Why is it you have to put down the United States? Sprint is a U.S.
Company. Vonage a U.S. Company. Digium a U.S. Company. What in the
heck does it have to do with you? Even if Vonage is tied up in court
the rest of the world doesn't care. VOIP will live on. We are not sue
happy, This is big
Good explanation Rich. Unix was built for the riggers of the Telecomm
industry. You won't find Windows running the PSTN. Unix and Linux are
used where their needed for real time processing and the highest
reliably. Windows is a productively OS that is easy to use for non
technical people. I
Sherwood, I have never known the RTP audio to be on only one port in
sip. I believe it's always on 2. The one way audio is always a
nat/firewall problem in sip.
Sherwood McGowan wrote:
Guys,
I've been poking around trying to find a good answer for this via
voip-info, google, etc...
This can be done by modifying the source code.
trixter http://www.0xdecafbad.com wrote:
I have been asked if asterisk can change codecs dynamically based on the
calling party's caller id. I couldnt find anything, and dont know that
this is something that asterisk can do, but it occurs to
IL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Michael
D Schelin
Sent: Saturday, September 10, 2005 10:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] civil emergency comms: Asterisk
+ HAM
The two best forms of communications in a real disast
The two best forms of communications in a real disaster and one always
has been is #1 Ham radio. and #2 satellite telephone. Ham radio is
global and has proven time and time again to be the most reliable when
the infrastructer has been damaged. The U.S government is the biggest
user of
Ben, That is the correct choice for an Asterisk box. good luck.
Ben Brown wrote:
Thanks for the replys. I'm
convinced. PRI it is.
Peter Svensson wrote:
On Mon, 5 Sep 2005, Ben Brown wrote:
So the only difference with PRI is caller ID? What I am trying to
Go T1 with PRI signaling. Farming and line coding is for all T1's. We
use ESF (extended super frame) B8ZS ( I forget B8ZS stands for but it's
a newer line coding) . If you have it avaible to you, Signaling type
should be PRI. The rest of your numbers 4-7 are in the PRI signaling.
No sound
.
Is a non-PRI T1 significantly harder to configure with Asterisk?
Can Asterisk choose the context based upon the CallerID with a PRI?
Thanks for your reply
BEN
Michael D Schelin wrote:
Go T1 with PRI signaling. Farming and line coding is for all T1's.
We use ESF (extended super frame) B8ZS ( I
The Asterisk Software is not the problem. I'm thinking and I could be
wrong that your having a total line balance mismatch with the card your
using. Check the line impedance and the card's. Most people using
Asterisk don't have that much echo. Anyway It would be nice to see a
manual Hybrid
I don't think this will work but it's worth a try.
Fall VON 2005 http://von.com is happening September 19-22, at the BCEC in Boston.
As usual, we have a special offer for members of the pulvermedia community, which is valid for the month of June only. Register using priority code JUNE and
Call Digum. They support the license codec install.
Matthew Schumacher wrote:
List,
I purchased 2 g729 licenses but I can't get it to answer a g729 call
from a cisco router with a vwic card. In the debug output below you
will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263)
PRI is one of many signaling formats available on a T1 circuit. T1 and
PRI are not the same thing. Have your carrier change the signaling
format of your T1 to PRI. PRI is 23 B Channels and 1 D Channel
(Signaling) and is an end office protocol. It has many of the features
of a full blown SS7
The Multitech mvp24xx and the 130 are true T38 devices and work well.
Cory Andrews wrote:
We use the MultiTech FaxFinder 100 and 110 (1 and 2 port fax,
respectively). We have it integrated with an Asterisk server, faxes
are routed to the FaxFinder, converted to PDF files and sent to an
I'm not hiding anything from this user group. Buy a Cisco gateway and
put in you own T38 network together. I can't respond that fast to the
hundreds of emails I've receive. All I'm saying is if you want T38 now
then buy our service. If not, then wait for the Asterisk community to
release it.
Why do you put me down? I have not done a thing to you and I'm not a
spammer. Please stop this activity It's not professional. If I were to
give you bad service please feel free to comment negatively but I've
never dealt with you nor do you have an account with us.
Sincerely
Michael D
Rich is correct. Example: Night security guards may need to catch an
inbound calls that could ring at more than one station. Maybe one is
doing rounds and the other is at another desk off site. Sometimes call
forwarding is too slow. There are many reasons why this could be used.
Rich Adamson
Hello everybody, for all of you that have searched for a real fax
solution, look no further. We now have T38 faxing. Please contact me for
more information.
Thanks
Michael D. Schelin
ShellTel
626-814-2354
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Asterisk
, then i would be terribly afraid of his companies
technical ability for their actual VOIP service..
scary..a VOIP provider that can't even provide themselves...
On 7/19/05, Jason Stewart [EMAIL PROTECTED] wrote:
On 18/07/05 17:06 -0700, Michael D Schelin wrote:
I
for their actual VOIP service..
scary..a VOIP provider that can't even provide themselves...
On 7/19/05, Jason Stewart [EMAIL PROTECTED] wrote:
On 18/07/05 17:06 -0700, Michael D Schelin wrote:
I was waiting for everyone to reply so here is mine.. Check out
herd of nothing but good
things about your customer support. I've called and left messages to
your support team. I waited 7 days for this unit and have no way to
configure it. Email me the CD.
Michael D. Schelin
Owner
Shelltel
___
Asterisk-Users
wanted to get feedback ether way, or maybe
a contact name so I can get this paper weight working and tested. Has
anyone used the 2102? Please let me know.
Michael D. Schelin
Shelltel
JD Austin wrote:
Michael D Schelin wrote:
Here
is
a letter I sent them for my $150 paper weight
I agree with you but not 100% with them. An IP to Ip call on an ATA flat
out is better . Now don't even get me started about cellular. My Service
dosen't drop calls in the middle of conversations. VoIP is a notch
better than Cellular.
Michael Graves wrote:
Here's t
link:
Proxy servers can do that.
Brian Capouch wrote:
A group which my school is part of wants to start using DNS SRV
records to allow email-style dialing amongst members of the group.
I have gotten the records in our zonefiles, and things work pretty
much just fine.
However, since the DNS
search for [EMAIL PROTECTED] It works well and is very easy to install for
beginners like me.
Michael Felder wrote:
Can anybody recommend an Asterisk GUI to help a newbie confg ?
Kind regards
Michael Felder
IT Medic Australia Pty. Ltd.
P: 03 9557 2213
F: 03 9557 2214
M: 0419 568 217
E:
G279 on Asterisk works great.
Jean-Louis curty wrote:
thanks I 'll try ... :-)
jl
2005/7/4, Jean-Louis curty [EMAIL PROTECTED]:
Hi,
I'd like to understand what should i do to use G729 codec in a legal way,
how do I order licences ? to whom ? how do I install them on asterisk etc ?
I thought everyone should know this.
Jorge, After reading your page in the
http://voip-info.org/tiki-index.php?page=Asterisk+SS7
please advise Your U.S. customers that SS7 is not done the same way as
in the rest of the world and the requirements are different. The U.S
carrier's require 2
Call Mike at ShellTel 626-276-9009
List Receiver wrote:
Having used Broadvoice for a while with marginal service, I want to move
on to another provider. So my question to the List is who is good? I
know now one service is perfect but somebody out there has to be decent.
Who have you guys
Hello, I'm not sure about Asterisk and in band DTMF without careful
reading, but i do know that most ATA's and soft phones all have in band
capabilities if set. G729 may not pass in band DTMF correctly all the
time,in fact it's very poor and this is the reason for out of band. I
think from
That sure sounds like it's Analog trunks to me. I believe you will need
a channel bank to go from T1 to 24 ds0's and another to go back to T1. I
could be wrong but I don't think so. I don't think that's what you were
really looking for as far as an echo canceller.
[EMAIL PROTECTED] wrote:
I
Sorry Guys if I look dumb on this with my post but I've never seen T1's
come on in that way before. Just disregard my post.
Andrew Kohlsmith wrote:
On Friday 24 June 2005 16:17, [EMAIL PROTECTED] wrote:
The shelf has 4 25-pair amphenol connectors. The two on the line side are
marked
I can do that. Please contact me off th email net. 626-814-2354 Michael
D. Schelin - ShellTel
Lee Barken wrote:
hi Leon,
We are initially looking for US only, but eventually would like to add
international toll free numbers. We would like inbound IAX2 or SIP.
Thanks,
-Lee
On Wed
Your also not in the U.S. Out here in Southern California it's $500.00 -
$600.00 a month for T1's.
Filippo Carone wrote:
* Barton Fisher ([EMAIL PROTECTED]) ha scritto:
I found someone offering T1's for $290 a month + Loops or 3 Meg for
$561 a month + Loops. Is this a good deal?
when
Hi Martin, There was an great post last week about echo. It stated that
the order of the lines matters. It does. The channels must be listed
last for the echo cancel and most other things to work. Rx and TX gain
is one of the things also affected. Now I'm using TE110 card in my
system. I hope
I'm sorry all, lines means config lines of code.
Michael D Schelin wrote:
Hi Martin, There was an great post last week about echo. It stated that
the order of the lines matters. It does. The channels must be listed
last for the echo cancel and most other things to work. Rx and TX gain
Are you kidding! $4000.00 is cheap for a ds3 board! Even if you don't
use all of the 28 t1's it's better because you will now be able to put
in as many T1's as you will most likely need. Expansion will be just
simple configuration change. Also as I've read in these forums, the
interrupt
Thanks Rich, seems other things are now working for me as well. good FYI!
Rich Adamson wrote:
Problem solved.
this zapata.conf works (i.e echo is gone, echo cancellation is detected
on zap show channel):
context=from-pstn
switchtype = national
signalling = pri_cpe
echocancel=yes
Just asking the forum community - Is there an advantage in changing the
syntax?
Steve wrote:
I stink at regular expressions, but can always find what I need to get a
job done using google :-)
Don't use vi (unless you figure out how to do it in vi).
I won't be much help there.
in this
I have used G729 and it sounds almost as good as G711U. The problem is
the way Asterisk uses it. It does not sound robotic and it's not suppose
to sound that way. Most Carriers want the calls to be in g711u so
thats why I use G711u otherwise I want to save money on bandwidth. G729
on Asterisk
trials and this is not bad information. The
man must compare codecs on his own and see what works for him. For me
we've stuck with G711u because it's best through the PSTN. If I was
running a pure IP to IP system I would use G729, Iblc, or GSM.
Mike
Steve Underwood wrote:
Michael D Schelin
Clue or clueless? Your call.
Steve Underwood wrote:
Michael D Schelin wrote:
Steve, you should really test the Codec and have G729 running as a
pure IP to IP call you can not hear the difference on good networks!
Well, it does to anyone without hearing damage. It sounds very obviously
Hello, I too am having an echo issues. My partner an I have discussed
this in depth and believe that digital circuits can not create the echo
problem. It's when it hits the Analog network or in my cases ATA's that
are having echo problems. I have another gateway that does not have any
echo on
I have found that the audio is hot from some carriers and low on others.
I have found that this is causing the echocanclers problems. Before I
reduce it down by 3db I will see if some of the problem in in the Supura .
Andrew Kohlsmith wrote:
On May 26, 2005 01:58 pm, Colin Anderson wrote:
I can give you all the simulataneous calls you need for $.02 / min. in
the U.S. and Canada. Please call me at 626-814-2354. Michael Schelin
Shelltel
Kanuri, Seshu (Company IT) wrote:
Mike,
Many of the providers I've tried contacting either
won't call me back, or want me to
Mike - If you don't mind Los Angeles Area DID's then I can supply you
with Fixed costs with no per minute charges on your inbound calls. If
This is what We sell. Please call Michael Schelin at Shelltel 626-814-2354.
Ed Greenberg wrote:
Hi Mike,
Understand that your supplier will be paying
supply Co-Location, Power, and a fixed
amount of non shared Internet bandwidth. This can be used for Calling
card providers or call centers. We are tied onto the Verizon Backbone.
Please call Michael Schelin at ShellTel 626-814-2354 for more
information.
Michael D Schelin wrote:
Mike -
If you
The standard Windows recorder will play GSM files. You must make sure
you set the correct values. Codec, playback rate, etc.
Walt Reed wrote:
On Mon, May 23, 2005 at 03:34:44PM +0100, Brett, Gary said:
Does anybody know of a WINDOWS application (preferably freeware) that will
I'm using it and it's great. I have it doing very basic routing from
the PSTN to SIP.
Manjit Riat wrote:
Hi,
I
was going to
order the T100P but it is replaced by TE110P. On further reading the
TE110P
does not need an external router (The one that separates the
I think that it's fixed now with the V6 beta code.
Anton Krall wrote:
That's what I was starting to think.. Since I've always used ulaw or alaw...
Seems that firmware 1.0.5.23 has ilbc broken.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Kevin
TCP is too slow for Real time Apps. If you have packet errors TCP will
try to resend the packet. This will create latency issues. This is why
UDP is used for Voip. 1 or 2 missing packets is not going to be missed.
If you look at your Stats. you'll see a few of them.
Stewart Nelson wrote:
The delay in the air is minor. Radio travels very fast through the air.
Almost at the speed of light. It's the electronics that are causing the
delays. The less electronics touching your signal the better. The up
and down is very fast. But then you have all the converts and the land
line
this is beta code! I'm beta testing The t38. Don't use this unless your
testing. It is not backwards compatible.
Julio Arruda wrote:
Doug Lytle wrote:
Grandstream owners,
I just noticed that there is a new firmware release, for those that
are interested:
http://www.grandstream.com/BETATEST/
2
Please Give me a call. I'm the owner of Shelltel. 626-814-2354 We're not
the same cookie cutter VoIP carrier.
Bryce W Nesbitt wrote:
I started out happy as a clam with my new Broadvoice account and
asterisk machine. About 10 days ago things began to change
Who's happy with their voip
Isn't amazing what has happened in the last five or six years with the
Internet. There is no design flaw with IPv4. It was created back when
you were in diapers and with todays pda's having more power than the
systems back then. An industry protocol that is going strong 30 or more
years is
Hi Guys, give me a try. I'm Michael Schelin of ShellTel and we are a
business Voip service provider. I have very little down time and we
work 100% with Asterisk. Please call 626-814-2354 or email me
[EMAIL PROTECTED]. I'm a little more the the discounters but when you
need help I'm there! No
As far as I know Asterisk does not support normal PSTN type call
forwarding. I.E. the user would type *72 etc. This is called call
forking. My Mulitech gateway does but at a huge price. Also T38 is
supported. I have several carriers that I use that have Asterisk. All
of the Asterisk boxs
Hi all, What would cause the greetings not to play. The u command is
supposed to play the unavailable greeting. It doesn't work. with this
setup. Maybe I'm missing something. The voice prompts play well. What
do you think? Thanks
exten = 9007,1,VoicemailMain
exten = _.,2,Voicemail(u${EXTEN})
Have you turned off all the unused I/O ports, IE: serial, USB, Printer?
Aza wrote:
I have a problem with a Dell 1850 and a TE410P card as do a few others who
posted over the weekend.
The problem in this case isn't so much echo but static and chop on all calls
using ZAP channels. My zttest results
not when a sip call is accessed. what is
going on. When debugging the sip call there is nothing stated about
playing the client s greeting.
snacktime wrote:
On 5/4/05, Michael D Schelin [EMAIL PROTECTED] wrote:
Hi all, What would cause the greetings not to play. The u command
You Bring up a great point. I understand these codes and my system
brings them in via ss7 but as youself I don't know how to protect my
network from these charges. I will follow this post to see if anybody
has a fix.
Rodrigo P. Telles wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
Hi everybody, How do I get rid of spawn's ?
example
-- Executing Dial(Zap/23-1, sip/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
-- Accepting call from 'xxx3672728' to 'xxx2769906' on channel 0/23,
span 1
-- SIP/xxx.xxx.xxx.xxx-0adc is ringing
--
Hi, Is there a script in amp for adding the extensions? And can it be
modified? When adding a new extension it rewrites all of the
information it the context blowing out my additions.
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Hey Mojo, I'm thinking you might try using priorty 's to set some kind
routing. just a thought..
Mojo Jojo wrote:
We recently had our PRI installed, we currently have 100 toll-free's
pointing to it.
I have almost everything working great but..
I have setup the first few numbers we want to use
I just read a great paper that said turn off anything that won't be
used. Serial, USB , Printer ports, ETC. No Xwindows!
Daniel Salama wrote:
Hi,
I've been reading on the wiki as well as on this list, different
suggestions of what to look for when designing an asterisk server with
a lot of
This works from-pstn just fine.
exten = 10 digit Inbound PhoneNum from the
pstn,1,Dial(sip/[EMAIL PROTECTED] IP address,,r,)
How can I add the Variable exten with the proxie IP address. I want the
exten to call my proxie.
exten = _.,1,Dial(sip/[EMAIL PROTECTED] IP address,,r,)
Asterisk 1.0.7 built by [EMAIL PROTECTED] on a i686 running Linux
[EMAIL PROTECTED] version 0.9
CentOS release 3.4 (final)
Linux 2.4.21-27.0.1.EL
Hi All, I really need help on this. What would keep Asterisk from
playing out audio files using the (Playback command) but I can play the
busy tone .
Chris, Try upgrading Lenux. I did mine with Yum Update and now I got voice.
Chris Stinson wrote:
Has anyone else had this issue?
Original Message
Subject: [Asterisk-Users] Voicemails stopping
Date: Tue, 26 Apr 2005 13:04:55 -0500
From: Chris Stinson [EMAIL PROTECTED]
Reply-To:
FYI To All, I fixed my problem by doing a Linux upgrade by typing Yum
Update and taking the CentOS updates. My problem is solved. I have no
clue what was going on but now I have Audio Now.
Michael D Schelin wrote:
Asterisk 1.0.7 built by [EMAIL PROTECTED] on a i686 running Linux
[EMAIL
/asterisk/messages file or even on
the console with verbosity at 3 or more? I'm guessing you have a path or
permissions problem, but you should see either in the logs or the console.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Monday
Network-network-network I've sold and used them personally for about a
year now. The Network is everything. Their soild on the right network
or can be hell on the wrong ones. (some cisco systems). Also Asterisk
is not a proxie or a switch. Some things don't work (like call
forking.) I'm a sip
Hi All, I really need help on this. What would keep Asterisk from
playing out audio files using the (Playback command) but I can play the
busy tone . playtone(Congestion) ?? I have verified this with ethereal
and see the audio only going one way. In to Asterisk bun nothing coming
out.
You are talking about a sip proxie server. I don't like ser. I use a
full commercial proxie that works great but it's expensive. I believe
asterisk can do what you want but I'm not sure. I use Sipquest for my
services. I'm a provider.
Irakli Natsvlishvili wrote:
100k question - does asterisk
was because I specified the
file extension in the filename.
Eg.
Playback(file.wav) is INCORRECT. Needs to be specified as Playback(file).
Some more info may help to get your question answered!
- Original Message - From: Michael D Schelin
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
Hi Everybody can someone tell me why I can hear audio? My call is to my
proxie which is directing it to my Asterisk box. The Voice mail is
playing but I think its playing to my proxie.
the phone is on 198.31.185.246:63257
Here is from the sip debug. Thanks
Sip read:
INVITE sip:[EMAIL
Hi All, What would keep Asterisk from playing out audio files (Playback
command) but I can play the busy tone . playtone(Congestion) ?? I have
verified this with ethereal and see the audio only going one way.
Because I can hear the audio with the play tone I know there is
something
Hi Everybody can someone tell me why I can hear audio? My call is to my
proxie which is directing it to my Asterisk box. The Voice mail is
playing but I think its playing to my proxie.
the phone is on 198.31.185.246:63257
Here is from the sip debug. Thanks
Sip read:
INVITE sip:[EMAIL
Thanks all. I too have found out that the card is both.
Mike
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Craig Guy [EMAIL PROTECTED] wrote:
Can anyone with a TE110p confirm that it will fit and work in both a 3.3 and
5 volt pci slot? From photos it looks to be a
Please give me a call 626-276-9009 I'm Mike Schelin of Shelltel a
service provider in Southern Cal.
Brian Capouch wrote:
Wiley Siler wrote:
Multiple providers...
I am currently using one for outgoing exclusively due to the low latency
and excellent call quality
You mind saying who that is?
Hello Henry
em=1-23 should be bchan=1-23
you have it set for analog
also
signaling=pri_cpe
Henry Devito wrote:
Don't you need one of these
directives so the PRI knows which is master and which is slave?
pri_cpe: PRI signaling, CPE side
pri_net: PRI signaling, Network
Ok you guys enough. The debate will go on forever. The only thing
that seperates the boys from the men in this world is marketing. Beta
vs VHS.
Is Unix is better then Windows - Yes, but it doesn't matter. We live
in a Windows world because Microsoft is the greatest marketing company
on the
t_spans':
tor2.c:277: warning: implicit declaration of function
`sprintf_R1d26aa98'
make: *** [tor2.o] Error 1
Can someone tell me what is going on?
Thanks
Michael D. Schelin
SHELCOMM/ ShellTel
626-814-2354 or 626-276-9009
___
Asterisk-Users mailing
Ok I [EMAIL PROTECTED] up. I didn't realize the card is 3.3 volts and my new computer
is 5V. Can anyone point me to a PCI to PCI bridge. Any suggestions?
Mike
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Asterisk-Users@lists.digium.com
I discovered my computer is 5v and the TE110P is 3.3V Could these
errors be because there was no card?
Michael D Schelin wrote:
Hi All, I just installed a TE110P card and I'm trying to compile the
code. I followed to the letter the instructions. This is what happens
I'm a service provider and my system does not requior 5060. See if your
provider can use other ports. I would think asterisk server can be
remapped. They can't block them all!
Kanuri, Seshu (Company IT) wrote:
Yes, there is a solution.
Use IAX2 both on Server and Clients and bypass all that
Good point. Here is another Suggestion. Why not use the existing analog
phones to their PBX and go out to channel banks for their phone line
trunks. Then go to Asterisk for the rest. They don't have 700 trunks.
This will save on equipment costs and you will get some of the benefits
of
Oh one more thing. There is a 300 foot limit to Ethernet. Also the
minimum number of wires is 4.
shane fowler wrote:
we are looking at the ability of being able to convert large phone
system over to asterisk or if it's possible at all. The building is
two sections containing a large office
If your rooms analog phones are wired with cat 3 cabling you can do 10
Mb over it. Convert all the rooms to Ethernet and use large switches.
One Asterisk box should do the trick. Remember not every room will be
using the phone system at the same time. This should work for you.
shane
If your in Los Angeles Call me I've got 130,000 numbers with caller ID
from my ss7 network. Trust me there's a whole lot more to it than what
he just said.
Mike
trixter http://www.0xdecafbad.com wrote:
On Thu, 2005-04-14 at 19:37 -0700, snacktime wrote:
I knew xo/level3 were
What would cause this error. My server is not busy. I'm trying to ger
the voice mail to work without any PSTN extensions or cards. Just a sip
Mailbox.
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Hi all How do i set up voice mail playback using * as the inertupt . I
can't seem to figure out hou to use VoiceMailMain([EMAIL PROTECTED])
Thanks
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Hi all How do i set up voice mail playback using * as the inertupt . I
can't seem to figure out hou to use VoiceMailMain([EMAIL PROTECTED])
Thanks
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Hi everyone, Why doesn't this work? I can't get in. Is it because I
changed the root?
User: admin
Pass: password
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To UNSUBSCRIBE
Your right on the overpriced junk! But yes now it works great.
Rod Bacon wrote:
I found new firmware at:
ftp://ftp.us.zyxel.com/P2000W/firmware/P2000W_WJ.00.10_Standard.zip
The phone is now finally (almost) useful. Still a cheap piece of crap,
with new bugs to replace the old, but at least it
There is very little difference between configuring a static IP or
DHCP. You need the basic 3 things like the IP address, Sub net mask,
and Gateway address. For DNS use the dns servers address's supplied by
you ISP. Make sure you turn on the use DNS setting in the Sipura unless
you use IP
The only way you ll be able to call extension to extension is if
Asterisk is on the same node behind the nat. like the extensions or if
each extension is on a different node. I run a proxie server and have
ran through this problem many time. I bet you can call out bound to the
outside world
, 07 Apr 2005 21:17:03 -0700
From: Michael D Schelin [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] T.38 fax with SIP devices
To: Scott Wolfe [EMAIL PROTECTED],Asterisk Users Mailing List -
Non-Commercial Discussionasterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type
Grandstream has the same problem. Very common. A simple DTMF debouncer
curcuit will fix it.
Doug Meredith wrote:
Eric Wieling [EMAIL PROTECTED] wrote:
Daryll Strauss wrote:
Yep, I've seen it and from reading http://www.voxilla.com it's a
pretty common problem.
I may be able to help. I'm a provider in Southern CA. What you need to
do is eliminate all pots lines by moving them over to VOIP completely.
This will take some time but will save you a lot of money. Please call
me for more info. I can provide you service and if your interested
LNP your
Hello everyone, I need to configure a stand alone Voice mail box. Calls
will come in via sip. I have read and read until my eyes hurt for 2
weeks now. Can someone email me the basic config files needed to do
this. The examples are overly complicated. I just need a simple basic
configurations
can tell you in about a year or so that will be a thing of the
past.
snacktime wrote:
On Apr 7, 2005 6:32 PM, Michael D Schelin [EMAIL PROTECTED] wrote:
I may be able to help. I'm a provider in Southern CA. What you need to
do is eliminate all pots lines by moving them over to VOIP
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