> On Thu, 14 Oct 2004 20:44:40 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> > GNU/Linux was licensed under a BSD-style license then Red Hat could
> > easily close the source - just as Apple did when they stole BSD code
> > to create "their" OS/X effort.
> >
> With all due respect, Kevin, you are
Hi,
Over the last few months a number of Asterisk users have asked me about
implementing signaling protocols most of us would consider obsolete.
There is an amazing about of very old equipment still in use. However, I
currently have no way to judge whether these inquiries are serious, or
more t
Michael Rowley wrote:
Do you have a link to this? I have done a search, and cannot locate
it. I think I may be searching with the wrong terms.
Any help would be appreciated.
Michael
http://www.voip-info.org/tiki-index.php?page=Cisco+7940-7960+auto-answer+config
--
Kristian Kielhofner
__
Stan Brinkerhoff wrote:
> A friend of mine has a real panasonic PBX setup at his house, and is
> able to pick up the phone, dial an extension, and it broadcasts what
> he says over every phone in his house without the phones having to be
> picked up.
>
> What is this feature called?
>
See:
> Well basically we are a small operation. To begin with we really don't want
> any voicemenu ("press 1 for this, press 2 for that"). I just want all the
> incoming calls to ring to the 2 receptionists (and then maybe everywhere
> after a while). However, on our current analog (comdial) system y
Hi to all,
I just wanna know
if Asterisk works properly with Cisco 5350/5400 Gateway. I register my Asterisk
box on a Cisco 5350/5400 Gateway for outgoing calls, and Asterisk is sending
errors messages to the Cisco Gateway, and shut it down after a short period. Is
there any specific confi
Hi Eveyone,
I am a software engineer working for a small firm in Tokyo.
I found Asterisk to be very very intersting.
I am looking for Japanese translation sites or web pages or
documents.but could not find any.
I am planning to use Asterisk on my project, and was thinking if there
are not transla
On Sat, Oct 16, 2004 at 02:34:20AM -0400, Jason Price arranged a set of bits into the
following:
> just downloaded the latest cvs on * and chan_sccp , it wont build, i
> have edited the make file per the README
> [EMAIL PROTECTED] chan_sccp]# make
> Now compiling sccp_channel.c 27
Continuing on/adding my $0.02 to Joe's reply on this thread
--On Friday, October 15, 2004 23:30 -0500 Joe Greco <[EMAIL PROTECTED]>
wrote:
If Extension 1000 is a Zap channel (i.e. an interface card that has an
RJ11 into which you plug a phone), that may not be able to receive more
than one ca
Yes, it's compatible.
If you have "signaling forward unconditional" on the dial-peer on on
sip-ua, won't work (look at previous threads from this list).
If that didn't work, maybe you can post your config for review.
On Oct 16, 2004, at 3:03 AM, Daniel Eboa wrote:
Hi to all,
I just wanna know if
Hello Emilio,
I can't send the Cisco Gateway config since is handled by my SIP provider; I can send
my sip.conf file. Can't send me the cisco Gateway config that match with my asterisk
config ??
-Original Message-
From: Emilio Panighetti [mailto:[EMAIL PROTECTED]
Sent: samedi 16 oct
On Sat, 16 Oct 2004 16:04:18 +0900, Memo Memo <[EMAIL PROTECTED]> wrote:
> I am a software engineer working for a small firm in Tokyo.
> I found Asterisk to be very very intersting.
>
> I am looking for Japanese translation sites or web pages or
> documents.but could not find any.
>
> I am planni
Chris Travers <[EMAIL PROTECTED]> writes:
> Now, TCP connections will probably be interrupted in any case if
> your IP address changes, but that is the nature of the protocol.
That's why I use a tunnel. All TCP connections are bound to the
address of the tunnel end point, which doesn't change.
Hi All,
Is my setup possible? Or maybe the right question is "Is this correct?"
1 test server: installed ser and asterisk(didn't really understand much
of it yet).
1 CISCO 1750: with 2 FXO.
2 UA's: X-Lite
Using SER only, I can make calls between extensions.
Using Asterisk alone, can I also ca
Julien Goowin wrote:
>[SEP]
>addon = 1 ; How many 7914's are connected
>Also check that you are using current chan_sccp with asterisk 1.0, but
>7914 support has been in chan_sccp for a few months now.
>You probably also wish to add some more speeddials and lines to be able
>to properl
On Thu, 14 Oct 2004 20:44:40 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> GNU/Linux was licensed under a BSD-style license then Red Hat could
> easily close the source - just as Apple did when they stole BSD code
> to create "their" OS/X effort.
With all due respect, Kevin, you are talking out
Hi,
I read a lot about having problems with latencies, interrupts on some
motherboards.
Could you please be so kind to recomend me brand or type of motherboards
that are working flawlessly for Asterisk (Gigabyte, Asus, Intel, some
other...) ?
Is there any site about PC motherboards particularly
Hello All,
I purchased yesterday two G729 licenses from Digium to
my asterisk box. I used the register utility and i follow
the installation procedures as describes the README.
I forced my sipuras to use G729a protocol and on my sip.conf
too.
I get a message that there's no compatible codecs!!!
Wha
Hi all,
I have troubles getting outgoing calls to work with sipgate. Tried all
possible constellations, copied posted configs, nothing. Registration or
incoming calls never a problem. If I use a SIP phone from my network it
works. * doesn't.
I always get 482 Loop Detected
-
On Sat, Oct 16, 2004 at 12:13:03PM +0200, Robert Rozman wrote:
> Could you please be so kind to recomend me brand or type of motherboards
> that are working flawlessly for Asterisk (Gigabyte, Asus, Intel, some
> other...) ?
I have had great success with the VIA EPIA-M. This board, plus a TDM400,
I missed the last bit of the subject -- the VIA board has its own fanless
processor, and as such won't socket a P4. It's still a good choice, though :)
--
-- Chris
GPG key FEB9DE7F (91AF 4534 4529 4BCC 31A5 938E 023E EEFB FEB9 DE7F)
___
Asteri
I tried the mailbox statement and it didn't work.
-Original Message-
From: Brian McSpadden [mailto:[EMAIL PROTECTED]
Sent: Saturday, October 16, 2004 1:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IaXY MWI
I don't know for sure, but I wo
On Sat, 16 Oct 2004 12:23:15 +0800, Dinesh Nair <[EMAIL PROTECTED]> wrote:
> in this vein, does digium have plans for multi-port IAXy devices ? US$100 a
> pop seems expensive in this part of the region, viz a viz 4-8 port SIP FXS
> devices.
It would appear that Digium are already busy with the stu
- Original Message -
From: "Cirelle Enterprises" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]>
Sent: Friday, October 15, 2004 4:58 PM
Subject: Re: [Asterisk-Users] T100P Frame Errors
- Original Message -
From: "Steven Cri
Hi,
I thought I would convert my old PowerMac G4 into a Linux box /
Asterisk server, but I haven't been having a lot of success.
Call signaling seems to work fine - detects phones going on/off hook
and phones ring when you dial them (from the console) but audio doesn't
seem to be working at all
In article <[EMAIL PROTECTED]>,
Stefan de Konink <[EMAIL PROTECTED]> wrote:
> Tony Mountifield wrote:
> > I want to create a CGI or PHP script that, when invoked from the web,
> > will make a call to Asterisk on a given extension number, and the audio
> > that is played to that extension gets strea
On Sat, 16 Oct 2004, Christoph Kampka wrote:
> exten => _920.,2,Dial(SIP/$(EXTEN:3)@sipgate)
${EXTEN:3}
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Please keep us informed about your progress. I'm interested in this too.
Regards,
Robert.
- Original Message -
From: "Tony Mountifield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, October 16, 2004 3:24 PM
Subject: [Asterisk-Users] Re: Web stream from an extension?
> In
> Joe Greco [EMAIL PROTECTED] wrote:
> > > Your scenario can be played out any number of ways, with or without
> > > source. You routinely send your life critical hardware down to Bob and
> > > Doug's repair shop? You have bigger issues in place.
> > >
> > What are we talking about "Bob and Doug's
On Sat, 16 Oct 2004 14:06:24 +0100, Nicholas J Humfrey
<[EMAIL PROTECTED]> wrote:
> I thought I would convert my old PowerMac G4 into a Linux box /
> Asterisk server, but I haven't been having a lot of success.
>
[snip]
> I am running a custom compiled 2.4.27 kernel on Debian/sarge and
> version z
> Continuing on/adding my $0.02 to Joe's reply on this thread
>
> --On Friday, October 15, 2004 23:30 -0500 Joe Greco <[EMAIL PROTECTED]>
> wrote:
>
> > If Extension 1000 is a Zap channel (i.e. an interface card that has an
> > RJ11 into which you plug a phone), that may not be able to recei
You can do all of with with each one, and with both at the same time
(.e., register your UAs through ser to in term rute to Asterisk.
Ser is, simply put. a (stateless) SIP Proxy. It's like a SIP
layer-level router. Which is good if you need something that can scale
a lot.
Asterisk is a full-fea
to know for sure how's the configuration on the gateway's side, do this:
From Asterisk console, type sip debug ip XXX.XXX.XXX.XXX where
XXX.XXX.XXX.XXX is the IP address of the Cisco gateway.
Then make a call and look at the debug. If in the debug you see
something like ("---UniqueBoundary"), the
Jefferson,
I have it working great.
What do yo see in * show translation command ?
Kind regards,
Miguel
Date: Sat, 16 Oct 2004 07:27:33 -0300
From: Jefferson Carvalho <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] G729 and Sipura.
To: Digium/ Asterisk-Users <[EMAIL PROTECTED]>
Message-ID: <[EM
A friend of mine has a real panasonic PBX setup at his house, and is
able to pick up the phone, dial an extension, and it broadcasts what he
says over every phone in his house without the phones having to be
picked up.
What is this feature called?
Would it be possible to set this up with Aster
Hi,
On Sat, 16 Oct 2004, Stan Brinkerhoff wrote:
> A friend of mine has a real panasonic PBX setup at his house, and is
> able to pick up the phone, dial an extension, and it broadcasts what he
> says over every phone in his house without the phones having to be
> picked up.
>
> What is this fea
http://www.voip-info.org/wiki-Asterisk+Data+Configuration
The newest example on this page dated Oct 6 2004 appears
to be incorrect
Regards
Greg Cirino
___
Cirelle Enterprises Inc.
603-425-2221
www.cirelle.com Website Design
www.cirelle.net ProSpeed High Speed Dial-
An incomplete class 1 modem for hylafax... Really looking forward to
this feature. I'm pretty sure my Sipura in front of the actual fax
modem is causing me all the headaches with HylaFax. :)
As an aside, we're all very grateful for your work and support Steve.
On Sat, 16 Oct 2004 13:07:58 +0800
Stan Brinkerhoff wrote:
A friend of mine has a real panasonic PBX setup at his house, and is
able to pick up the phone, dial an extension, and it broadcasts what he
says over every phone in his house without the phones having to be
picked up.
What is this feature called?
Would it be possible to
Hi.
I looked at some examples with Cisco gateways with FXO ports, but I
have DIDs on ISDN lines. I don't know what I'm missing. In fact, my
gateway can connect directly to the IP Phone's IP address and to SER,
and I see what it seems a normal SIP message on Asterisk's debugs, then
somehow looks
I've use Sipuras with * using G729 - with no problems.
Double check that G729 is turned on in the sipura and your sip.conf is
correct - if anything post excerpts from your sip.conf.
On Oct 16, 2004, at 6:27 AM, Jefferson Carvalho wrote:
Hello All,
I purchased yesterday two G729 licenses from Digi
Hi,
I have a Grandstream ATA today connected to my 750k broadband
connection via an older router / firewall that doesn't have any QoS /
ToS capability. It works fine apart from the obvious problem of when
large emails come in or somebody else on the network starts d/l-ing
something big off the
Adam Holt wrote:
Hi,
I have a Grandstream ATA today connected to my 750k broadband connection
via an older router / firewall that doesn't have any QoS / ToS
capability. It works fine apart from the obvious problem of when large
emails come in or somebody else on the network starts d/l-ing somet
Broadcom has released those new Wifi Voip chipsets that some of us have
been hearing about. From the looks of the product sheets from Broadcom,
they look very exciting:
- 802.11g support
- SIP support
- Bluetooth
- g729, g711, g726
- 802.11 qos, roaming
- 1.3 megapixel Camera
- USB 1.1
- 262k c
I am not having any luck getting a NOANSWER call to go to voicemail.
If I set a timeout time on the Dial command I can get the "t" for
timeout to work.
I let the call ring 20 times and it continues to ring it doesnot go to
the +101 priority.
Where is the setting that controls number of rings noa
Christopher SEKIYA wrote:
I missed the last bit of the subject -- the VIA board has its own fanless
processor, and as such won't socket a P4. It's still a good choice, though :)
I was under the impression that those boards (and all/most VIA's) have
serious PCI latency issues and thus are not go
Well basically we are a small operation. To begin with we really don't want
any voicemenu ("press 1 for this, press 2 for that"). I just want all the
incoming calls to ring to the 2 receptionists (and then maybe everywhere
after a while). However, on our current analog (comdial) system you can
Jerry Geis wrote:
I am not having any luck getting a NOANSWER call to go to voicemail.
If I set a timeout time on the Dial command I can get the "t" for
timeout to work.
I let the call ring 20 times and it continues to ring it doesnot go to
the +101 priority.
Where is the setting that controls n
anyone find a bug/fix for this i have clean my cvs files and tried to
rebuild.. same issues.. and my 12sp is still looking at me unused =>
Jason
On Sat, 16 Oct 2004 17:04:18 +1000, Julien Goodwin
<[EMAIL PROTECTED]> wrote:
> On Sat, Oct 16, 2004 at 02:34:20AM -0400, Jason Price arranged a set of
Hello all,
as a relative newbie I obediently ordered the recently recommended
"Newton's Telecom Dictionary" but it did not provide the answer to one
of my question. Perhaps one of you can do so.
What is the difference in functionality between an FXO and an FXS port?
I know that the line from telco
On Friday 15 October 2004 23:28, Joe Greco wrote:
> > I know what you're trying to do (we're both playing Devil's Advocate).
> Not really. There is a product. If you go into a hospital for major
> surgery today, you stand a nontrivial chance of being hooked up to a
> descendant of the device in
The way I explain the difference between fxo and fxs, is electrically...
The telco lines, PROVIDES voltage and current to makes the telephone
ring, so the FXO is a device, adapted to recive this voltage an current.
The phone, NEEDS voltage and current, to ring, so the FXS is a device,
adapted to
On Sat, 16 Oct 2004, Wolf Paul wrote:
> What is the difference in functionality between an FXO and an FXS port?
> I know that the line from telco goes into an FXO and standard POTS phone
> goes into an FXS, but since both types of ports must be able to both
> make and answer calls, what is the a
On Sat, 16 Oct 2004 19:05:34 +0200, Wolf Paul <[EMAIL PROTECTED]> wrote:
> What is the difference in functionality between an FXO and an FXS port?
> I know that the line from telco goes into an FXO and standard POTS phone
> goes into an FXS, but since both types of ports must be able to both
> mak
> as a relative newbie I obediently ordered the recently recommended
> "Newton's Telecom Dictionary" but it did not provide the answer to one
> of my question. Perhaps one of you can do so.
>
> What is the difference in functionality between an FXO and an FXS port?
> I know that the line from telc
Andres Tello Abrego wrote:
The way I explain the difference between fxo and fxs, is electrically...
The telco lines, PROVIDES voltage and current to makes the telephone
ring, so the FXO is a device, adapted to recive this voltage an current.
The phone, NEEDS voltage and current, to ring, so the F
I have blown one fxs module in that way :(
It´s a warning, and you MUST be careful of tring no to do so.
Steve Underwood wrote:
Andres Tello Abrego wrote:
The way I explain the difference between fxo and fxs, is electrically...
The telco lines, PROVIDES voltage and current to makes the telephone
r
While I agree that the VIA is an exciting platform for all kinds of
embedded applications, it is important to note that the FPU on the Eden
and C3 processors is considered to be very low performance. Since
Asterisk makes heavy use of the FPU, a system based on these CPUs will
not be able to handle
The cat's ass would be an IAX channel bank, modular so it can be
provisioned for as many (few?) FXS/FXO ports as desired.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Benjamin on Asterisk Mailing Lists
> Sent: October 16, 2004 7:20 AM
> To:
Do you have a link to this? I have done a search, and cannot locate
it. I think I may be searching with the wrong terms.
Any help would be appreciated.
Michael
On Oct 16, 2004, at 11:34 AM, Kristian Kielhofner wrote:
Stan Brinkerhoff wrote:
A friend of mine has a real panasonic PBX setup at his
I have the same problem. I just gave up.
jim
On Sat, 16 Oct 2004, Jason Price wrote:
> anyone find a bug/fix for this i have clean my cvs files and tried to
> rebuild.. same issues.. and my 12sp is still looking at me unused =>
>
> Jason
>
>
> On Sat, 16 Oct 2004 17:04:18 +1000, Julien Goo
>or
>if policy is not followed, well then Bad Things are MUCH more likely to
>occur. With or without source being available.
>
>I suppose that having source can make the possibility for the occurance of
>Bad
>Things marginally higher but it all comes down to design and policy, IMO.
And thus, y
> I have a Grandstream ATA today connected to my 750k broadband
> connection via an older router / firewall that doesn't have any QoS /
> ToS capability. It works fine apart from the obvious problem of when
> large emails come in or somebody else on the network starts d/l-ing
> something big o
you might consider http://m0n0.ch/wall on a soekris.com or
pcengines.ch board which does nice trafficshaping for little money.
m0n0wall is a freebsd based opensource firewall appliance
On Sat, 16 Oct 2004 12:25:06 -0600, Rich Adamson <[EMAIL PROTECTED]> wrote:
> > I have a Grandstream ATA today c
On the site note, I was able to get a reply from Welltech, that accepts
this and 3 other bugs with 35xx and 38xx SIP versions, and saying that
all those require a new firmware, and that it'll take a lot of time for
them to release one, quoting: "The next released version will be take a
long tim
Joe Greco [EMAIL PROTECTED] wrote:
> > Distributing the source would allow peer review; The hospital staff,
> > and other interested parties, could point out potential bugs, suggest
> > possible enhancements and even provide patches for consideration that
> > could turn out to be of benefit to hea
> I can't seem to keep my Asterisk box registered with stanaphone for
> inbound calls. It works for about 5 minutes after a restart or reload,
> then calls to my Stanaphone number say I'm unavailable. I can't really
> blaim their end because if I configure my Sipura 2000 to talk directly to
> them,
The horse has been dead for a long while. Please stop beating it.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Saturday, October 16, 2004 4:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk
I have a Panasonic switch here and it a paging system on the switch.
It will output the page message to all phones and also to an RCA (Phono)
socket on the side of the switch to a PA amplifier if required to drive a
100Volt line system around a building.
Dave
-Original Message-
From: [EM
i have a simple question, why are you switching to an ip solution? does
your exsisting telephone network work for your company? if so, why not
just use asterisk to provide your exsisting network with calls, that way
you can continue to use your exsisting hardware, and still use asterisk.
if your j
The real problem is that the old comdial is dying and cannot support any
more extensions. Not to mention constant static problems on speakerphone.
(even after repunching the all the cables + replacing them).
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
Thanks Mark,
I have tried your config and variations on it but have the same problems.
Placing a call out using intervivo, regardless of dtmfmode setting, DTMF
tones are not recognised by the recipient. The same applies to receiving
dtmf digits.
Also, unless I set insecure=very (which I shouldn'
The Polycom phones will do this. Use the meetme feature. It's well
documented on the Wiki.
John
David J Carter wrote:
I have a Panasonic switch here and it a paging system on the switch.
It will output the page message to all phones and also to an RCA (Phono)
socket on the side of the switch to
iH
getting the following in the console when i go off hook from IAXy
chan_iax2.c:5713 socket_read:Rejected connect attemp from 192.168.0.5
requested/compability 0x4/0x24 incompatible with our capability 0xff3
any idea what this is telling me?
thanks
- hcir
You are more then welecome to edit the page and fix it. It is a WIKI
after all. Simply sign up for an account, then click the little "EDIT"
button at the top of the page you wish to edit.
-Brian
Cirelle Enterprises wrote:
http://www.voip-info.org/wiki-Asterisk+Data+Configuration
The newest examp
could you paste the file u are trying to provide?
rich allen wrote:
iH
getting the following in the console when i go off hook from IAXy
chan_iax2.c:5713 socket_read:Rejected connect attemp from 192.168.0.5
requested/compability 0x4/0x24 incompatible with our capability 0xff3
any idea what this
well it looks like i had a typo in the iax.conf. instead of
disallow=all i had dsallow=all which i didnt notice til i about to
paste the file.
thanks for responding
- hcir
could you paste the file u are trying to provide?
rich allen wrote:
iH
getting the following in the console when i go off h
You need to either download 12.3(11)T or 12.3(10)LD.
Kurt
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:) I supposed that
iaxy are s beatiful!
rich allen wrote:
well it looks like i had a typo in the iax.conf. instead of disallow=all
i had dsallow=all which i didnt notice til i about to paste the file.
thanks for responding
- hcir
could you paste the file u are trying to provide?
rich allen w
[macro-say-digits]
;
; Usage: exten => 666,1,Macro(say-digits,915045551212)
;
; This macro uses SayDigits to read numbers with logical pauses like a human
; might say. Please send changes/patches/suggestions to [EMAIL PROTECTED]
;
; Should handle 1-6 digits, 7 digits, 8 digits, 10 digits, 11 digit
Title: Message
Hello
List,
My * server is NAT'd
behind a firewall.
What ports do I need
to open to allow a Grandstream IP to connect to it remotely?
Thanks
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Dear Everyone,
(Hi, Benjamin & Isamar )
I am happy with the reply, and kind of looking forward to work with
you guys to help Asterist project.
I will send my contact details to both of you personaly.
Thanks.
On Sat, 16 Oct 2004 17:10:44 +0900, Benjamin on Asterisk Mailing Lists
<[EMAIL PROTEC
Use only g729 and not g729a. I tied it and it works greate with Granstream
Phone.
- Original Message -
From: "Jefferson Carvalho" <[EMAIL PROTECTED]>
To: "Digium/ Asterisk-Users" <[EMAIL PROTECTED]>
Sent: Saturday, October 16, 2004 3:27 AM
Subject: [Asterisk-Users] G729 and Sipura.
> He
The term "FX" refers to an old telecom service that provided a "Foreign
eXchange" circuit - basically an analog extension that the telco would
terminate at a remote location and connect back into your PBX.
The terms "FXS" and "FXO" came to refer to the respective ends of the
circuit, where "O" sta
Be sure, not to write "allow=g729a" in your sip.conf !
Correct entry is "allow=g729"
Regards
Jefferson Carvalho schrieb:
Hello All,
I purchased yesterday two G729 licenses from Digium to
my asterisk box. I used the register utility and i follow
the installation procedures as describes the README.
I
Actually, the IAXy is really very immature, and is desperate for several
enhancements:
1. The MAC address needs to be visible on the unit. As it stands the
only way to determine the identity of the thing is packet sniffing or
looking through logs. This is tolerable in a lab, but ridiculous in
prod
> Joe Greco [EMAIL PROTECTED] wrote:
> > > Distributing the source would allow peer review; The hospital staff,
> > > and other interested parties, could point out potential bugs, suggest
> > > possible enhancements and even provide patches for consideration that
> > > could turn out to be of bene
> Joe, could we stop this now? It's obvious that if you go to a GPL
> project and start slinging mud at the GPL, you are in the wrong place.
No, I was making a point. It's possible for the GPL to be a negative
effect on a project. We led into the discussion in a fairly reasonable
train of tho
Joe Greco [EMAIL PROTECTED] asked his mother to write:
> Kevin Walsh wrote:
> > There's no point in continuing this "discussion" as you're clearly not
> > a member of the FOSS community. It seems that you're just here in a
> > futile attempt to gather support for your out-dated development model.
Title: Message
I also would like to know the ports needed
for * to run without any issues. I am not using NAT but I do want a firewall in
front of the machine.
Thanks,
Todd Routhier
Lightwave Technologies, LLC.
--
Start Your Dialup Internet Service!
http://www.YourOwnISP.c
I hope this is appropriate for this list..
Can anyone suggest a good place on the web to buy ATA's and IP phones.
Hopefully a site that carries a lot of different products with good pricing.
Also, looking for a site that maybe has reviews or user ratings of different
types of equipment. I need to
netstat -a -p -n | grep asterisk
will give you a very good idea of what ports & protocol does asterisk
have opened..
Your Own ISP .com wrote:
I also would like to know the ports needed for * to run without any
issues. I am not using NAT but I do want a firewall in front of the machine.
Thank
Jim Van Meggelen wrote:
Actually, the IAXy is really very immature, and is desperate for several
enhancements:
Yeap, rigth.
1. The MAC address needs to be visible on the unit. As it stands the
only way to determine the identity of the thing is packet sniffing or
looking through logs. This is toler
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Andres Tello Abrego
> Sent: October 16, 2004 11:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] IAXy setup
>
>
> Jim Van Meggelen wrote:
> > Actu
The real problem is that the old comdial is dying and cannot support any
more extensions. Not to mention constant static problems on speakerphone.
(even after repunching the all the cables + replacing them). We just really
want a new system that will be expandable for the future.
> -Origina
Try Chagres.net
I dont work for them , but I get a very good service from this guys
and good price and quick response.
Regards
Humberto
(List Please tellme if this kind of comments are permitted)
On Sat, 16 Oct 2004 22:02:56 -0500, Your Own ISP .com
<[EMAIL PROTECTED]> wrote:
> I hope thi
The audio is carried on two RTP streams: one for each direction. Is it
possible those streams are being blocked by a firewall or something of
the sort?
The "attempting native bridge" message means that Asterisk is bridging
the two calls together without doing any codec translation... uLaw to
- Original Message -
From: "Adam Holt" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Hi,
I have a Grandstream ATA today connected to my 750k broadband
connection via an older router / firewall that doesn't have any QoS /
ToS capability. It works fine apart from the obvious problem of wh
On 17/10/2004 03:27 Vahan Yerkanian said the following:
On the site note, I was able to get a reply from Welltech, that accepts
this and 3 other bugs with 35xx and 38xx SIP versions, and saying that
all those require a new firmware, and that it'll take a lot of time for
oh, the holy grail: a res
> Joe Greco [EMAIL PROTECTED] asked his mother to write:
Oh, brilliant. You completely ignored a reasonable request to demonstrate
your position, instead supplying us with that imaginative attribution.
> > Kevin Walsh wrote:
> > > There's no point in continuing this "discussion" as you're clearl
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