc2833 RTP packet from 66.108.217.191:2256 (type 101, seq 279, ts
-1975142833, len 4, mark 0, event 0009, end 1, duration 1600)
Sent RTP packet to 209.212.88.161:32872 (type 101, seq 63407, ts 60776, len
4)
Sent
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
possible need for onsite tech for Wilmington NC and Boston MA
if you reply pls make sure asterisk does not appear in subject line so it
does not get filtered
to list folder
thanks in advance
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Dont just search. Find. Check out
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
_
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Security.
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]: ***
ver 1.17.2
[EMAIL PROTECTED] 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.
[EMAIL PROTECTED] openh323_v1_17_2]#
thx in
pwlib ver 1.5.2
/usr/bin/ld:
./obj_linux_x86_d/asn_grammar.o(.gnu.linkonce.r._ZTV5PListI7PStringE[vtable
for PListPString]+0x1c): unresolvable relocation against symbol
`PAbstractList::Compare(PObject const) const'
/usr/bin/ld: final link failed: Nonrepresentable section on output
collect2:
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
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
I have customer wtih 30 stations in cubicles but they only
have 1 rj45 per cubicle and that is for lan and internet.
I would prefer the voip to be on separate net connection for quality
purposes
but customer does not want to recable. How to avoid voice quality problems
?
I have read about
2 SIP phones on Y data connector on 1 ethernet -
will that cause problems ?
thx in advance
_
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