Hi
I have recently updates to the latest cvs of asterisk, openh323 and pwlib as
recommended.
The OPenh323 and pwlib compile fine.
When compiling the Asterisk-oh323 I get the following errors, I have checked that the
envorinment variables are set correctlty as below.
PWLIBDIR=/usr/src/pwlib
apply the openh323 patch (it's in the root of ast-oh323), recompile
openh323 and it should work fine
David Hindmarsh wrote:
Hi
I have recently updates to the latest cvs of asterisk, openh323 and pwlib as recommended.
The OPenh323 and pwlib compile fine.
When compiling the Asterisk-oh323 I
Just had another thought, about replacing the cb and zaptel card with a
sipanalog gateway... Can anyone recommend one? (in case I can't get
this straightened out)
Here's our config:
cisco 7960's running 6.3 sip code
latest cvs of *
t100p zaptel card
adit 600 channel bank
7 pots lines and
On Wed, 2004-04-21 at 20:20, David Carter wrote:
I'm considering using Asterisk with some type of Cisco phone, and currently
considering either the 7940 or 7960 because of its more-complete functionality
(compared to the 7905).
I'm currently wondering:
Do all the expected functions
We are experiencing problems on a FXS interface where the client is dialing
numbers, but digits are being dropped somewhere from the dial string.
Typically one or two digits are not being presented. We've tried different
handsets to no avail, and I am assuming that it is some sort of timing
Hello,
We have a quadBRI in NT mode with bri_cpe_ptmp signalling and when connect a
ISDN telephone to this nothings happen.
What can I do?
My config files are this:
Zaptel.conf:
loadzone=es
defaultzone=es
# qozap span definitions
# most of the values should be bogus because we are not really
Also, a better alternative is MAD player. And there is a patch for
Asterisk that adds support for it.
http://bugs.digium.com/bug_view_page.php?bug_id=0001365
Michael.
brian k. west wrote:
We find that mpg123 0.59r works best. mpg123 0.59s-mh4 = the devil.
What versions does everyone use
Le jeu 06/05/2004 à 18:52, Michael Manousos a écrit :
This new version (0.6.1) of asterisk-oh323 fixes the one-way audio
problem of the previous release.
Hi, what is the difference between chan_h323 and asterisk-oh323? Are
they mutually exclusive? Is one better than the other?
chan_h323 came
Paul Berger wrote:
Le jeu 06/05/2004 18:52, Michael Manousos a crit :
This new version (0.6.1) of asterisk-oh323 fixes the one-way audio
problem of the previous release.
Hi, what is the difference between chan_h323 and asterisk-oh323? Are
they mutually exclusive? Is one better than the
[EMAIL PROTECTED] wrote:
Hi
I have Asterisk with two E100P cards
One connected to PSTN and other to my local PBX
I'm running into problem with faxes.
Faxes are connected to PBX and asterisk should just bridge the fax call
from one span to another.
Problem is that even if the fax image reach
Le ven 07/05/2004 à 11:59, Michael Manousos a écrit :
They are mutually exclusive because they try to do the same thing.
Why 2 different projects for the same goal? (i hope I'm not starting a
flame war :-))
OK, try it and let me know what you think.
I seem to have a problem when compiling
Hi there,
since a couple of days I can't seem to be able to compile CVS HEAD on
RH7.2. On a RH7.3 machine with bison-1.35-1 it appears to be fine
though... any advice?
Philipp
System: RH 7.2
bison-1.28-7
Related issue:
http://rpm.pbone.net/index.php3/stat/4/idpl/411535/com/bison-1.35-
When talking to me, people are complaining the volume was not high enough.
The phone only allows to change the volume of the speaker/earpiece. Is
there an alternative solution? Is it possible to increase the volume in
asterisk?
Frederic
___
Hm...
since a couple of days I can't seem to be able to compile CVS HEAD on
RH7.2. On a RH7.3 machine with bison-1.35-1 it appears to be fine
though... any advice?
Actually this doesn't seem to be related to bison - I can't even compile
my old CVS-HEAD-05/03/04-19:58:33 anymore, getting
Paul Berger wrote:
Le ven 07/05/2004 11:59, Michael Manousos a crit :
They are mutually exclusive because they try to do the same thing.
Why 2 different projects for the same goal? (i hope I'm not starting a
flame war :-))
Because there are two (or even more) ways to solve the problem.
This
Can you send me off-list a full debug (-vvvcd) output of the call?
Michael.
Michael Niehren wrote:
Still one-way audio problems with version V0.6.1.
Hi Michael,
using asterisk as ISDN2H323-Gateway. Call from ISDN - Asterisk - H323 is now
ok, but in the other direction there is still only
Hello,
i have a Winmodem (Softwaremodem) i know, this is a problem under linux.
But asterisk loaded a Modem channel.
What i wanted to know is, can i use this channel to make a PSTN call??
If yes, how can i do that. Which *.conf files do i have to change??
Has someone experience with that??
Kindly
We find that mpg123 0.59r works best. mpg123 0.59s-mh4 = the devil.
What versions does everyone use without problems.
0.59r is PERFECT
Been using mpg123-0.59q-1.i386.rpm since late September with no problems.
Not had a need to change; ain't broke, don't fix it.
Le ven 07/05/2004 à 13:18, Michael Manousos a écrit :
Because there are two (or even more) ways to solve the problem.
This topic has been discussed in the past several times. Check the
archives for details.
Sorry, I should have started there...
Use latest CVS asterisk.
I was using the
Has anyone found any good online resources for performing transmission
tests for POTS lines? There is plenty of info on this list about
adjusting gains
on X100 cards, etc. but I am looking for test procedures using test sets.
I'm talking about tests for echo loss, distortion, etc. Thanks in
Frederic Steinfels wrote:
When talking to me, people are complaining the volume was not high enough.
The phone only allows to change the volume of the speaker/earpiece. Is
there an alternative solution? Is it possible to increase the volume in
asterisk?
Frederic
On Fri, 7 May 2004 10:51:56 +0200
Pedro Vela [EMAIL PROTECTED] wrote:
Hello,
We have a quadBRI in NT mode with bri_cpe_ptmp signalling and when connect a
ISDN telephone to this nothings happen.
Do you have a power feeding module for the quadBRI, I think you need that, but not
shure.
Best
I have noticed that when I switched to macros in my extensions.conf, there
is now a 5 second delay.
The macro starts with an announcement and then voicemail.
Has anybody noticed the same?
is it a feature?
URiel
___
Asterisk-Users mailing list
[EMAIL
Hi folks,
this does look like a bug to me: Asterisk replaces the @63.214.186.6 by
@context which obviously leads to a failure. Any comments, do I have a
configuration issue on my side that I missed?
Cheers, Philipp
-- Executing Dial(SIP/philipp-bd5f, SIP/[EMAIL PROTECTED]
out|90) in new
Sorry very very very newbie here,
I just started setting up a asterix box as a test environment for my
work to see if it is a viable solution.
I have a standard TMD400P Development Kit with a FXS and FXO module on
it, and a standard analog handset plugged into the FXS module and a
Analog phone
Hello there!
Somebody tried the meetme|b which initiates the conf-background
AGI
Actually I want to originate another call from a conferencemy
AGI originates the call and connects it to the conference, but the call is nowhere
My extension
exten = 21,1,meetme(21|pb)
and my
Are there any other wireless IP phones out there other then the Cisco
7920??
--
James Moran [EMAIL PROTECTED]
Potential Technologies
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
* James Bean; [EMAIL PROTECTED] on 07 May, 2004 wrote:
Sorry for my lack of knowledge in this area but if someone could point
me in the right direction or send me a zaptel.conf and zaptela.conf that
would work in my situation it would be very much appreciated, some of
the basic text files I am
Upgrade bison...i had the same problems until i upgraded bison.
On Fri, 2004-05-07 at 07:17, Philipp von Klitzing wrote:
Hm...
since a couple of days I can't seem to be able to compile CVS HEAD on
RH7.2. On a RH7.3 machine with bison-1.35-1 it appears to be fine
though... any advice?
Hi everybody,
I would like to create SIP call flow Diagram under Windows. Is anybody
know a program to perform it? I have already Ethereal and I would like
an explicit diagram just to show where something have problems...
Thanks
Ignace
___
Why not get a analogue to IP adapter, then use a Digital Cordless phone.
Much cheeper than the 7920 and works wonders for me.
I've got a couple of adapters for sale at the moment, e-mail me if your
interested !
Paul.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Philipp von Klitzing wrote:
Hi folks,
this does look like a bug to me: Asterisk replaces the @63.214.186.6 by
@context which obviously leads to a failure. Any comments, do I have a
configuration issue on my side that I missed?
We don't support 302 redirects to other hosts/domains now.
It's a
Hey there,
There is ZyXel Prestige 2000W Wi-Fi IEEE 802.11b
David
- Original Message -
From: James Moran [EMAIL PROTECTED]
To: Asterisk [EMAIL PROTECTED]
Sent: Friday, May 07, 2004 8:59 PM
Subject: [Asterisk-Users] WI FI IP phones??
Are there any other wireless IP phones out there
I would like to create SIP call flow Diagram under Windows. Is anybody
know a program to perform it? I have already Ethereal and I would like
an explicit diagram just to show where something have problems...
Might take a look at:
On Fri, 2004-05-07 at 07:41, Philipp von Klitzing wrote:
Hi folks,
this does look like a bug to me: Asterisk replaces the @63.214.186.6 by
@context which obviously leads to a failure. Any comments, do I have a
configuration issue on my side that I missed?
Cheers, Philipp
--
We need to have about 30 phones on one floor
On Fri, 2004-05-07 at 09:18, Paul Tyreman wrote:
Why not get a analogue to IP adapter, then use a Digital Cordless phone.
Much cheeper than the 7920 and works wonders for me.
I've got a couple of adapters for sale at the moment, e-mail me if
Title: RE: [Asterisk-Users] Cisco 7940/7960 SIP functionality questions
Is the 7970 still problematic?
-Original Message-
From: Fran Boon [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, April 21, 2004 3:29 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7940/7960 SIP
And
what problem do you have with registering ?
Jeremy
Jones has recently posted his SNMP walkthrough from a mediatrix 1104 - you might
reference that, configuring 1204 should be very similar to that of
1104.
Regards,
Dave
-Original Message-From:
[EMAIL PROTECTED]
James Moran wrote:
We need to have about 30 phones on one floor
And you really think that WiFi phones are suited for this application?
Not an RF engineer, are ya?
John
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
My thoughts also. :P
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Friday, May 07, 2004 7:50 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] mpg123 versions ?
We find that mpg123 0.59r works best.
Run /usr/src/zaptel/ztmonitor 32 -v
And adjust your gains in /etc/asterisk/zapata.conf accordingly.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bam
Sent: Friday,
Only works on zap interfaces. What
are you using?
bkw
-Original Message-
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Atif Rasheed
Sent: Friday, May 07, 2004 7:57 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] meetme
conf-background.agi
Hello there!
i prefer zyxel p200w
for a picture see
http://www.voipbox.de/images/private/protzundco/equip.jpg at the upper
left corner
James Moran wrote:
Are there any other wireless IP phones out there other then the Cisco
7920??
___
Asterisk-Users mailing
Hi Andreas,
I guess it is better to buy a B1 or C2 :-). They are not very expensive at
ebay. Or you buy digium hardware, it surely runs with *...
Or have a look at www.junghanns.net (author of chan_capi)
He sells a 4 Port BRI ...
Bye
Felix
-Original Message-
From: [EMAIL PROTECTED]
We need to have about 30 phones on one floor
And you really think that WiFi phones are suited for this application?
Not an RF engineer, are ya?
at ~80kbps per phone and (guessing) 5.5mbps average connect I would be curious
to see how bad 30 simultaneous conversations would be with a CSMA/CA
i prefer zyxel p200w
Looks just like the Pulver OEM'd WiSIP.
Regards,
Andrew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
The 1204 does not have the software routines implemented for register.
Their approach is the 1104 registers with the 1204.
And what problem do you have with registering ?
Jeremy Jones has recently posted his SNMP walkthrough from a mediatrix 1104 - you
might
Hi!
I know that is a very posted matter but i have a
question:
Some one can translate that messages for me? what
is the mean of that messages? can i do something to correct this and get
the caller id to work?
May 7 11:26:19 ERROR[1288925632]:
callerid.c:192 callerid_feed: fsk_serie made
Hi!
302 Moved is not fully supported by chan_sip. Personally I like this
because the way Asterisk currently supports 302 Moved will prevent
calls from being forwarded outside of Asterisk's dialplan. I would
just create an exten = joesmith,1,GoTo(xxx,n) where xxx is the
extension you want
Hi!
Upgrade bison...i had the same problems until i upgraded bison.
Which means upgrading glibc ... :-((
In other words: Asterisk won't work with RH 7.2 (and the like) anymore,
basically. Still I wonder why I was once able to compile the March 5 CVS,
but can't do so anymore. Might be
I've had this too, reported it as a bug last week and got my butt kicked
for not being responsive enough in providing support to sort it out. You
could file another bug report but be sure to have a thick book ready to
stuff down your trousers.
Iain
--On Friday, May 7, 2004 10:43 am -0400
On Fri, 2004-05-07 at 09:58, Philipp von Klitzing wrote:
The way it works now I am not able to call an *unkown* Nikotel SIP user
(i.e. I am not aware of which username is mapped to the 99xx
number), and that's not really so nice. Of course the other option is to
tell Nikotel to
You are assuming that they all have to go to the same Access Point and be on the
same channel. For a high density setup you can get APs that allow you to turn
down the signal strength so they can be more densly placed. With the Wisip or
the Zytel you really need to go with g729 anyway for them to
It's not the switch. It's lightly loaded 100Mb.
-brian
Bisker, Scott (7805) wrote:
What kind of switch do you have your phones plugged into? If your switch is highly loaded, or you are doing lots of multicast or broadcast, your SIP streams are going to suffer unless you are filtering that
At 09:43 AM 5/7/2004, you wrote:
It seems that each time I get a new checkout of * from CVS my Cisco 7960
works worse than before. I know this stuff's in flux, so I mention this in
case it's news. Anyone else having trouble? What I'm seeing (er,
hearing) is really choppy audio. The previous
Hi!
| exten = 999,1,SetGroup(moh)
| exten = 999,2,CheckGroup(1)
| exten = 999,3,Answer
| exten = 999,4,MusicOnHold(default)
|
| See?
| You can limit that to just 1 user at a time or what ever you wish :
|
| bkw
Great! So this is a means that can be used as an outgoing limit
Hello,
I have fix the problem, i haven't notice that's in general i have
videosupport=yes
with this in sip.conf, it's doesn't disable videosupport :
[provider]
host=x.x.x.x
type=peer
videosupport=no
silenceSuppression=no
Now working with videosupport=no in general
At 17:08 07/05/2004, you
It seems that each time I get a new checkout of * from CVS my Cisco 7960
works worse than before. I know this stuff's in flux, so I mention this
in case it's news. Anyone else having trouble? What I'm seeing (er,
hearing) is really choppy audio. The previous version I had installed
had
I've got Asterisk STABLE-CVS-4/19/04 with 12 Cisco 7960 phones 6.0 Firmware using
ulaw, 6 Polycom IP500 ulaw phones, and 192 Zap channels. I have Gig-E Copper to my
server and 100Mbit-Full to all my phones. I haven't had any choppy audio at all. My
switch is a Cisco 4500.
-sb
Ah, this reminds me that I forgot to mention that our network looks like
this:
Cisco --- SIP Asterisk IAX Aterisk IAX
Asterisk PRI PSTN
-brian
Tom wrote:
At 09:43 AM 5/7/2004, you wrote:
It seems that each time I get a new checkout of * from CVS my
Just an FYI if you can run tethereal -n udp port 4569
And watch the timestamps(should be even 20ms increments per call leg). Both
ends will need to be updated also. If not you will get some very strange
timestamp issues and jitter and timestamps might not be right. If you
have one end on
Or any channel for that matter.
Bkw :)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing
Sent: Friday, May 07, 2004 10:35 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] HOW TO PROGRAM NEW MODULES
Hi!
Hi all;
I have been searching for an answer to a question that a customer asked
me and I have only found a few older answers. So, wanting to find out
if anyone has any experience with this issue and can help provide me
with some advice.
I have a customer which is strongly interested in using
Paul Berger wrote:
Le jeu 06/05/2004 à 18:52, Michael Manousos a écrit :
This new version (0.6.1) of asterisk-oh323 fixes the one-way audio
problem of the previous release.
Hi, what is the difference between chan_h323 and asterisk-oh323? Are
they mutually exclusive? Is one better than the
Hi!
I don't want to re-invent the wheel if someone has already hacked a way
to do this.
One of my customers has a number of stores, and he wants to leave one
voicemail that would be delivered to all the managers at once. Each has
a voicemail account on his server.
I have googled
Upgrade each asterisk (iax2) and the problem will go away. As bkw
mentioned, the problem sources from the location with the older
iax2 code (which probably includes the Stable cvs I believe).
NuFone had the problem in mid/late April as well, but they apparently
updated their code when the issue
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Eric Wieling wrote:
| Allow ULAW or ALAW, not both, at least for trying to solve a problem.
What is the difference between these codecs? Which is better?
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN
I've had a quick fiddle to little avail, the readings looked prey good to
be honest before I started fiddling. Looking a little closer it appears
that it is the digit 1 that is being lost more that any other.
At 15:25 07/05/04, you wrote:
Run /usr/src/zaptel/ztmonitor 32 -v
And adjust your
The SIP 6.1 image has auto answer available, which would function the same
as the SCCP implementation.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Travers
Sent: Friday, May 07, 2004 12:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco
I have been searching for an answer to a question that a customer asked
me and I have only found a few older answers. So, wanting to find out
if anyone has any experience with this issue and can help provide me
with some advice.
I have a customer which is strongly interested in using
Le ven 07/05/2004 à 18:03, Jeremy McNamara a écrit :
fact I created chan_h323 because the author of asterisk-oh323
wouldn't listen to the rest of the community on proper implementation of
an Asterisk channel driver. After many complaints from others about
asterisk-oh323 on the mailing
No I'm not but it's a hospital that nurses are on call and need to have
a way to contact them. On Fri, 2004-05-07 at 09:52, John Fraizer wrote:
James Moran wrote:
We need to have about 30 phones on one floor
And you really think that WiFi phones are suited for this application?
Not
The manufacturer think its a bad phone. So Im getting another one today.
They said thay have gotten the Zultys phones to work with asterisk with no
problems. Will let everyone know.
Zultys sait ULAW was the most common but did not state why most use it.
Kyle
- Original Message -
PROBLEM: My x100p only does bad things. When I plug the line from the
wall into the card, other phones in my house go nuts, no dialtone, crazy
clicking, random tones, etc. Following the steps below, the line is
screwed up any time I test after step 2. Just plugging the line from the
wall
On Fri, 2004-05-07 at 11:17, Jason A. Pattie wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Eric Wieling wrote:
| Allow ULAW or ALAW, not both, at least for trying to solve a problem.
What is the difference between these codecs? Which is better?
Neither is better. ulaw is used in
On Fri, May 07, 2004 at 10:45:13AM -0400, Rick Beasley wrote:
PROBLEM: My x100p only does bad things. When I plug the line from the
wall into the card, other phones in my house go nuts, no dialtone, crazy
clicking, random tones, etc. Following the steps below, the line is
screwed up any
James Moran wrote:
No I'm not but it's a hospital that nurses are on call and need to have
a way to contact them. On Fri, 2004-05-07 at 09:52, John Fraizer wrote:
James Moran wrote:
We need to have about 30 phones on one floor
And you really think that WiFi phones are suited for this
Hmm I'll look into it. Thanks.
On Fri, 2004-05-07 at 12:54, John Fraizer wrote:
James Moran wrote:
No I'm not but it's a hospital that nurses are on call and need to have
a way to contact them. On Fri, 2004-05-07 at 09:52, John Fraizer wrote:
James Moran wrote:
We need to have
Hi!
Upgrade bison...i had the same problems until i upgraded bison.
Which means upgrading glibc ... :-((
Ok ok, I got it - compiled bison from source and disregarded those good-
looking tail-shaking RPMs. ;- Works fine now.
Cheers, Philipp
___
Hi!
able to support intercom/paging. Having searched the archives, it
appears that this question was asked about 6 months ago, and the answer
was that the Cisco phones support this using SCCP and having one line
set to auto-answer, but at the time this was not supported in the SIP
Hi all
OK this may sound like a good one but maybe someone can tell me.
Simple context is - I want to unplug my existing conventional PBX from the
Telco and place * with it's TE410P in between.
Now the difficult part, the existing connection is E1 PRI (Q.931) with 6
B-channels. I need to be
Why not vocera?
http://www.vocera.com
they seem to have the exact product you are looking for and seem to
primarily server hospitals..
-Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Moran
Sent: Friday, May 07, 2004 1:06 PM
To: Asterisk
I turned down the rxgain and txgain to -22.0 and -16.0 respectively and
things started to look a whole lot more acceptable. Then the client sticks
on his BT DECT phone and I start losing all the 1s from the dial string.
Does anyone know if BT DECT phones have dodgy DTMF tones?
At 17:19
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 05 May 2004 12:35 pm, Darren Nickerson wrote:
Folks,
The silence was deafening ... I had a few private replies but overall I'd
have to conclude that most people on this list aren't interested in faxing
thru Asterisk. You're all
Hey everyone,
I want to run different lines directly to different extensions on two
FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to
extensions 102
Does anyone know of a way to do this?
Thanks!
Chris
___
Asterisk-Users mailing list
Have you looked on how much they cost.
On Fri, 2004-05-07 at 13:20, Mark Musone wrote:
Why not vocera?
http://www.vocera.com
they seem to have the exact product you are looking for and seem to
primarily server hospitals..
-Mark
-Original Message-
From: [EMAIL PROTECTED]
I just created my own rpms...or you could have downloaded the fedora
srpm and rebuilt it.
DP
On Fri, 2004-05-07 at 12:57, Philipp von Klitzing wrote:
Hi!
Upgrade bison...i had the same problems until i upgraded bison.
Which means upgrading glibc ... :-((
Ok ok, I got it - compiled
Do you have a jitterbuffer enabled on your inter-asterisk IAX trunks? If so, try disabling it cleared everything up for me. With jitter buffer enabled using the default settings my audio across the IAX trunk was terrible. BTW, my 7960's are running 5.3 firmware so I probably don't see the
I've actually engineered some WiFi at come medical clinics and it does
depend on the gear you purchase. Cisco addresses this in their marketing
and technical spec sheets. The two major hospitals in my area use wireless
for their phones and mobile laptops for the nurses as they go room to room
I use callflow (callflow.sourceforge.net)
works under linux with ethereal dump, and produces
html+images pages, for viewing them via a web browser.
Matteo.
Il ven, 2004-05-07 alle 15:14, Ignace CARIA ha scritto:
Hi everybody,
I would like to create SIP call flow Diagram under Windows. Is
I am using ISDN with CAPI and Eicon Diva card.
On ISDN calls in and out, some people are saying they find
it hard to hear us. Its only the odd few though, not everyone. We can hear them
no problem.
Do I just increase the txgain?
What is the limit for txgain, or are there any
Chris Wilson wrote:
I want to run different lines directly to different extensions on two
FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to
extensions 102
You should be able to stick different channels into different default
contexts in zapata.conf. Then just have the context for
Symbol have the netvision line of h.323 wireless phones used in hospitals
with multiple logins etc... , i have one here in my office and it works very
well with a simple 3com officeconnect gateway, makes direct calls, have
integration with various pbx.. a good product.
www.symbol.com
Miklos
Looks like the pbx isn't sending any info such as called exten
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lee Redmayne
Sent: Friday, May 07, 2004 12:13 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PRI, multi D
James Moran wrote:
We need to have about 30 phones on one floor
I have seen a couple of test where people claim that wi-fi phone network
should use max. 5 simultanoues calls per accesspoint or your audio will start to
break up. I would take a look at www.kirk.com. They have a DECT
That does work, I use that same approach to get analog extensions in a
norstar system to dial a specific sip phone in *. Works really well. We
then also tie the calleridname to which channel they dial out from as
well.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
P.S. I can send examples of needed also.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Wilson
Sent: Friday, May 07, 2004 12:34 PM
To: [EMAIL PROTECTED]
I am surprised you needed to turn the rxgain down so much, usually it is
just the opposite. I experienced the same problem you did when my txgain
was too low.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
-Original Message-
From: [EMAIL PROTECTED]
On Fri, May 07, 2004 at 12:01:02PM -0600, Jerimiah Cole said:
Chris Wilson wrote:
I want to run different lines directly to different extensions on two
FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to
extensions 102
You should be able to stick different channels into
Title: MGCP Problem
Turns out this was a typo in my extensions.conf file all along. Many thanks to the person who pointed it out. The answer was staring me in the face the entire time, but I just couldn't see it. Apologies to all
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