On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Mar 3, 2006, at 11:35 AM, Tom Vile wrote:
I have had the same issue with Sipura as well, they gave me quite the
run around on 1 bad 2002 ATA and decided to just forget about it and
bought a new one to save time.
So they gave you shitty
I realized today that my call waiting isn't working properly. If I am
using the FXS attached phone and a call comes in the FXO, it just goes
directly to voicemail, with no indication (call waiting beep).
If I flash there is a second dial tone, and I can initiate a second
call.
If I am
On Mar 2, 2006, at 3:46 PM, Wai Wu wrote:
You can really mix G729 encoded frames. So I would guess that licenses
are not needed for non-G279 devices. BTW, there is a difference
conference app (forgot the name) that only mixes the two parties that
have the loudest volumn. It sounds more
On Mar 2, 2006, at 12:32 PM, Anton Krall wrote:
Looks very nice.. Is it GPL, GNU?
Maybe if you trimmed you posts and pasted relevant quotes, we could
have some idea what this question means...
___
--Bandwidth and Colocation provided by
On Mar 2, 2006, at 9:46 AM, Matt wrote:
Hi,
Occassionally Asterisk will go down and I have to restart it.. not
often.. but sometimes. When it does the manager interface stops
working, as does the CLI.
My thoughts was to poll the manager interface once every 5 minutes for
a value. If I don't
On Mar 1, 2006, at 1:46 AM, [EMAIL PROTECTED] wrote:
On Tue, 28 Feb 2006, Martin Joseph wrote:
On Feb 28, 2006, at 2:35 PM, [EMAIL PROTECTED] wrote:
On Tue, 28 Feb 2006, Cory Andrews wrote:
Here is a link to some additional resources which may be helpful in
configuring the 5801 and other
On Mar 1, 2006, at 5:20 PM, kevin ling wrote:
Hi,
I have another model 3702a (2FXS 2FXO) voice gateway. You can
implement
one-stage dialing on this device.
1. using sip show peers to make sure two ports (1fxo/1fxs) was
registered
to asterisk.
2. login 3701 web and change the defualt
On Mar 1, 2006, at 3:03 PM, [EMAIL PROTECTED] wrote:
On Wed, 1 Mar 2006, Arsen Chaloyan wrote:
The inbound PSTN DTMF works excellently, e.g. people
calling from PSTN
into the * box are able to pick IVR items with DTMF
reliably.
exactly!
There are some other problems with DTMF and spa3000.
On Feb 28, 2006, at 1:22 AM, [EMAIL PROTECTED] wrote:
I am having problems with a Zoom 5801 and *.
It does not appear possible to route voip calls out the FXO, all voip
calls get routed to the FXS no matter what.snip
If there is a routing function of some kind on the modem setup,
perhaps
On Feb 28, 2006, at 6:50 AM, Ash Thakrar wrote:
Hi Mark,
Thanks for your reply.
For the phase you have indicated the time it took was immediate, no
delays
there.
I have seen on the list several discussions of how additional delay on
ringing can be due to Asterisk trying to get caller ID
On Feb 28, 2006, at 7:14 PM, Anton Krall wrote:
Anyway the phone can compensate? I don't think it works that way but
worth
asking..
If the phone has an input gain (for phone users voice) then adjusting
it down can help echo that is being generated at the far end. ie if
it's too loud
On Feb 28, 2006, at 6:43 PM, James Harper wrote:
What about the fonebridge (http://www.red-fone.com/fonebridge.html)?
It uses POE, so you could hack something together to supply 48V @ 15W
if
you don't have access to a power point, and it appears to have a 2 port
switch built in so you could
On Feb 28, 2006, at 2:35 PM, [EMAIL PROTECTED] wrote:
On Tue, 28 Feb 2006, Cory Andrews wrote:
Here is a link to some additional resources which may be helpful in
configuring the 5801 and other Zoom products
http://www.zoom.com/salessupport/index.php?pd=ATA_Documentation/
I just found out,
On Feb 28, 2006, at 9:51 PM, Andres wrote:
Ed Greenberg wrote:
I need to set up an office full of Cisco 7960 phones behind NAT with
the server out in Colo.
The first test phone registers fine, but the second one does not
register.
The first phone's registration looks like so:
Short version:
Flash device with latest SIP firmware (currently 1.04)
Set Network (I am using the LAN port only) and SIP config as
expected.
Set Line configuration so that the FXO is hotline to the asterisk
extension you want to ring with incoming PSTN calls (mine is set to
2020).
Set System
On Feb 27, 2006, at 6:09 AM, Dr. Michael J. Chudobiak wrote:
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always
works fine for an
On Feb 26, 2006, at 8:03 PM, Joseph Blake wrote:
We're setting up asterisk at the office (really doing some testing
right now) and it is going to be hosted on a dual G5 XServe running OS
X.
I love it. Glad to hear it. Should be a monster.
We're an apple certified solutions provider, etc.
On Feb 27, 2006, at 2:55 AM, Dr. Michael J. Chudobiak wrote:
Can someone recommend an IAX provider for US DIDs who will:
snip
3) Have great audio quality
This is somewhat a meaningless question, as the route from you to the
call terminating service can make or break the quality.
You
On Feb 26, 2006, at 12:57 AM, Alexander Burke wrote:
Hello, list!
After Googling and checking out the voip-info wiki, I haven't had much
luck in locating a decent web-based voicemail system for Asterisk to
check your VM while you're away from the office without using a phone.
Can anyone
On Feb 26, 2006, at 5:37 AM, Wooi Koay wrote:
I have a POTS and a sip incoming into my asterisk server. When I call
the POTS number from outside (cell or landline) and trying to
authenticate myself when enter #, 8 out of 10 times I got an
authentication incorrect. If I call in to the sip
On Feb 26, 2006, at 5:43 PM, hugolivude wrote:
Say, thanks to all you for your time in responding.
I hope I don't sound unappreciative (I have no time for flamers) but
I don't understand how changing from SIP to IAX would make any
difference. I don't have any problems with the signalling
On Feb 26, 2006, at 11:59 AM, Paul wrote:
snip
Unless you have good QOS routing be sure that mail server is somewhere
where you don't have voip phones. I had a mail server at an office with
400k sdsl. I would be on a call and let an incoming call go to voice
mail. The incoming email with wav
Hi again,
Kind of sheepish about asking for help, as I have only spent a day
banging my head off this...
I got my new Welltech 3701a, 1FXS,1FXO gateway.
I flashed it with what is seemingly the appropriate firmware (SIP
V1.04). This seems to have gone ok, and it is now registering both
On Feb 25, 2006, at 10:04 PM, Anton Krall wrote:
Nice!
Did you ever think about trimming your messages?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
On Feb 23, 2006, at 4:58 AM, Adam Robins wrote:
Thanks,
We already have a cron reboot of all of our Asterisk servers every
night. We've been doing this for over a year due to memory leak
issues.
??? What do you think this is windows 95??? I had a problem like that I
would be looking at
On Feb 23, 2006, at 6:11 AM, Tele Cost Price Reducer wrote:
hi ,
i have some options we are working with at vast deployement with no
problems:
www.tigernetcom.com - type 102 is a nice ATA, like GS 486 but far away
better.
we import directly from the producer at great prices so if anybody
On Feb 22, 2006, at 10:24 AM, Rusty Dekema wrote:
On 2/22/06, Matt [EMAIL PROTECTED] wrote:
Yes.. there are provisioning tools that you have to get.
Unfortunately it's this catch 22 loop. You have to prove that you can
offer 200+ ATAs to customers, or you can't get the tools, but yet, you
On Feb 22, 2006, at 3:44 AM, Jean-Marc Salsa wrote:
Thanks,
But, I do not have phones connected to Asterisk ...
but only one peer : my softswitch ...
So call flow is Phone - Softswitch - Asterisk - Voicemail
I can force the link Sofswitch - Asterisk ( Codec and DMTF Mode )
Codec is PCMx
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
On Feb 19, 2006, at 6:06 AM, Phil Blundell wrote:
Overall, I'm happier with the SPAs than the handytones, though neither
of them are entirely perfect. Oh well.
Thanks for the update...
I am being told by the freaks at Grandstream that there will be a
firmware update forthcoming to try to
On Feb 19, 2006, at 9:41 AM, Dovid Bender wrote:
Some people have to stap on others to make them selves
feel good. Very unfortunate.
Some people have no sense of humor. Very unfortunate.
___
--Bandwidth and Colocation provided by Easynews.com --
On Feb 17, 2006, at 6:36 AM, Robert Webb wrote:
Sorry, this is off topic to asterisk itself, but is about the list
server.
I had a power failure lastnight at home, where my email server
resides, and my network was down for about 20 minutes, that was after
45 minutes of uptime on UPS.
On Feb 16, 2006, at 1:11 PM, Adolfo R. Brandes wrote:
turby wrote:
is there recomended source files for t.38 pass? latest cvs does not
work for me.
is it possible publish working src?
You mean T.38 passthrough? I've just uploaded an asterisk-1.2.4
backport of the lastest svn
On Feb 16, 2006, at 3:40 PM, [EMAIL PROTECTED] wrote:
http://www.zoomtel.com/products/voip_products.html
anyone using these? they look very interesting in that they support
ilbc, and they offer a separate cheaper model without g729 license.
i'm wondering if their EC is better than the
On Feb 13, 2006, at 10:20 AM, Eric ManxPower Wieling wrote:
snip
The nearest CO my POTS line goes to is 11 miles away.
snip
i take it you aren't a DSL customer?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
On Feb 13, 2006, at 2:45 AM, Simone Cittadini wrote:
C F ha scritto:
Am I the only one having trouble with this list?
Since the begining of the week I have not been receiving mail from the
list like I used to, is this a gmail problem? or is it subscription
problem? or is something wrong with
On Feb 12, 2006, at 1:05 AM, Nitin Gupta wrote:
Sorry for re-posting this message -
I am trying to run the latest stable Asterix version 1.2.4. on 64 bit
amd procesor.
Things are working but the playback sounds that I hear when tring to
connect over IAX are of very high frequency.
i.e a
I am wondering if the instructions for hard wiring a Tellabs canceler
are applicable to a regular old two wire loop?
Or is this only something that works for people with T1?
Any comments from people that have tried this are appreciated.
___
On Feb 8, 2006, at 9:22 AM, Ariel Batista wrote:
I normally don't like talking bad about products. But I would like to
say that the Welltech/Wellgate are not products that are support to
work with asterisk. I have invested many hours of work in getting
there device to work with Asterisk.
On Feb 6, 2006, at 5:08 AM, ammar Ali wrote:
Jose,
There are No open source IP phones, I was only joking, I assumed you
should know what an open source is.
The AG-168V is an open sourced ATA. Although the idea that Walmart
would give something (useful) away for free, was funny to me.
Any feedback on this brand and in particular on doing business with
WelltechUSA?
I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I
am hoping to replace the near worthless Grandstream HT-488.
This company is telling me that I need to wire $ directly into there
bank
Why?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
On Feb 1, 2006, at 4:24 AM, Olle E Johansson wrote:
Cosmin Prund wrote:
As the subject line says: Is PING a good indicator of network
latency? If
not, how can I measure latency?
Using Asterisk is a good way. If you define a phone in sip.conf and
turn on qualify=, we will measure the latency
I have a strange problem with echo.
My setup includes a Grandstream HT-488 which is both an FXO and a FXS.
I noticed last evening that if I called the FXS through my asterisk box
from my cell, the resulting connection was fine for me at the cell end,
but produced dramatic and conversation
On Jan 31, 2006, at 1:19 PM, Brent Torrenga wrote:
Has anyone had problems getting their preffered codecs on the Teliax
web
interface taking effect?
I have two accounts, two separate yet similarly configured * servers.
On one
account the settings took right away - on another server I am
On Jan 31, 2006, at 12:08 PM, Fran Sedano wrote:
Hi!
No one can help me with this??
x-tad-smaller- Original Message -/x-tad-smallerx-tad-smallerFrom:/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerFran Sedano/x-tad-smallerx-tad-smaller
On Jan 29, 2006, at 11:24 AM, Rich Adamson wrote:
Anyone tried to muck around with using the 488 for both fxs and fxo
with asterisk?
I've been playing with one for the last couple of days, and it looks
like its a little lower quality then the spa3k. No gain settings,
echo
canceller is
On Jan 29, 2006, at 10:30 AM, Warren Burstein wrote:
I took a look at the asterisk-1.2.3 Makefile, seems to me that the
WARNING is just a list of all the .so files found in the modules
directory that aren't also found in a subdirectory, it isn't checking
that they were built with the current
On Jan 29, 2006, at 1:24 PM, Michiel van Baak wrote:
On 13:09, Sun 29 Jan 06, Martin Joseph wrote:
I removed the following to get it starting up again:
app_enumlookup.so
app_groupcount.so
app_md5.so
app_txtcidname.so
func_cut.so
Both the README and the UPGRADE listed that those functions
Hey Gurus,
I have a very simple asterisk setup that basically lets me share a PSTN
line from one location to another. I would like to have the phones at
both locations ring when the PSTN # is dialed(inbound calls from PSTN
to asterisk).
I tried something like:
exten =
On Jan 28, 2006, at 12:54 AM, Ronald Wiplinger wrote:
Martin Joseph wrote:
snipI tried something like:
exten = 2020,2,Dial(SIP/2005,25,trIAX/2010,25,tr)
I thought this might cause both 2005 and 2010 to ring when 2020 was
dialed, but only 2005 rings?
Below works for me:
PHONE_LOCAL
On Jan 28, 2006, at 6:50 AM, Mark Adams wrote:
x-tad-smallerHi Everyone,/x-tad-smallerx-tad-smallerI know this may be off subject but I am not sure who to ask. I am currently looking for voip termination that is closest to replicating U.S. pots service. I run I.V.R. systems and I want to point
On Jan 28, 2006, at 7:15 PM, Vic wrote:
Hi, Zoa,
yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them.
We will also need an IVR function as well.
I am not up to speed on Asterisk yet, so, I am a
On Jan 28, 2006, at 2:13 PM, Rene Kluwen wrote:
Is somebody here using a RoadRunner/Time Warner connection and able to
successfully with SIP (or IAX2)?
We are experiencing high latency up to the point that the voice
conversation
is not understandable anymore. This goes for both SIP and
On Dec 21, 2005, at 10:26 AM, [EMAIL PROTECTED] wrote:
Regards to All,
I recently setup an Asterisk system ([EMAIL PROTECTED]) and it works like a
charm so
far. It is in a SOHO behind another Linux iptable NAT firewall with no
problems.
Hopefully this isn't too dumb a question, and its the
On Dec 21, 2005, at 2:31 PM, Matt Roth wrote:
List users,
I have some additional information related to the dropped audio.
Huh, I have noticed this type of popping on an SIP to SIP connection
using ulaw also, but I figured it was just me. I am running * 1.21.
I am kind of a newbie to
On Dec 22, 2005, at 5:58 AM, Thomas Miller wrote:
Everything was working great until last night. All
calls since last night are getting No Authority
Found message. I am using IAX2
Is anybody else having this problem?
I had two days of outage with them last week, but I got CHANUNAVAIL
Marty
On Dec 20, 2005, at 1:01 PM, Rhonda Herron wrote:
Hi,
I have been battling with the following problems for a while and was
hoping someone could shed some light on the subject.
I am using AT320 402 IAX2 phones with 1.49 firmware (latest) connected
to an Asterisk server running [EMAIL
On Dec 8, 2005, at 3:27 AM, Andrea Riela wrote:
snipWith g711 all works like a charm, but for audio quality, and
bandwidth utilization, I'm trying now to work with g729 between CME
and ISP. What about Asterisk? this is a pass-thru example, or maybe
I've to pay a g729 license?
Yes, you
On Nov 28, 2005, at 3:55 PM, BJ Weschke wrote:
On 11/28/05, Martin Joseph [EMAIL PROTECTED] wrote:
snipI am only able to get comedian voicemail (ie dialing 1234) to
record or
playback messages if I use the GSM codec? Is this normal and
expected?
If I use ulaw or alaw I get either trash
On Nov 29, 2005, at 9:27 AM, Mojo with Horan Company, LLC wrote:
What's the 'format' line of the [general] section of your
voicemail.conf?
It's format=wav49|gsm|wav
You should try not to just tack one line on top of a long message to
list... ;~)
snip
Marty
On Nov 29, 2005, at 12:25 PM, Michaël Gaudette wrote:
Hi,
I`m a beginning Asterisk and Sendmail user. I am trying to setup my
voicemail to send emails to a certain email address. It doesn't work,
and I
think I've figured out what it is. There is probably a spam-feature
at my
provider
Hi,
I am a newbie, and I am setting up a simple system to share a PSTN
line with another location.
In the process of setting this up I am also testing the various codecs.
I am only able to get comedian voicemail (ie dialing 1234) to record or
playback messages if I use the GSM codec? Is
checked my username and password with the sip.conf file
(secret=...) and it looks ok. You can find the messages below.
Any ideas?
Martin
The trace:
My UA sends the INVITE to Asterisk:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK-276428-27856
From: 4302
On Nov 23, 2005, at 1:10 PM, Denis Vella wrote:
Hi,
I'm trying to use modems with Asterisk+VoIP Gateways in an attempt at providing an Internet service.
Home_PC-->Modem-->PSTN-->VoIP_Gateway_FXO-->Ethernet-->Asterisk-->Ethernet-->VoIP_Gateway_FXS-->Modem-->PPP_Server-->Internet
I've been
On Nov 23, 2005, at 11:14 AM, Michael wrote:
I am trying to route my calls through an outside IAX provider. I am
having a problem with which codec to use. The only way I have
successfully been able to make an outgoing call is if i do:
disallow=all
allow=g729
in the sip.conf file
On Nov 22, 2005, at 9:09 AM, Joash Herbrink wrote:
My p2000w Works with asterisk.
Here is the sip.conf entry
[1006]
type= friend
subscribecontext = all-local
accountcode = 1006
amaflags= default
username= 1006
secret = whatever
host
On Nov 22, 2005, at 11:08 AM, Doug Meredith wrote:
hugolivude [EMAIL PROTECTED] wrote:
You need to be
careful when buying the Linksys because version 5.0 saw a move from
Linux, which runs Sveasoft's Talisman firmware, to VxWorks, which does
not.
Why would I care what OS an embedded device
+video
states:
Video
in Asterisk
Some channels have support for video calls in Asterisk
SIP
IAX2
H.323
I have managed to verify a
reasonable level of video support in SIP but have failed to find any evidence
in oh323 or ooh323.
Many
thanks, John
John Martin
On Nov 17, 2005, at 9:22 PM, Henry Junior wrote:
Has anyone compiled 1.2 on OS X? If so, do all the realtime
components compile properly? Thanks, HJ
I built it under 10.4.3 using the xcode that came on my 10.4 DVD (gcc
4).
What exactly do you mean by the realtime components?
Marty
On Nov 8, 2005, at 10:39 AM, Bill Michaelson wrote:
Is anyone using a Grandstream ATA-488 FXO port to connect a PSTN trunk
to their Asterisk box (via SIP, of course)?
Is it possible to have such a beast operate reasonably?
If so, is it also possible to use the FXS port concurrently and
On Nov 17, 2005, at 9:02 AM, Kevin P. Fleming wrote:
Doug Meredith wrote:
Just a configuration management note. The normal (and safe) practice
would be to make the second copy 1.2.1. Once 1.2.0 has been released,
you can't change it. It is done. Calling the second copy 1.2.1 would
have
Hi
All,
Can
anyone recommend a version of mISDN and mISDNuser (dates of CVS or archive held
on someones server) that will work with the chan_isdn in Asterisk 1.2.
Many
thanks.
John
www.AuPix.com
___
--Bandwidth and Colocation
any negotiation.
So I think about using T.38, which is supported by SN1400, but not by
Asterisk. Is there any T.38 channel implementation for Asterisk 1.2.0?
I've found only some info at http://www.ionidea.ua/oss/asterisk/, but
it's for Asterisk 1.0.7 only.
--
Regards,
Martin Edlman
Fortech
make uninstall?
Matteo Piazza wrote:
Is there a command to remove completely asterisk?
I want clean the server before the installation of 1.2 version.
Matteo
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
i would recomend this channel for h323:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Abdul Lateef wrote:
Hi all,
I have H.323 Terminator and i want to terminate our
all SIP clients to this terminator, Is it possible to
add H.323 Terminator in Asterisk?
Please give me a little hint
On Nov 9, 2005, at 6:58 AM, Michaël Gaudette wrote:
Hi,
I'm trying to send some DTMF dialtones (for an extension on the other
end).
My call is done from a Zap channel, to Asterisk, throught an IAX
provider,
to a PSTN line in some university.
The phone number I am trying to reach is
i think, TDMoE is not supported/developed anymore. This is known bug.
Franz Wu wrote:
Hi all
my system 1:
celeron 1.2GHz + intel 810e (asus TUW-LA) + 256MB SDRAM
onboard vga (intel 810e chipset)
RTL8100 NIC
debian sarge 3.1r0a / kernel 2.6.8-2-686
asterisk / libpri / zaptel from CVS HEAD @
hi, did you solve this problem, which i exactly have?
lokotes wrote:
Hi,
Background:
I'm running 2x * boxes.
Box A has a registered user which dials a number. The connection is
sent to Box B which acts as pstn gateway (sangoma 1xE1 card).
Problem:
On Box A before executing Dial() command I
TDMoE is useless. I've tested it on newer intel P4 machines with 2.4 and
2.6 kernels. There is CPU peaks causing by TMDoE driver.
If you want pass modem data, try IAX u/alaw codec. In my environment it
works great (switched lan)
trixter aka Bret McDanel wrote:
On Wed, 2005-10-19 at 10:43
I am a newbie and want to step up to VoIP and switch from analog
connetion to my Astrisk/Lineox box.
Any suggestions on configuring Vontage and what to get/ask
when signing up?
Has anyone experienced problems with Vontage and Asterisk. I'm using
Asterisk (Current Stable) and Sipura-841
Hi,
I seem to be unable to get Asterisk to recognise the '#' key being
pressed to acknowledge an incoming call from a queue. No matter how many
times I press the key to acknowledge, the Asterisk server acts as if I
have not.
I have installed the ztdummy module, and it seems that Asterisk is
Tzafrir Cohen wrote:
On Sat, Oct 15, 2005 at 03:21:38PM +0100, Colin Martin wrote:
Hi,
I seem to be unable to get Asterisk to recognise the '#' key being
pressed to acknowledge an incoming call from a queue. No matter how many
times I press the key to acknowledge, the Asterisk server acts
,
Martin
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
be more specific on what to look at?
Thanks for your help.
Martin
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
I am trying to compile the astrisk-1.0.9 tarball on a RedHat 9 linux box
with dev environment. I get a lot of the following as a result of a make
/usr/bin/ld /usr/lib/crtn.o: invalid string offset 10 for section
`.shstrtab'
and final show stopper
./gentone busy 480 620
make[1]:***[busy.h]
.
Martin
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman
ooh323c
installeed but do not know how to configure :(
maybe googling or reading README can help
woomera
let me know if there's any one who has tried this.
i've been testing this. It can do only alaw/ulaw and this is unusable
for me. It works, but i've got some segfaults (using gnugk
Kanishka Somaratne wrote:
hi
has any one used OOH323C i tried this it is installed but do not know
how to configure has any one used this, what is the best h323 addon to
use with asterisk
OOH323 has no jitterbuffer and does not work with cisco gw (incoming
calls with g729). OH323 (latest
Brian C. Fertig wrote:
yes.. I have looked. they are different. But when I unregister 1 the other will register..
Its only when I have 2 of them trying to register at the same time I have an
issue. But yes
the ID's are different in both of them.
maybe you have the same aliases
have you done
modprobe zaptel
before the
modprobe wcfxs
??
On Friday 23 September 2005 12:21, somesh s wrote:
Hi,
Your card uses 'wctdm' or 'wcfxs' depending on what
version of asterisk you're using
Can you explain about what do you mean by this?
I tried modprobe wctdm also same
but haven't heard anything back.
Regards...Martin
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
asterisk and
tele-comms is new to me although im experienced with linux sys-admin,
networking etc.
Many thanks
Martin
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
]: Sending to 192.168.1.100 : 5060 (non-NAT)
Sep 9 11:47:36 VERBOSE[2444]: Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK289a5fe76
From: Martin sip:[EMAIL PROTECTED]:5060;tag=d6d383eca9b6910
To: Martin sip:[EMAIL PROTECTED]:5060;tag=as3c7c47f1
Call-ID
and easily one for
each device.
Regards...Martin
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
Hello.
I couldn't see anything in festival.conf.
The voice is currently male, and robotic.
How do I change this ?
Regards...Martin
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users
This is a single aastra 9113i sip phone.
asterisk 1.0.9
Why do I keep seeing this in the logs ?
--
Sep 8 18:44:25 VERBOSE[18779]: Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000 ms
Sep 8 18:44:31 DEBUG[18779]: Setting NAT on RTP
On Thursday 08 September 2005 19:05, Martin wrote:
This is a single aastra 9113i sip phone.
asterisk 1.0.9
Why do I keep seeing this in the logs ?
--
Sep 8 18:44:25 VERBOSE[18779]: Scheduling destruction of call
'[EMAIL PROTECTED
Address lookup
canonical name asterisk.org.
aliases
addresses 216.27.40.102
Service scan
FTP - 21Error: TimedOut
SMTP - 25 Error: ConnectionRefused
HTTP - 80 Error: ConnectionRefused
POP3 - 110 Error: TimedOut
NNTP - 119 Error: TimedOut
digium.com
...Martin
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman
901 - 1000 of 1799 matches
Mail list logo
