I had a similar issue a while ago. Check your dial plan. Are you
forwarding to your cell phone's V-Mail as fallback? I had the issue
where I was getting callbacks from asterisk if one phone was on DnD and
the calll wasn't answered. Becarefull of your dial() commands and the
delays you use.
Zeeshan,
First off, if your fear of being sued is what stops you from doing
business then get out of the industry or get over it. Its a risk we all
take everyday (not just in VoIP). You build up a core of Insurance and
Defensive Patents to protect yourself. Risk is just part of doing
Steve Totaro wrote:
What if a train derails and slices through the main fiber connections.
OK, so you have XO, Global Crossing, Verizon, and UCN all for
redundancy. Well guess what? They are all most likely running over
those strands of fiber. You better have a VSAT connection too!
Stephen Bosch wrote:
Of course not -- but how many hundreds of millions have been invested in
their infrastructure?
You missed the point. The standard formula I use is 5 days out or
more precisely 2% of gross revenues each year. For google its still a
kings ransom, but for a small
Steve Totaro wrote:
Setup those cell phones to use chan_mobile and you have a very nice
solution. Unless the phones are assigned to people who use them as
their own. You could possibly add some lines on a family plan $10/mo
extra w/T-Mobile and use those strictly as PSTN fialover lines.
Hello all,
I'm looking for software for my asterisk logs that will compile the
information into nice web-based charts and graphs. Something that works
similar to webalizer for apache. I want to be able to spot trends of
usage, call volume levels, disconnect/failure levels, and basically
Per Jessen wrote:
Radio-amateurs have done phone-patching for decades (where allowed) -
there must be someone who can point you in the direction of an easy
solution.
/Per Jessen, Zürich
The BIG problem here is that most Radio Amateur software and hardware
operate in a half-duplex
Eric ManxPower Wieling wrote:
Michael Collins wro
Except that for some users 1.2.18 is NOT stable. I've had to roll
back to 1.2.15 on my production servers in order to prevent core dumps
at least once per day. No, I am not willing to turn my production
servers into testing servers to solve
Matthew J. Roth wrote:
In fact, it seems that somewhere between 200 and 300 calls, the two
servers start to exhibit similar idle times despite one of them having
twice as many cores.
Sounds like you are running into the hardware limitations of your
systems PCI or Front Side Bus (FSB)
Tzafrir Cohen wrote:
What I say is that you have the worse of both worlds:
- downtime of at ~1/2 a minute (avarage, if a cron runs every minute).
In the case a restart is all it take.
- A bigger downtime in case a restart is not what it takes. Because your
logs will be flooded.
- And a
http://www.psh-inc.com
Tzafrir Cohen wrote:
On Fri, May 04, 2007 at 01:59:41PM -1000, Mark Coccimiglio wrote:
What I do is add an entry in the crontab file as such:
* * * * * if [ ! `/bin/pidof -s asterisk` ] ; then /usr/sbin/asterisk; fi
Its simple and it works. Additionally if asterisk
What I do is add an entry in the crontab file as such:
* * * * * if [ ! `/bin/pidof -s asterisk` ] ; then /usr/sbin/asterisk; fi
Its simple and it works. Additionally if asterisk crashes then cron
restarts the server in about a minute. Just be careful with your configs.
Mark Coccimiglio
Just run down to your local Radio Shack...and KISS.
http://www.radioshack.com/product/index.jsp?productId=2062696
Mark C.
Klaverstyn, David C wrote:
This is what I want. Do you have any URLs to such a device as I cannot
find any.
-Original Message-
From: [EMAIL PROTECTED]
These are the patent numbers in the lawsuit (Thanks Pat and Sal)
6,137,869
6,430,275
6,104,711
6,282,574
6,359,880
6,128,304
6,298,062
Mark C.
Yuan LIU wrote:
From: Kenneth Padgett [EMAIL PROTECTED]
Date: Mon, 9 Apr 2007 23:49:31 -0400
[good stuff sniffed]
I'm not doubting that
Ok here is a real geek question,
I building my own linux kernel for my asterisk system and came across
the kernel setting for the timer frequency. I have one of 3 hardcode
choices 100Hz, 250 Hz and 1000Hz. From what I understand the default
Freq was changed from 100Hz in kernel 2.4 to
Ok this is a simple question...
What has been your experience with the WellTech 38xx series (I'm looking
specifically at the 3802)
VoIP gateway? I'm looking for a good (and hopefully not too expensive)
VoIP/T.38 gateway for my office.
Asterisk intergration is not a major factor at this time
My experience has been to be consistant. The only time I have had
problems with DTMF is when I am not using the same DTMF encoding
technique on all hardware. Your choices are: INFO, RFC2833 or
INBAND. Some equipment also has an AUTO option but I would not
recomend it. Stick with INFO or
M.Hockings wrote:
I don't really know the name of what I want to look for but maybe
someone could tell me if it would be available.
I have a number of old analogue cell phones laying about here and I
was thinking it would be useful if I could set up a short range base
station for them
Catalyst 3500
series for something like that. Be carefull and look closely some
systems only support 2 ports on 1000baseT and the rest are 100BaseT.
Good luck and happy hunting,
Mark Coccimiglio
___
--Bandwidth and Colocation provided
shadowym wrote:
Regardless of the 1600's spec's which are outdated in many ways by todays
standards, ANY cheap 1600 or PIX etc. you buy on Ebay will likely have MANY
MANY hours on it. Sure, they are built to last but they do not last
forever. I would consider ANY of these boxes as somewhat
Jon Pounder wrote:
you should take your own advice - an acre is 200ft x 200ft - what
idiot would
pay a consultant $7000 to tell them they need one access point in the
middle.
I have a BA in Electronic Engineering, a Masters in Computer Science and
I'm an FCC licensed
radio operator.
. Not even a blink. Finally,
untill everyone is using 10Mps FTTH the broad band link is still the
slowest part of the connection. Not to shabby for antiquated technology.
Mark C
Martin Joseph wrote:
On 2007-01-06 00:48:11 -0800, Mark Coccimiglio [EMAIL PROTECTED] said:
Mike
I'm using
Mike
I'm using a Cisco 1605R [running IOS 12.3(5a)] small office router with
Fair-Weight queueing enabled. Works great. The nice thing about
Fair-Weight queueing is that it dynamically adapts to lower the priority
of higher demand traffic (e.g. large downloads). If you want quality
stick
Try setting in sip.conf:
nat=route
This tells asterisk to send all responses back to where the inquiry came
from rather then from the info contained in the sip packet.
Good luck,
Mark Coccimiglio
IS Director
Payroll Services Hawaii, Inc.
http://www.psh-inc.com
Elpidio Ramos wrote
I was wonder if anyone is rumming this combination of hardware:
Colomachine.com: CM62
Digium Card: TE405P
I need a rackmount to send to a data center and this combination fits my
budget. Has anyone else used colomachine with asterisk? how has it
performed? I plan to run the latest
Do you have STUN Enabled? I had similar when I had STUN turned on. I
found it better to turn off stun and place in sip.conf nat=route.
Also use NAT Keep-Alive on the ATA that is NAT Timeout on the Router.
Good Luck,
Mark Coccimiglio
IS Director
Payroll Services Hawaii, Inc.
http
: Mark Coccimiglio [EMAIL PROTECTED]
Sent: Mon, December 18, 2006 2:27 pm
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] GXP2000, Linksys RV082 Firewall /
NAT,Registrations
Do you have STUN Enabled? I
have the SPA3000 on UDP 5070 and RTP
16399-16401. I don't use STUN (tends to cause more problems then it
solves).
On the Server side I have the NAT firewall/gateway forwarding UDP port
5060 and RTP 16393-16401 to *.
In sip.conf set nat=route for each NAT client.
Hope this Helps,
Mark
I bought one over a year ago along with the USB phone. Was never able
to get the card to work properly with anything (even the software it
came with). For less money I got am X100P clone on ebay and that works
great.
Leo Ann Boon wrote:
Anyone has any experience with these cards? Looks
From what I understand the B410P us intended for use OUTSIDE North
America. I contacted them a little over a month ago looking for a USA
compatable card and was told that there isn't sufficient market for the
hardware. Oh well, so if you are in Europe or most other places you
will have a
Ok here is one for you.
I know we all do the this for 911:
exten = _911,1,Dial(Zap/1/911)
exten = _9911,1,Dial(Zap/1/911)
And this probably is more then acceptable for most of us. However I
have a system setup that uses SIP for most calls and 1 POTS line. We
use a least cost routing that
Hey all here's an update.
I do care to thank everyone for your information on BRI interfaces
that operate in USA/NA. I know the responses are were limited, but the
selection of hardware is also limited. (Shame because BRI would fit my
needs perfectly). To continue, it's now been over 4
Sounds like Propaganda to me.
Dean Collins wrote:
While not
strictly on topic I think this could be an
interesting opportunity for the Asterisk development community.
As some of
you might already know JavaOne will be
happening in San Francisco
in 2 weeks time
Ok,
I have to agree here. IF my simple fax server log/tiff archive is
not enough to satisfy a client that the fax is genuine I would not want
them as my customer. I don't care how much money they spend. Business
is business and what I do is what I do. There has to be at least a
little bit of
I have 2 of the Sipura SPA-841 (now Linksys/Cisco) and they work great
once I got the latest firmware. They cost ~$100 each.
Tomislav Parčina wrote:
Has anybody used Huawei EP201S IP phone? I need 9 SIP phones that cost around
100USD, and those phones are one of options.
Can anybody
on my extra DiD's? Any one
what a phone number in Hawaii? :) Its such a shame I can't leave well
enough alone and suck it up on POTS (eck). I'll keep you informed as to
my progress (or lack there of).
Mark Coccimiglio
n3whx @amsat.org
sip:[EMAIL PROTECTED]
Walt Reed wrote:
I'm in a similar
such as my own. VoIP may be an option, but I would need a ITSP
that would allow calls to transfer from my asterisk box to the remote
phone set. My link to the internet is fast, but its pointless to route
a call into the office just to stream it back out. More work more work
more work.
Mark
.
Mark Coccimiglio
[EMAIL PROTECTED]
sip:[EMAIL PROTECTED]
Lacy Moore - Aspendora wrote:
Mark,
I was in the same situation. Our current system uses BRI for
almost all lines. I looked for some kind of solution and finally gave
up. The BRI products here just seemed way too expensive for me. I
, Broadvoice, etc...) but before I commit
either way I'm exploring all my options.
Your opnion matter here to please let me know.
Mark Coccimiglio
[EMAIL PROTECTED]
sip:[EMAIL PROTECTED]
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