Yes this is quite an issue. The POE converter is 'optional'. I bought
a 480i a while back and after waiting a few days had to order the POE
cos the dealer hadn't told me it was actually required!
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Dear friends,
As I commented some while ago in the list, occasionally when DTMF Tones are
sent, they appear in RTP Payload and in Events too, producing duplicate
tones being recognized. This behavior happens in Asterisk as well as in
Gateways such as Cisco, for which we had the opportunity to
Hi everybody.
I'm trying to setting up a V4 BRI EICON card on ASTERISK 1.0.7
My linux is a debian.
It was working during a few days an suddenly (after a lot of reboot)
I've got this error message that seems to be very popular but I couldn't
find any
answer on the net :
Hello all,
I want to sniff all these info to test a sip ip phone talking
to a asterisk server. I have used tcpdump, but It just shows the
UDP, length: 602
Anyway to see the sip uri. Host info?
Regards,
Dinesh.
___
On Mon, Feb 20, 2006 at 06:24:16PM -0500, Alexander Burke wrote:
I really appreciate the replies I've gotten about this so far
(especially the support for wanting to run it on Solaris!).
The core issue seems to have been missed, though -- is there any way
to run a complete Asterisk solution
On Tue February 21 2006 18:53, Dinesh [EMAIL PROTECTED] wrote:
Hello all,
I want to sniff all these info to test a sip ip phone talking to a
asterisk server. I have used tcpdump, but It just shows the
UDP, length: 602
Anyway to see the sip uri. Host info?
Regards,
Dinesh.
Hi Dinesh.
I'm configuring a sip trunk. My problem is if I configure the sip
device to dial to a sip phone, it works ok but when I dials to s or
, asterisk picks up the call immediatly and places it's own ring
tone instead of waiting until one of the extension configured for
answer the call picks up.
In article [EMAIL PROTECTED],
Alexander Burke [EMAIL PROTECTED] wrote:
Hello, list!
I'm hosting a mirror of the book Asterisk: The Future Of Telephony
by O'Reilly Press, published under the Creative Commons license; I
believe this license allows me to do this, but if I'm mistaken,
please
2nd vote for ADPCM - depends on how fat you can get though? I would guess
though that this is over a smallish pipe?
After a lot of time and various experiments, my preferred codec is G.726/32
in combination with RTP header compression - low impact on the WAN and the *
server but quality that is
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
But using the native transfer on the phone causes the system to think the
agent is still on the call
Yes, and I have desabled that options on my phones. Sometimes I have delay if I
use transfer or three way calling on Cisco phones.
Hello
I need some asterisk expert on setting up incoming DID on asterisk
Please email me back or msn me on sam__tam AT hotmail DOT com
$£ waiting..
Sam
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
Where are you located?
Paul Hales
Melbourne, Australia
- Original Message -
From: Sam Tam [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, February 21, 2006 8:52 PM
Subject: [Asterisk-Users] $ for an hr of
This sounds more like a dialplan issue - and what has got to do with
anything?
PaulH
- Original Message -
From: Alejandro Vargas [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 21, 2006 8:16 PM
What version of chan_capi do you use? Your capi.conf is for an old
chan_capi. If you use an old version, please update to chan_capi
from sourceforge.net and adapt your capi.conf.
Armin
On Tue, 21 Feb 2006, cédric Buzay wrote:
Hi everybody.
I'm trying to setting up a V4 BRI EICON card on
2006/2/21, Alejandro Vargas [EMAIL PROTECTED]:
I'm configuring a sip trunk. My problem is if I configure the sip
device to dial to a sip phone, it works ok but when I dials to s or
, asterisk picks up the call immediatly and places it's own ring
tone instead of waiting until one of the
2006/2/20, Rich Adamson [EMAIL PROTECTED]:
I'd suggest reading over the info at www.voxilla.com as the interface
from the pstn to asterisk is a little different from what one would
consider normal.
I solved the problem. It were easy: if you has enabled the authomatic
fax detection, asterisk
Hello, Steve!
At 03:55 AM 02/21/2006, you wrote:
ztdummy was only used for timing. Linux 2.6 provides this function in
the kernel and I assume Solaris already has timing functions there.
Page 36 of Asterisk: The Future Of Telephony
(O'Reilly Press) states that you either require a
Digium
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Does anyone know how to setup a linear type of queue strategy? By that
I mean that agents will be tried in a particular order and the call will
be routed to them unless they are on the phone or not logged in.
I want a 3rd party app to
While it doesn't explicity say so, it seems to
very strongly imply that either a PCI card or
ztdummy are *required* for some Asterisk
functionality (namely music-on-hold and
conferencing, apparently). Is this actually not the case?
I'd say support for one of these options should be
Hi
I have a test setup of a sirrix card installed in NT mode connected to a
PBX. I keep getting the following error:
D-Channel receive message aborted, discarding frame (RSTAD=0x1c)
What does this mean? What could be causing it?
Garth
___
On Tue, Feb 21, 2006 at 06:16:06AM -0500, Alexander Burke wrote:
Hello, Steve!
At 03:55 AM 02/21/2006, you wrote:
ztdummy was only used for timing. Linux 2.6 provides this function in
the kernel and I assume Solaris already has timing functions there.
Page 36 of Asterisk: The Future Of
Basically, I've setup the phone following the instructions at
voip-info.org, and it registers for about 10 seconds, then after
receiving the SIP NOTIFY from the * server, goes into flashing display
mode, which indicates some sort of connectivity error. I've tried all
The flashing
Interesting. I installed Fedora Core 4 and whenever I load ztdummy I
get stuttering and a robotized voice but when I don't modprobe ztdummy
it works fine.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Kennedy
Sent: Tuesday, February 21,
I kind of like the idea of 2n's stargate but when i read the
manual (the one available for download), there were a lot of
complicated issues in configuring the device, (i mean, you have to like
set jumbers on the m/board,etc) and there was a clause that said that
callc could only be routed form
Is it possible to send an API command to dial an extension and playback a
specific announcement using application and appdata commands.
Scenario:
User adds different announcements daily (can't used fixed name for Playback
file).
Call command dials user and plays back specific announcement
Guys.
I would like to hear some comments about people using polycoms 600 IP phones
and what their doing with their minibrowsers? Any inetresting apps that you
might want to share?
Thanks
AK
___
--Bandwidth and Colocation provided by Easynews.com --
On Tue, Feb 21, 2006 at 04:53:43PM +0800, Dinesh wrote:
Hello all,
I want to sniff all these info to test a sip ip phone talking to a asterisk
server. I have used tcpdump, but It just shows the
Ethereal would probably be a batter analyzer. Not sure how well it
seppurts sip, though.
I am not running trunked IAX.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Willis
Sent: Monday, February 20, 2006 8:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New
Title: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
This is not going over the Internet. It is going over
an MPLS IP-VPN.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael J.
LiberatoreSent: Monday, February 20, 2006 7:55 PMTo:
Asterisk Users
Marc,
I have a box with two TDM400P's. All of the ports are
FXO's. System is working fine on CentOS.
Regards,
Dovid
Can someone give me a definite answer as to wether
or not you can
reliably run multiple TDM400P's in the same machine?
I need 4 x FXO and 4 x FXS to connect to both the
PSTN
Hey everyone,
I've been doing a lot of research into a decent server for Asterisk
but I seem to be running and circles and now I am turning to you. The
issue I have is it needs to be rack mountable (so a Dell SC430 isn't
going to work) and preferably have 3 pci ports. The problem that I
seem to
I know one of the guys there that runs the place. They
know a lot about asterisk. I cant say all that I know
but I will just say that soon they will be very
asterisk friendly. As far as getting a plan without an
adapter they do have a plan. It is called a
myDevicePlan. I am not sure if its on
Ethereal would probably be a batter analyzer. Not sure how well it
seppurts sip, though. Unlike tcpdump it won't work on-the-fly. But you
can also get tcpdump to dump raw data and analyze it off-line with
ethereal.
Ethereal can also show SIP traffic on-the-fly!
update list of packets in
Again. What do you need ? Incoming and outgoing,
trunking etc. ?
I personaly use.
Voipjet.com
myPhonecompany.com
Teliax.com
I have heard others talk about:
JunctionNetworks
There others that are just not coming to mine. If I
remember them I will try to email them as well.
Dovid
Everything. I
I do not know a lot about centrex but I know that most
PBX's support POTS lines (usually for faxing). You can
have them switch over the lines that they send you to
pots and then you can plug the lines in to a TDM400P.
Regards,
Dovid
--- Devin Heckman [EMAIL PROTECTED] wrote:
Hi,
I'm looking
I had exactly the same experience running IAX2, but also experienced
half-duplex calls on top of that (though I think that's a different but
with IAX handoff), and in the end dropped it completely for SIP.
We run g729 over dedicated fibre, and the resyncs were occurring all
over the place
Viktor Tatianin wrote:
Hi Steve
I attempt change in zapata.conf
cas=1-15:1101 but use zttool view ABCD bits 1010
Regards,
Viktor
Have you put the E1 in CAS mode with something like:
span=1,1,0,cas,hdb3
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing
key system, but I have seen several threads suggesting that this is
not a supported configuration
This bad boy might be what you need:
Hello, Mitchel!
At 07:41 AM 02/21/2006, you wrote:
I've been doing a lot of research into a decent server for Asterisk
but I seem to be running and circles and now I am turning to you. The
issue I have is it needs to be rack mountable (so a Dell SC430 isn't
going to work) and preferably have 3
Supermicro!
Cory J Andrews
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, February 21, 2006
Thank you for validating that I am not going mad!
I made some additional tweaks for today. We'll see how it goes. If not
well, then I'll try SIP for tomorrow.
Thanks,
Adam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Fern
Sent: Tuesday,
Hi. I'm having trouble controlling the user info when sending e-mails
from asterisk via sendmail to a Microsoft exchange server.
When I receive the email the sender is always
[EMAIL PROTECTED] and the name of the sender is always
Added by portage for asterisk. I want to change both
Hi everybody,
I'm facing a strange problem after upgrading Asterisk from 1.0.9 to
1.2.4.
Sometimes, when receiving an incoming call from pstn, although my sip
phones ring correctly (I've got both softphones and hardware phones),
noone can pick up the call. Asterisk CLI shows me that the phones are
Hi,
I'd like to know if and how can I set CallerIDNum for outgoing calls on a
PRI line with DDI.
Does anyone know if italian Telecom permit this?
Thanks
--
Domenico Viggiani
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
Dovid Bender wrote:
Again. What do you need ? Incoming and outgoing,
trunking etc. ?
I personaly use.
Voipjet.com
myPhonecompany.com
Teliax.com
I have heard others talk about:
JunctionNetworks
There others that are just not coming to mine. If I
remember them I will try to email them as well.
Ok I update from sourceforge the chan-capi-cm 0.6.4 with the good
capi.conf and now it's working fine.
I still not know why the old one (package from debian) are not working
with the old capi.conf ?
Thanks for your help ,
Cédric
Armin Schindler a écrit :
What version of chan_capi do you
I want to sniff all these info to test a sip ip phone talking to a asterisk
server. I have used tcpdump, but It just shows the
Ethereal would probably be a batter analyzer. Not sure how well it
seppurts sip, though. Unlike tcpdump it won't work on-the-fly. But you
can also get
Arne Morten Johansen wrote:
Hi. I'm having trouble controlling the user info when sending e-mails
from asterisk via sendmail to a Microsoft exchange server.
When I receive the email the sender is always
[EMAIL PROTECTED] and the name of the sender is always
Added by portage for asterisk. I
Hello,
I hope it's ok to post here for a job offer.
A dynamic IVR company has a current opportunity for a RD Jr developer.
The right candidate will have a background in developing and managing Linux
based software systems, some experience in the IVR industry is a huge plus.
Expertise in the
Hy,
I'm writing from Spain.
I have the 1.2.4 asterisk version and 1.2.3 zaptel version. I've heart
that this asterisk's version detects correctly de hang up of PSTN, but
in my case this thing doesn't happen.
Moreover, my asterisk sends the next messages in the CLI:
Feb 21 15:03:13
Same setup with two TDM400 (8FXO) running for over a year.
On Tue, 2006-02-21 at 01:37 +0100, Thomas Artner wrote:
Am Tuesday 21 February 2006 00:24 schrieb Marc Archer:
Hi All,
Can someone give me a definite answer as to wether or not you can
reliably run multiple TDM400P's in the
I believe that Centrex is ISDN correct?
Sean
On Tue, 2006-02-21 at 04:55 -0800, Dovid Bender wrote:
I do not know a lot about centrex but I know that most
PBX's support POTS lines (usually for faxing). You can
have them switch over the lines that they send you to
pots and then you can plug
Yeah I did change those. I'm using 1.0.8 (Or was it 9?).
It seams that the system overrides these settings?
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry Flanagan
Sendt: 21. februar 2006 14:54
Til: Asterisk Users Mailing List - Non-Commercial
Arne Morten Johansen wrote:
Yeah I did change those. I'm using 1.0.8 (Or was it 9?).
It seams that the system overrides these settings?
You may need to put the asterisk user into the trusted user list of
sendmail - by default sendmail will not allow users apart from trusted
one to change
Just one more question. In /etc/passwd there's a line with asterisk and
added by portage in it. Can I just change this without screwing up
everything? Or is there a command to change user info or something? As you can
see, I'm not so good in Linux.
-Opprinnelig melding-
Fra: [EMAIL
3 TDM cards here, I had artifacts if any of the cards were sharing
interrupts, the trick was to add the cards 1 at the time to get them
each on their own irq. The system isn't in production yet, so I don't
know how well it'll hold up under load, so far so good in testing
though.
9xFXO 1xFXS
Hi
guys,
just a question: can
i use the pppd application with a HFC PCI card using
bristuff.
Thanks for
all
Giordano
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options
Hi
Take a look this site:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Solaris+Support
roberto2006/2/20, Steve Kennedy [EMAIL PROTECTED]:
On Tue, Feb 21, 2006 at 12:17:43AM +1100, Mark Edwards wrote: At 06:33 AM 02/20/2006, you wrote:Please forgive the question, but what is the
Hi everyone,
our setup includes a NEC PBX connected to our asterisk via bri lines.
The NEC has a doorphone feature, which is just an extension that calls you when
someone rings. When connected to this extensions, a 'flash' signalling opens
the door.
So now, i'd like to trigger this from
Usually analog but can be IP as well. In Singapore, Singtel offers both
analog and IP centrex services.
Sean Cook wrote:
I believe that Centrex is ISDN correct?
Sean
On Tue, 2006-02-21 at 04:55 -0800, Dovid Bender wrote:
I do not know a lot about centrex but I know that most
PBX's
On Mon, 2006-02-20 at 17:04 +0100, Marc Patino Gómez wrote:
Hi,
Can you post your working config, I'm wasting my time to config h323-sip
Is working now :)
I'm using asterisk-oh323 0.7.3 on my asterisk 1.2.4 box.
I've to configure in oh323.conf with gatekeeper=DISABLED and the context
of
I had to add this same feature recently for a client that has centrix
lines and wanted to use the conference feature of the centrix lines
which requires a flash, here is the setup:
PSTEN CENTRIX LINES ADIT 600 FXO CARD ASTERISK ADIT 600
FXS CARD AVAYA PARTNER ACS R6.
When someone is on the
Hi Carlos , how do you did this part ? I also included a bit timeout of 120 seconds in the dial command.
Thanks in advanced.
Regards
Athiel2006/2/10, Carlos Chavez [EMAIL PROTECTED]:
On Fri, 2006-02-10 at 08:38 -0200, Darlon wrote:
Try to change the value of protocolvariant in
Thank you very much Darren.I did look at Dell's website for the info but was not able to find the PCI voltage info. Perhaps I looked at the wrong place or missed it. Googling also did not give me answers.I called Dell myself and the tech support person was very helpful. He confirmed that
Does anyone know of a way to specify what extension is dialed when 0 is
pressed in the voicemail system. I have a situation where there is more
than one secretary and they want the 0 to redirect to the appropriate
secretary for the two groups of people.
So an example would be:
555-1234 -
Steve Kennedy wrote:
On Tue, Feb 21, 2006 at 06:16:06AM -0500, Alexander Burke wrote:
Hello, Steve!
At 03:55 AM 02/21/2006, you wrote:
ztdummy was only used for timing. Linux 2.6 provides this function in
the kernel and I assume Solaris already has timing functions there.
Page
Arne Morten Johansen wrote:
Just one more question. In /etc/passwd there's a line with asterisk and added by
portage in it. Can I just change this without screwing up everything? Or is there a command
to change user info or something? As you can see, I'm not so good in Linux.
Yes,
Hi list
i use SVN branch , i have real time working good with IAX2
The problem i have is for sip_buddies , any SIP acount register does
not store ip addres inside the table.
This only for SIP iax2 works great.
i also have in sip.conf
rtupdate=yes
any ideas ?
--
Cheers Andrea
Andrea
Peter,
Diffrent companys offer diffrent services. For example
myPhoneCompany offers DID's for both inbound and
outbound. Thier basid DID plan is $5.00 with unlimited
incoming and 60 outgoing minutes. Each additional is
$0.029. Or $10.00 a month with 500 outgoing and the
same rates as above.
Ok,
Right now I have
disallow=all
allow=ulaw
allow=g723
Does it read it bottom up maybe?
On 2/16/06, yusuf [EMAIL PROTECTED] wrote:
Matt,
I you dont define a sip user/peer and just use a dial, asterisk will
automatically use the codec that it prefers, in my experince whenever i
dial SIP
JP,
There isn't much to show :)
Yes.. I am running the STUN server on the asterisk box so that VoIP
ATA's and phones behind firewall's can connect to the asterisk server
with no ports needing to be opened.
Setup is...
download stund.
unzip.. compile... run
WALA! Stun server :)
Then just put the
At 07:24 AM 02/21/2006, you wrote:
* Using Flash() in dialplan - doesn't work since channel is
Dial()-ed and doesn't allow applications at that very moment
* Typing *0 on phone = zap channel doc says this should send flash,
but doesn't seem to work in bridged scenarios (ZAP=*=ZAP or SIP=*=ZAP)
When incoming DATA calls arrive on ISDN, Asterisk recognise that this is
DATA call, but behaving like it is voice call: Answering call, playing
IVR messages, etc...
How to stop that? I want that only VOICE calls are answered by Asterisk,
and DATA/FAX to be ignored.
(I'm using Asterisk 1.2.1
On Tue, 2006-02-21 at 09:52 -0600, acriollo wrote:
Hi Carlos , how do you did this part ? I also included a bit timeout of 120 seconds in the dial command.
Thanks in advanced.
Regards
Athiel
It should say a BIG timeout, not a BIT, sorry. Just do a
I have been struggling with this issue for about a year now. There were
just too many IP phones to choose from at all sorts of price points and not
enough information about any of them. Now I am looking at the situation
again and if anything it has gotten worse. There are even more phones
FYI...
Just installed one of the new Sangoma A200D analog pstn cards with the
hardware echo canceller on a trial basis. The card has four fxo interfaces.
Excellent audio quality, excellent echo cancelling, and excellent audio
levels.
The four pstn lines at this location are rather long analog
This is also very dependent on where you are and who your ISP is...
I used Teliax and there setup instructions and support are excellent.
Unfortunately for me, my ISP (frickin comcast) has a very poor route
to Teliax's servers. This seems to be somewhat changeable, but is
consistently
Macro(SIP/700-8d41, record-enable|700|OUT) in new stack
-- Executing GotoIf(SIP/700-8d41, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(SIP/700-8d41,
recordingcheck|20060221-113809|1140539889.439) in new stack
-- Launched AGI Script /var/lib/asterisk/agi
ITSP seeking C programmer to work on Asterisk and SER.
[EMAIL PROTECTED]
Located in Northern NJ
Sorry if I should not post this here
Doug
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE
Greetings to all.
I am currently implementing call queues for a customer and have come across
several problems.
The customer is an airline representative, and will be using call queues for
different airline reservations. The customer requires that any agent be able
to login to any number of
What's the benefit of using stund vs nat=yes in your sip.conf for that
device? I haven't had any issues behind firewalls when I enable that
option, and no ports are needed to be opened.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent:
Doug Lytle wrote:
Doug Lytle wrote:
[EMAIL PROTECTED] wrote:
I put a Tellabs 64ms echo canceller into my facility this weekend and
am praying that it removes are echo problem. If it does, I plan on
making it a standard on my Asterisk installs that have a channel bank
or T1.
Well, the
[Mr.] Mustard,
There's no one-stop IP phone review site that I know of (that has one
person/company comparing all of the IP phones side by side).
You're right, the gxp-2000 is a little on the low end as IP phones go.
However, you're also getting a lot of features for your buck with the GXP.
I
Speaking of this book, where can I get it? Obviously I can read the
pdf, but I lack the facility to print it in any usable fashion. The
labor and materials that I have spent on trying to print it thus far
probably outweighs the cost of the silly thing. Is it only available
online, or do you
Hello all,
How to tape letters in xlite softphone, when using the
Directory application (or generally when is needed).
Thank you.
--
Bayrouni
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
Hi
Are there any packages to implement vertical services in asterisk
commercial (or free)
Thanks
Patrick
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
I hear some
people praising the GXP2000 phones and I gotta wonder what
they are smokin (regardless of firmware revison) so I just
don't know who to believe anymore.
As one of those who's praised the GXP2000, I feel I should just add that
it's all relative *to the price point*. The GXP2000
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nate List
Sent: Tuesday, February 21, 2006 7:17 PM
...
In my Outbound Routing I have the Trunk Sequence set up so that 0 is
Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk
Sequence 0 is
I put a Tellabs 64ms echo canceller into my facility this weekend and
am praying that it removes are echo problem. If it does, I plan on
making it a standard on my Asterisk installs that have a channel bank
or T1.
Well, the day is almost over here and not one echo reported
Dear all,
Can somebody share his experience with me in using fedora core 3 as asterisk
server using quad port card (e1/pri) at full capacity.
goksie
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
For every person that says phone x is great there is
someone else complaining about it.
Its very simple why there are soo many answers to the what phone to use
question. The answer really comes down to a matter of personal preferance
and end-users needs. Mind you, some phones are better
Hello, Bob!
At 01:32 PM 02/21/2006, you wrote:
Speaking of this book, where can I get it? Obviously I can read the
pdf, but I lack the facility to print it in any usable fashion. The
labor and materials that I have spent on trying to print it thus far
probably outweighs the cost of the silly
Can somebody share his experience with me in using fedora core 3 as asterisk
server using quad port card (e1/pri) at full capacity.
Runs fine and is very stable. Full capacity is 100% dependent on exactly
what asterisk is doing (eg, transcoding), the PC hardware, etc, and has
nothing to do
I agree with most of Raymond's other points, but I have to take issue with
this one:
1) If it doesn't support PoE I won't implement it. Support
phones with wall-warts or bricks is just a added hassle and
adds TCO as most end up being replaced once or twice during
the lifetime of the phone
I have a server in my lab running asterisk (v1.2.1)
and ztdummy. (No zaptel hardware is present in the
server).
I have to free up this server to be used for a
completely different application.
What is the best step-by-step procedure to permanently
remove/uninstall asterisk, asterisk-addons,
What's the benefit of using stund vs nat=yes in your sip.conf
for that device? I haven't had any issues behind firewalls
when I enable that option, and no ports are needed to be opened.
For some strange reason, even with nat=yes sometimes when a user's IP
changes, the phone doesn't realise
PS A central resource of various Voip terminators and the
quality of routes to/from various ISP's would be a great
boon. Is there such a thing?
When we've added asterisk servers (in datacentres) to our collection, one of
the things I've always asked the datacentre to provide is a traceroute
My understanding is nat=yes tells asterisk the device is behind a nat
(and works even if it isn't) but stun actually keeps stuff open in the
person's local firewall.
On 2/21/06, Bill Gibbs [EMAIL PROTECTED] wrote:
What's the benefit of using stund vs nat=yes in your sip.conf for that
device? I
Mimmus, It looks like this took care of the problem.
Thanks for your help,
Nate
Mimmus wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
Nate List
Sent: Tuesday, February 21, 2006 7:17 PM
...
In my Outbound Routing I have the Trunk
On Tue, 21 Feb 2006, mustardman29 wrote:
I hear some people praising the GXP2000 phones and I gotta wonder
what they are smokin (regardless of firmware revison) so I just don't know
who to believe anymore.
The GXP2000 is probably the best phone you can buy _for under $100_.
Got it? Under
1 - 100 of 156 matches
Mail list logo
