On Tue, 2005-03-15 at 20:09 +0100, Andreas Meyer wrote:
Sorry for not being clear enough but my headphone is attached to the
soundcard at my local PC. Now when I start Asterisk on that machine it
is using port 5060 and sjphone can not connect because it also uses port
5060.
netstat -panu
On Tue, 2005-03-15 at 16:05 -0700, Zanzamar Majere wrote:
Is anyone else having issues pulling up voip-info.org?
There's been a 'wiki down' thread running all day on this list. So it's
been noticed, yes.
Regards, Bruno.
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On Wed, 2005-03-16 at 13:13 +0100, Christian Schoepplein wrote:
Hello!
I'm new to asterisk and because I try to configure the package for my
needs the last days without success, I'd like to ask a basical qestion.
I need asterisk to work together with the German VoIP provider sipgate
On Fri, 2005-03-18 at 20:33 +0100, Stefan Stolz wrote:
Hello,
i tried to configure Gnomemeeting for Asterisk, because its, how it looks,
the
only tool which gifes me all i want for the use in linux...
I have allready installed and running h323 support in asterisk and edited the
On Fri, 2005-03-18 at 22:02 +0100, Bruno Hertz wrote:
To receive calls with GM, you have to add a line like
exten = yourexten,1,Dial(OH323/yourip:1720)
to the context which handles your incoming calls.
Correction:
exten = yourexten,1,Dial(H323/yourip:1720)
It's because I use OH323
Kris Edwards [EMAIL PROTECTED] writes:
I've seen some posts about ppl using gnomemeeting via oh323, but is
anyone using it w/ sip?? (only their cvs supports sip, but I figured
somebody was trying it.. I'm grabbing it now :)
I tried to get GM/Opal going some six weeks ago but it didn't even
Jay Ray [EMAIL PROTECTED] writes:
Thx manI will try to start it from withing DDDNo one responded in DEV
list
No one answered because your question was way too dumb (sorry).
If you attach with a debugger to a running process, the process
will be stopped. You then have control of it
Kris Edwards [EMAIL PROTECTED] writes:
This is the best linux sip phone I've used so far. Audio quality has
been perfect and it seems really stable, so hopefully it will be out of
beta soon.
I might actually pay for the full version! (not counting console games,
that would be the second
hank smith [EMAIL PROTECTED] writes:
do you know if it is gtk2?
It appears to be:
$ ldd xlite-linux-22
... blah ...
libgtk-x11-2.0.so.0 = /usr/lib/libgtk-x11-2.0.so.0
... blah ...
Regards, Bruno.
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Kris Edwards [EMAIL PROTECTED] writes:
Well, I'm certainly not selling xten.. Perhaps my enthusiasm extends
from my disgust with everything else. In particular, kphone, and
sjphone. I have noticed latency with xten in meetme, but if I just dial
somebody it works better than anything I've
Henry Devito [EMAIL PROTECTED] writes:
Forget this post I had a typo in my voicemail.conf file
sendvoicemail=yes was spelled wrong.
That fixes point 1) What about the others?
- Original Message -
From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Dana Olson [EMAIL PROTECTED] writes:
I've been meaning to try it again. A large number of builds have been
sent since I last tried. And boy, it was sooo slow and more
resource-intensive than its Windows counterpart.
Maybe, but I still recommend trying again. It's really making headway.
I
Andrew Kohlsmith [EMAIL PROTECTED] writes:
Call it archaic if you like but mailing lists get the job done faster, better
and without all the bullshit that forums bring to the table.
It's not archaic but reasonable. Clicking around in a funky web interface is
definitely not what I call
Dana Olson [EMAIL PROTECTED] writes:
What's wrong with using your keyboard's Num pad?
Nothing. Tried that, didn't work. Build 1.30.256b
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Martijn van Oosterhout [EMAIL PROTECTED] writes:
Ok, basic use case. I today go to a forum and read all the messages.
Next day I come along, how do I get a list of all the messages I havn't
read in thread order in such a way that if I decide to go somewhere
in the meantime, it knows what
Brian Capouch [EMAIL PROTECTED] writes:
Hmmm. I just got the latest beta build, which identifies itself as 1105d.
The keypad functionality is perfect.
Hmmm. Good for you. We were talking about sjphone, though :)
Regards, Bruno.
___
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Francesco Peeters [EMAIL PROTECTED] writes:
On the other hand imagine a forum with subtopics like sipura, softphones,
zap or whatever. Wouldn't that maybe help to put some load off at least
the casual reader and poster seeking or giving advice for topics he/she
specialized in, and maybe even
Tim Bass [EMAIL PROTECTED] writes:
the excellent movie Vanilla Sky)...
Ahem. . . .
B#2.
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[EMAIL PROTECTED] (Tony Mountifield) writes:
I totally agree. I run a local INN server and all the mailing lists I
subscribe to get turned locally into newsgroup postings in moderated
groups. When I make a posting, it gets mailed out through a filter to the
moderator address, which is just
Francesco Peeters [EMAIL PROTECTED] writes:
I think you took my Nah a itsy bit out of context there... ;-)
Hehe, I guess context is what your neurons link to - which, as you look at
them, might account for the itsyness :)
Totally OT:
I have been looking at this as a plugin for my own (non
Dana Olson [EMAIL PROTECTED] writes:
I'm pretty sure that I used SJphone to check my VM. I'll test again.
But there is a new beta out on their site (and it's newer than the
Windows build). Maybe they added a dialpad?
Thanks, Dana, I know keypad dtmf worked with sjphone at some stage,
but at
[EMAIL PROTECTED] (Tony Mountifield) writes:
Yes, based on a standard install of the INN rpm in Red Hat or Fedora.
I've just put together a page with a description and links to the two
perl scripts used. See http://www.softins.co.uk/mail2news
Geez, right on time :) I just installed inn and
tim panton [EMAIL PROTECTED] writes:
On 4 Apr 2005, at 09:25, Shaoul Jacobson - TELLINK wrote:
Hi,
QoS is nice (and important) but only works within a FULLY controlled
end to
end link.
Inside a BIG enterprise LAN, on leased lines its OK.
Using end to end MPLS should also be ok
Mind that
Bernie [EMAIL PROTECTED] writes:
can that number be reduced? I'm looking at a system that would be
deployed to remote offices over fairly limited bandwidth links and
need to find a way of balancing quality vs. bandwidth constraints.
B
William Boehlke wrote:
The simple answer is 64KB.
Maik Hassel [EMAIL PROTECTED] writes:
Hello everybody,
I started using the XTen Xlite softphones (just to get something up
and running quickly). Everything works fine now, but the sound quality
is somewhat disappointing.
The sending - e.g. everything I say, dtmf tones, etc - receives the
Jean-Michel Hiver [EMAIL PROTECTED] writes:
Jean-Michel Hiver wrote:
Oops, sorry for the list reply :/
Actually, why does the Reply-To point to the Asterisk Users mailing
list? This breaks the reply to sender only / reply to all / list reply
functionality of my mailer. It's really broken
Damon Estep [EMAIL PROTECTED] writes:
http://groups-beta.google.com/group/Asterisk-test
Stuff shows up fast! Anyone have insight on this, did I miss something?
Apparently, somebody created that group on google groups and subscribed
it to the * mailing list. As long as registered, anybody can
trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] writes:
a couple other lists that I am on got notices last night that they were
added to google groups. I wonder if this is a google marketing ploy,
seek out all lists and subscribe them then spam the various lists
informing the
Damon Estep [EMAIL PROTECTED] writes:
What I'm still wondering about is, while you can post to that group,
whether your postings are actually propagated to this list. Did
anybody
try that?
Regards, Bruno.
Postings to google are not mirrored here, tried it. I think we are going
to
John Novack [EMAIL PROTECTED] writes:
Bruno Hertz wrote:
Jean-Michel Hiver [EMAIL PROTECTED] writes:
Jean-Michel Hiver wrote:
Oops, sorry for the list reply :/
Actually, why does the Reply-To point to the Asterisk Users
mailing
list
Damon Estep [EMAIL PROTECTED] writes:
Why? I'd say it's only a config issue. As long as the google group
has this mailing list as it's only feed and posting to the group
is equivalent to posting to the list everything should be fine.
How do you propose getting posts from google to here?
Damon Estep [EMAIL PROTECTED] writes:
Why? I'd say it's only a config issue. As long as the google group
has this mailing list as it's only feed and posting to the group
is equivalent to posting to the list everything should be fine.
How do you propose getting posts from google to here?
Josiah Bryan [EMAIL PROTECTED] writes:
On Friday 08 April 2005 1:12 pm, Bruno Hertz wrote:
Well, the reason for the latter apparently is that, in some postings to
this list, there's actually two entries in the reply-to header, the posters
mail and the list address, while in others it's only
tim panton [EMAIL PROTECTED] writes:
On 8 Apr 2005, at 20:02, Bruno Hertz wrote:
I don't know at all how it's currently implemented. All I say is
that, from
the technical pov, proxying any list through such a group should be
feasible,
without incurring major troubles.
Given that Google
Damon Estep [EMAIL PROTECTED] writes:
On Fri, 2005-04-08 at 12:01 -0600, Damon Estep wrote:
Why? I'd say it's only a config issue. As long as the google
group
has this mailing list as it's only feed and posting to the
group
is equivalent to posting to the list everything should
Roman Volf [EMAIL PROTECTED] writes:
I have noticed that many threads
don't go as well as planned and wind up in the wrong place.
But you do realize that that's not google's fault :)
Regards, Bruno.
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Joe S [EMAIL PROTECTED] writes:
Hi,
I am new with asterisk. I was wondering if there is a way to call a
OH323 user or SIP user using Netmeeting/SJPhone with H323 as the
default protocol without having a gatekeeper.
I can make a call from SIP to OH323 by specifying it in the
Joe S [EMAIL PROTECTED] writes:
Hi Bruno,
Thanks for the input, one question. Let's say I define context=default
in my oh323.conf.
Then, in my extensiions.conf I have:
[default]
exten=1002, 1, Dial(SIP/1002); 1001 is an Xlite SIP UA
so how do I call a sip user like from
Joe S [EMAIL PROTECTED] writes:
Hi Bruno,
Thanks I appreciate your help its really working, I just dial 1002 for
NM, and Xlite is ringing.
Joe.
Welcome. Thanks for your feedback, too. Good to hear it works,
especially if similar questions come up in the future.
Regards, Bruno.
Guillermo Salas M. [EMAIL PROTECTED] writes:
Bruno Hertz wrote:
Joe S [EMAIL PROTECTED] writes:
Hi,
I am new with asterisk. I was wondering if there is a way to call a
OH323 user or SIP user using Netmeeting/SJPhone with H323 as the
default protocol without having a gatekeeper.
I can make
Damian Funnell [EMAIL PROTECTED] writes:
Hi Manish,
Sure can, although you will need a timing source.
Not necessarily. In a pure VoIP environment, I don't know of any
asterisk application which needs timing other than meetme.
I.e. if you need conferencing, you'll need ztdummy as a timing
Manish Sapariya [EMAIL PROTECTED] writes:
Hi,
I was going through some of the list postings...and I felt
like if want to do voip within a LAN, I might have to install
Asterisk on every machine.
I hope it is not the case.
What I understand is (or what I want is)
- Install asterisk on one
Sig Lange [EMAIL PROTECTED] writes:
Starting around Apr 14th Gmail has started marking all messages for
Asterisk-Users as spam. Prior to that on google
groups
someone created a asterisk-test group (seperate from this group). Is
this perhaps related? I believe it all has
happened within a
On Sun, 2005-01-02 at 23:16 -0500, Karl Brose wrote:
The public post is to warn others of these tactics.
Which is completely OK, thank you. Although it seems to be
common practice among recruitment agencies, companies and
even private individuals to place fake job offerings I still
consider
On Mon, 2005-01-03 at 09:14 -0500, C F wrote:
This is the original post:
I've been contracted by a company in NYC area to install and
congfigure an * system. However I live 70 miles from nyc, they want a
service contract with shomeon local. Are you interested? must show
that s/he know linux
On Mon, 2005-01-03 at 09:18 -0500, C F wrote:
As for the term contracted that I have been using, thats because I
HAVE BEEN CONTRACTED, I just don't have a commitment for asterisk, if
they don't take asterisk I am installing for them Artisoft
Tellevantage. I am a authorized reseller for
On Thu, 2005-01-06 at 12:15 +1100, Adam Goryachev wrote:
Personally, I've never had a problem with Debian.
I'd second that. Just last month, I tried to install asterisk
with AVM proprietary CAPI drivers on FC3, and it didn't work.
The driver just could not be loaded. I then switched to Debian
Though you probably won't use them, I'd still like to mention fyi that
proprietary AVM Fritz PCI Card drivers didn't work for me on FC3. They
did on Debian Sarge.
Regards, Bruno.
On Thu, 2005-01-06 at 19:32 +0200, Shoval Tomer wrote:
Hi all.
Can anyone comment why shouldn't we use FC 3 for
Hi folks
an issue I don't understand. I'm running * stable 1.0.3 on public
internet, with following iax.conf / sip.conf entries:
iax.conf
[100]
type=friend
username=Foo
context=default
auth=md5,plaintext,rsa
secret=secret
host=dynamic
callerid=Foo 100
qualify=no
sip.conf
[10]
On Wed, 2005-01-12 at 14:39 -0800, Erik Espinoza wrote:
Did you enable passthrough for the rtp ports on the asterisk box?
I had the same problem until I enabled udp 1:2 on the firewall.
I did. That's why linphone - * echo test works.
Maybe I made some progress however, by logging
OK, I'm coming to think linphone is bullshitting me.
I now tried the following call paths
firefly - * - iaxcomm works
firefly - * - linphone works
sjphone - * - iaxcomm works, especially sip-iax works
sjphone - * - linphone works
The opposite paths work too except
linphone - * - firefly
On Sat, 2005-01-15 at 05:37 +1100, Howard Lowndes wrote:
Can anyone _recommend_ a downloadable OSS softphone that _works_ under
Linux and is compatible with Asterisk.
So far I have tried kphone and linphone and had problems with both, and
I am still waiting to hear back from the X-Lite beta
On Fri, 2005-01-14 at 16:27 -0200, Denis Galvo - iSolve wrote:
Em Sex 14 Jan 2005 16:11, Dan escreveu:
I dont have problems when calling PSTN extensions, and calling VoceMail,
EchoTest, etc. The problem is related with the conversation between two
DIAX Softphones.
With * in the middle or
On Sun, 2005-01-16 at 16:52 -0500, Steve Kann wrote:
If the delay goes down after a couple of minutes after the transfer,
this could be the problem.
Just fyi, this is what I observed with those delays between iaxcomm
and firefly, i.e. they occurred on a transfer attempt and normalized
after
Hi folks
last weekend, I tried Windows Messenger first time and was stunned by
the little latency it gives. Until now, I've been using softphones on
Linux exclusively, like iaxcomm, linphone and sjphone, and they all give
me about 1, at times even 2 secs delay. Whereas Messenger really seems
to
On Mon, 2005-01-17 at 16:51 -0500, Steve Kann wrote:
What softphone are you using on Linux?
iaxcomm, linphone and sjphone, and they all give
If you use an iaxclient-based softphone on linux as root, it runs with
RT priority, and pretty low latency
Hmmm, on my side I can't say it makes
On Tue, 2005-01-18 at 07:43 +0800, Steve Underwood wrote:
Latencies that big should not be due to the softphone. They are often
due to the sound card driver.
Yeah, it's what I thought, but then, as said, I tried the planetccrma
kernel and drivers, which are supposed to support professional
On Tue, 2005-01-18 at 09:31 -0600, Bartosz Wegrzyn - asterisk wrote:
I am trying to register windows messanger with asterisk and it fails.
http://www.voip-info.org/wiki-Asterisk+phone+Windows+messenger
Check whether it's the realm.
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On Thu, 2005-01-20 at 14:51 -0800, Manjit Riat wrote:
Just got a headset for testing asterisk and am using X-Lite. I plugged
in the headset into the headset jack and is there any way to configure
X-lite to use the headset instead of the speakers? Or will I have to
plug the headset in the
On Thu, 2005-01-20 at 16:59 -0800, Manjit Riat wrote:
Oh sorry... just got carried away with all the help I got here.
No problem. Don't know about your headset, but usually it has
two connectors, which you plug into the mic and speaker jacks
of the sound card. XLite itself doesn't really care
On Sat, 2005-01-22 at 23:56 -0800, Kenneth Long wrote:
seem like some kind of port issue...
Probably. Both try to set up listeners on the IAX port
(4569 for IAX2). Disable or reconfigure one of them to
bind to a different port, whichever you want to answer
on it.
Also, don't forget to disable
The subject says it all. After digging through latency and other issues
with all kinds of linux softphones, I've found that only * works alright
for me as a VoIP client.
Problem now is that, unlike other apps, chan_oss resp. chan_alsa grab
the card once and won't release it until shutdown, while
Talking * 1.0.12 here.
Problem: I'd like to avoid retries with queue, i.e. if members choose to
ignore a call they should not be bothered again. On the other hand,
when a call times out according to the Queue(...) timeout, the call
should proceed to voicemail.
Setting retry in queue.conf to a
Just a point of order, there is no Asterisk 1.0.12. The latest is 1.0.5.
Sure, sorry, it's actually 1.0, i.e. CVS-v1-0-12/18/04-22:40:47.
Thanks, Bruno.
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On Thu, 2005-01-27 at 16:14 -0600, Eric Wieling wrote:
You might consider upgrading to 1.0.5 release
Thanks, I checked it out. With same config as for 1.0 I get:
Asterisk Ready.
-- Accepting AUTHENTICATED call from 192.168.0.10, requested format = 1024,
actual format = 1024
--
On Thu, 2005-01-27 at 20:35 +0100, Bruno Hertz wrote:
Anybody found a way around this (bug?), i.e. avoiding retries with
Queue(...|t) properly timing out at the same time ?
OK, I took a look at app_queue.c, and while the described behavior isn't
a bug, I still hacked the source to give me
On Sat, 2005-01-29 at 15:48 +0100, Zdik Kudrle wrote:
I'm running Asterisk with HFC-S card connected to HW PBX in my office.
When I make a call from home using iaxComm connected to Office Asterisk,
the outgoing latency is about 0.25 sec, which is quite OK. But to incoming
latency begins on
On Tue, 2005-02-01 at 14:09 +0900, Kuniyoshi Murata wrote:
Hi,
I'm thinking of setting up Asterisk for H.323 video phone clients.
Now, what is the difference between native H.323 that come with Asterisk and
Open H.323 for Asterisk ?
I can't tell you the exact differences, but oh323
On Tue, 2005-02-01 at 13:21 +0100, Robert Rozman wrote:
By the way: use asterisk-oh-0.7.x!
But shouldn't I use 0.6.5 cause I'm on cvs STABLE ?
You are completely right. 0.6.5 for STABLE and 0.7.x for HEAD.
Regards, Bruno,
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On Tue, 2005-02-01 at 16:27 +, Gareth Blades wrote:
Unattended transfers just does nothing. I cannot get it to do anything.
Not sure about this, but I'm under the impression that the # transfer
might need some client support.
E.g. I tried gnomemeeting - * - NAT - * - firefly and # did
On Tue, 2005-02-01 at 12:43 -0600, Eric Wieling wrote:
# transfers are controlled by features.conf and the t and T option
on the Dial line. It requires NO support in the client. In fact #
transfers are usually only useful if the client does not support
NATIVE TRANSFERS, i.e. real
On Tue, 2005-02-01 at 17:49 -0200, Denis Galvo - iSolve wrote:
I believe that your problem is related to DTMF problems with your
softphones.
Rather not, since the caller client was gnomemeeting in both cases,
and the caller tried to transfer the callee in both cases. With
sjphone as callee it
On Tue, 2005-02-01 at 21:29 +0100, Philipp von Klitzing wrote:
Gnomemeet -- * -- * -- Gnomemeet?
Gnomemeet -- * -- * -- firefly resp. sjphone, as said.
and insert codecs and protocols everywhere.
alaw gsmbridged for firefly
Gnomemeet --- * --- * ---
On Wed, 2005-02-02 at 07:12 +1100, Howard Lowndes wrote:
Surely there has to be one soft phone that works under Linux.
I've tried:
kphone - it sometimes complains about the need to release the sound
device
linphone - lowww
iaxcomm - needs some strange widgets
On Wed, 2005-02-02 at 09:09 -0500, Nabeel Jafferali wrote:
notransfer=yes
That prevents transfers but not bridging. As to my knowledge, there's
no way to prevent bridging.
Regards, Bruno.
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On Wed, 2005-02-02 at 14:36 +, Gareth Blades wrote:
If that is the case then it seems a serious limitation as it makes call
parking and attended transfers unusable.
Your only choice is to use the IAX native transfer where you cannot
speak to the recipient before transfering the call.
OK,
On Wed, 2005-02-02 at 14:21 -0800, Aaron Glenn wrote:
That's the least ambiguous subject I could muster. I'm relatively new
to Asterisk and while I'm certain there is a way to do this, I'm
unsure how. My question is this: How do I take an incoming call, put
the person on hold, and in the
On Fri, 2005-02-04 at 15:54 +0900, Andrew Kochetkoff wrote:
EndedByTransportFail
You should give more details, all there can be seen from your log is
that the call is dropped due to EndedByTransportFail, which can have
various reasons.
Apparently, this is a LAN call with no NAT involved,
On Tue, 2005-02-08 at 22:50 +0100, Stefan Gofferje wrote:
Queues are not too bad but lacking an important feature. As far as I
could figure out, they couldn't just ring without answering.
Hum? I'm running * 1.0.5, and Queue rings without prior answering the
line. Although most queue examples
On Wed, 2005-02-09 at 00:37 +0100, Remco Barende wrote:
Could you post your configs please? I need the same for my home setup :)
Sure, but there's nothing sophisticated:
extensions.conf
---
[myqueue-in]
exten = s,1,Queue(myqueue25)
exten = s,2,Macro(vm,${MYEXTEN})
exten =
On Thu, 2005-02-10 at 10:47 -0600, Steven Critchfield wrote:
This is a good example of why ease of use is not always a good thing.
Had you actually had to learn more before you had an install, you would
have been through a text or two that mention password strengths.
Apropos ease of use: on
On Thu, 2005-02-10 at 11:09 -0500, Jason Stewart wrote:
There's a chance that you may have been hacked, but the logs you post
look more like your mailserver is an open relay.
You sure? I run postfix myself and am not proficient in analyzing
sendmail logs, but looking at those lines
Feb 9
On Thu, 2005-02-10 at 10:42 -0600, Daniel Wright wrote:
You can always set up ssh to use host keys. Here are two howto's on what
else? How to set them up.
http://www.securityfocus.com/infocus/1806 Part 1
http://www.securityfocus.com/infocus/1810 Part 2
Great links. One may add that first
On Thu, 2005-02-10 at 09:57 -0700, Colin Anderson wrote:
5. Use key-based auth mechanism rather than password. It's my understanding
that the key is never sent, only a hash of the key. The target system
compares the hash against it's hash of the key, and if it matches, cool.
Not exactly, for
On Thu, 2005-02-10 at 13:14 -0400, [=Jorge Boscan Etura=] wrote:
Hi
I just installed * 1.0.5 on a my fc2, but when i try to use kphone to
begin testing it doesnt work because * is using /dev/dsp, how can I
configure/resolve this?
The * modules which may use the sound card (and enable you to
On Thu, 2005-02-10 at 13:30 -0600, Steven Critchfield wrote:
Nothing like sending a valid key to a man-in-the-middle.
There is no key sent to man-in-the-middle except the pub one,
which does no harm anyway. What does harm is you're logging
into/routing through a different machine that you
On Thu, 2005-02-10 at 21:44 -0600, Steven Critchfield wrote:
So you probably want to still turn off the
webserver and jabber server, they would be better off coloed anyways and
there are a lot of cheap colo places for non critical hosting.
As a sidenote, you can also set up traffic shaping
On Sun, 2005-02-13 at 04:46 +0100, Andres Gmez Garca wrote:
I've tried GNOMEMeeting also. It works fine with a P2P client
connections (ALSA works fine) but, even when I success connecting to an
asterisk server, I haven't hear anything. I mean, I don't hear the demo
successfull messages. I've
Addendum: I did a little investigation and found this
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=272259
Regards, Bruno.
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On Sun, 2005-02-13 at 18:10 +0100, Andres Gmez Garca wrote:
Thanks Bruno, I'll try it.
Also, you might take a look again at
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=272259
Following your mail, I wrote to that list (cf the last mails there),
and it looks like a working oh323 package
On Mon, 2005-02-14 at 22:29 +1100, Duane wrote:
I gave up trying to use linux soft clients they all seem to have some
fatal flaws or issues I could never fully get rid of
While I'd second that, Gnomemeeting is still pretty good and by far the
best softphone I've used on Linux. Currently, it
On Mon, 2005-02-14 at 10:47 -0700, Kyle Hagan wrote:
I used to use kphone and have very bad echo, I switched to sjphone and
it worked great.
It isn't too bad, but it has latency (compare it e.g. to asterisk as
softphone and you'll see what I mean) and no dial pad. So I found it
isn't really
On Mon, 2005-02-14 at 14:22 -0500, Dana Olson wrote:
Do you have this documented somewhere? Is this for the Linux Xlite and
SJphone only, or the Win32 ones as well?
We're talking Linux currently, don't know about Windows. Documented?
On the cornfed website it's specifically mentioned, with
On Tue, 2005-02-15 at 11:43 +1100, Duane wrote:
Yea I've been hanging out for them to support it for ages now...
Hehe. Not the worst thing to hang out for :) Anyway, OPAL seems
to have a reasonably working SIP stack by now, I did a test run
with the cli client and it worked. Some features are
On Tue, 2005-02-15 at 13:59 +, Alistair Cunningham wrote:
It can also handle video calls, though I have not used this myself.
AFAIK video only with SIP, which I didn't test myself either. With
H323 it does not work, audio only there.
Regards, Bruno.
On Sun, 2005-02-20 at 13:51 -0500, Paul wrote:
Or maybe a double fool because he also disrespected Debian GNU/Linux in
his reply.
*And* recommended Fedora, which makes it triple. I just dumped FC3 and
replaced it with Debian because Fedora's kernels constantly gave me
issues, e.g. with
On Fri, 2005-03-04 at 18:17 -0800, Dan Austin wrote:
There does not seem to be too much interest in this, but it has
helped me sell the idea of dumping a very expensive, but poorly
functioning, existing VoIP conferencing system. In the future
I can send announcements directly to the few
On Tue, 2005-03-08 at 17:13 +0100, Michael Vogel wrote:
So I want to register the SIP client at the asterisk server that itself
is registered at the different SIP providers.
Does that work the way I want?
It's what people do here all the time. One issue might arise though,
i.e. where your
On Fri, 2005-03-11 at 15:32 +0100, Stefan Stolz wrote:
Hello,
i tryed to read the Wiki, but i am not sure if i am right with Asterisk.
Until now i made my phone calls with ant-phone over my ISDN Fritz Card. Now i
tryed to search a way to phone from other computers in the internal net over
I'm considering that board as a mail and voip gateway for home use.
In view of all those statements about how little resources asterisk
needs, did anybody already try running asterisk on it?
Thanks, Bruno.
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