Displaying 20 results from an estimated 79 matches for "kg6omk".
2004 Jan 24
2
Subject: Re: Grandstream 100 sidetone
Chris Albertson wrote:
|What firmware version do you have?
program version 1.0.4.39
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
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2003 Oct 05
2
Good W2K softphone
...phones that run under
>Windows 2000 that I like or that even work well. What are other people
>using?
>
>=====
>Chris Albertson
> Home: 310-376-1029 chrisalbertson90278@yahoo.com
> Cell: 310-990-7550
> Office: 310-336-5189 Christopher.J.Albertson@aero.org
> KG6OMK
>
>__________________________________
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>http://shopping.yahoo.com
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com...
2003 Oct 08
2
SIP softphone volume control?
>I went back to the example system direct from CVS with small
>additions to sip.conf and extnsion.conf needed to make one
>xten X-Lite phone work. I can dail in and hear the anouncements,
>call the demo server at Digium. The audio quality I hear
>comming from Asterisk back to X-Lite is good (9 on a 10 scale)
>but the sound volume comming from the X-Lite extension is very low
2003 Oct 08
1
Mini-PC box to run server
...ky noisey
>
> I assume any PC sold today has enought CPU power to run a
> small asterisk system
>
> =====
> Chris Albertson
> Home: 310-376-1029 chrisalbertson90278@yahoo.com
> Cell: 310-990-7550
> Office: 310-336-5189 Christopher.J.Albertson@aero.org
> KG6OMK
>
> __________________________________
> Do you Yahoo!?
> The New Yahoo! Shopping - with improved product search
> http://shopping.yahoo.com
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.d...
2004 Nov 30
2
Dual NAT for SIP
Hi,
My installation at home use two NAT translations before it reaches the linux box where Asterisk is running on.
I use DSL with a Wireless router which fwd all packets to an Windows 2003 box an this windows box it NATing the UDP and RTC packets to my linux box.
If I try to connect to it from outside I get this error :
Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum
2003 Apr 19
0
RE: [Asterisk] How to select server ardware?
...hat would be required 500Mhz Caleron? 2.4Ghz P4???
> >
> >
> >
> > =====
> > Chris Albertson
> > Home: 310-376-1029 chrisalbertson90278@yahoo.com
> > Cell: 310-990-7550
> > Office: 310-336-5189 Christopher.J.Albertson@aero.org
> > KG6OMK
> >
> > __________________________________________________
> > Do you Yahoo!?
> > Yahoo! Shopping - Send Flowers for Valentine's Day
> > http://shopping.yahoo.com
> >
> >
> >
> >
>
>
> =====
> Chris Albertson
> Home: 310-3...
2004 Jan 14
4
re hardware requirement - asterisk
I have just checked the Openbsd box on the if interface.
fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500
address: 00:02:55:30:54:28
media: Ethernet autoselect (100baseTX full-duplex)
status: active
inet 192.168.1.1 netmask 0xffffff00 broadcast 192.168.1.255
inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1
xl0:
2003 Nov 14
7
Your thoughts..
I need to get your thoughts on something.. :)
I am trying to create a system to process the CDR call logs for
department accounting..
I think there are two ways of doing it.. Either I can create an AGI that
will run on the "h" extension and will lookup the last entry that
matches the account code of the call that just ended in the MySQL CDR
and calculate the call cost immediately..
2003 Jul 15
9
Poll - Would you pay $30-$50 for high quality speech synthesis?
Many of you are familiar with how lousy Festival sounds.
AT&T has a product, NaturalVoices, that sounds much better. There are
male & female voice fonts for US/UK/Indian English, French, Spanish,
and German.
I am considering offering a linux-based text-to-speech engine based on
the NaturalVoices runtime. An asterisk module would also be provided,
making it easy to add natural sounding
2013 Sep 27
0
No subject
...s the most cost effective legally
>abtained Ethernet<=>FXS box that will work with Asterisk?
>
>
>=====
>Chris Albertson
> Home: 310-376-1029 chrisalbertson90278 at yahoo.com
> Cell: 310-990-7550
> Office: 310-336-5189 Christopher.J.Albertson at aero.org
> KG6OMK
>
>__________________________________________________
>Do you Yahoo!?
>Yahoo! Tax Center - forms, calculators, tips, more
>http://taxes.yahoo.com/
>_______________________________________________
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>Asterisk-Users at lists.digium.com
>http://lis...
2003 Apr 04
0
non-telephony use of T400P?
...erface
> > card to do any protocol, just pass the bits.
> >
> >
> > =====
> > Chris Albertson
> > Home: 310-376-1029 chrisalbertson90278@yahoo.com
> > Cell: 310-990-7550
> > Office: 310-336-5189 Christopher.J.Albertson@aero.org
> > KG6OMK
> >
> > __________________________________________________
> > Do you Yahoo!?
> > Yahoo! Tax Center - File online, calculators, forms, and more
> > http://tax.yahoo.com
> > _______________________________________________
> > Asterisk-Users mailing list
>...
2003 Sep 09
3
Asterisk Security vulnerability report
Hello,
today I found this security report regarding Asterisk SIP Security.
http://www.securiteam.com/securitynews/5LP0720B5G.html
Maybe It could help somebody who isn't using a newer than 15th of August
cvs version.
Best regards
Lubo
2003 Oct 03
1
Editting variable contents
ChanIsAvail returns the channel ID plus "-<session>".
How can I edit ${AVAILCHAN} to remove this session ID, so I can use its
contents in a subsequent Dial statement?
Dialing on Zap just gives a warning, but dialing a SIP channel
completely errors out.
------ extensions.conf snippet-------------
;
; Main Home number (8901)
;
; Bedroom1
exten =>
2003 Oct 12
2
New Processor support..
Hey..
Has anyone played around with Asterisk on the Itanium2, Opteron or
Athlon64??
Can Asterisk (or Linux for that matter) actually make good use of a
64bit system??
Later..
2003 Oct 14
2
Digium should develop and sell just Dummy card. For timing...
I'm first to buy 5 pack. Even for > $30.
2003 Oct 22
0
Fwd: Download Asterisk
...________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
=====
Chris Albertson
Home: 310-376-1029 chrisalbertson90278@yahoo.com
Cell: 310-990-7550
Office: 310-336-5189 Christopher.J.Albertson@aero.org
KG6OMK
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2003 Oct 27
0
Fwd: Re: Asterisk on FreeBSD
...ofesion quality and very mature.
I'm still betting the root cause is the differece between the Linux
and BSD pthreads implementation.
=====
Chris Albertson
Home: 310-376-1029 chrisalbertson90278@yahoo.com
Cell: 310-990-7550
Office: 310-336-5189 Christopher.J.Albertson@aero.org
KG6OMK
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2003 Oct 30
2
Fwd: Re: SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
...ndependent
"fixes" that both do what has been suggested in this thread.
No need for a third except posibly a merge of the two.
=====
Chris Albertson
Home: 310-376-1029 chrisalbertson90278@yahoo.com
Cell: 310-990-7550
Office: 310-336-5189 Christopher.J.Albertson@aero.org
KG6OMK
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2003 Oct 30
2
Asterisk + Video
Is anyone using Asterisk as the gatekeeper/proxy for videophone calls?
Thanks,
--Ernest
2003 Nov 06
1
Need testers for new STUN build system
...#39;d like bug reports, success reports or patches if you can do that.
If you'd like a tar file e-mailed to you let me know OFF LIST.
=====
Chris Albertson
Home: 310-376-1029 chrisalbertson90278@yahoo.com
Cell: 310-990-7550
Office: 310-336-5189 Christopher.J.Albertson@aero.org
KG6OMK
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