On Mon, 2005-03-28 at 18:18 -0800, Derrick Knight wrote:
I have got my Asterisk server running with TDM400 card (2xFXO 2xFXS).
I originally had the system configured with a Panasonic fax machine on
one of the extensions. Due to the high volume of fax spam, I figured it
would be a much
Hello all,
I hope this is not off-topic, if it is please let me know.
I'm currently playing with an Asterisk at home, in order to get to know
it's ins and outs. Very very impressive indeed. I've got it hooked up to
my home phone line via a Wildcard clone board (Intel modem with Ambient
chipset),
Title: Spandsp compilation error
Hello everybody,
I'm trying to receive and sending faxes with asterisk using spandsp. But while compiling the spandsp0.0.2pre11 (tried also spandsp0.0.1), I get following errormessage:
gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c t4.c -Wp,-MD,.deps/t4.TPlo
Hi,
In a VoIP only environment, Asterisk has to use ztdummy
to have any chance of playing back understandable audio
files (without drops, hickups etc).
I have been using ztdummy to some degree of success, but
I also have a Wildcard TDM400P REV E/F Board 1 in the
Asterisk machine I'm using.
In article [EMAIL PROTECTED],
Andreas Sikkema [EMAIL PROTECTED] wrote:
Hi,
In a VoIP only environment, Asterisk has to use ztdummy
to have any chance of playing back understandable audio
files (without drops, hickups etc).
I have been using ztdummy to some degree of success, but
I
I have a potential client that wants to send many faxes simultaneously,
over E1 trunks.
How CPU intensive is spandsp's txfax? How many concurrent faxes could
be sent by a decent CPU (e.g. Xeon 3GHz) before timing starts to get
disrupted?
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] -
Andreas,
-Original Message-
I have been using ztdummy to some degree of success, but
I also have a Wildcard TDM400P REV E/F Board 1 in the
Asterisk machine I'm using. I'm not using this card for
anything at all, but I'm wondering how to set it up for
timing only. What do I have
Hi,
sorry for my h323 dumbness. VOIP provider terminates H323 calls - it can be
used as gatekeeper or gateway (they claim so). What option and what setup is
best to connect Asterisk to this provider ?
Any working examples ?
Thanks in advance,
regards,
Rob.
I have created a call file as shown in the files below. The number is
dialled and
connected (i.e. the call is placed to the PSTN) but it is immediately
disconnected and I get the following message on the console:
Starting Zap/3-1 at from-internal-custom,s,1 failed so falling back to exten
's'
Did you install libtiff libraries prerequisites before compiling
It may be an issue with your LD_LIBRARY_PATH as files do not seem to be found.
Selon Dennie Verstrepen [EMAIL PROTECTED]:
Hello everybody,
I'm trying to receive and sending faxes with asterisk using spandsp. But
while
Hi,
During tests with a IAX2/PSTN gateway I've been getting strange results for
processor idle time and load. I (re)search(ed) this issue for a while, but I
didn't get any good explainations. Can somebody help me?
I have several sites that rely on a central server for connection to the PSTN.
Hello all,
We recently configure an asterisk server to use with an VoIP provider
to make calls to a PSTN. We use (voipjet, nufone, diamond)
We feel that we haven't got the quality that we hope. Sometimes our
calls gets mute, or we feel communication cuts on our phone calls.
We have got an
I found my problem, I had installed out of date libraries of libtiff. Now it's
running. But thanks anyway.
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]
Verzonden: dinsdag 29 maart 2005 11:55
Aan: Asterisk Users Mailing List - Non-Commercial
On Tue, 2005-03-29 at 12:36 +0200, Ismael Gil wrote:
Hello all,
We recently configure an asterisk server to use with an VoIP provider
to make calls to a PSTN. We use (voipjet, nufone, diamond)
If you find the same problem with multiple ITSP's, then it may not be
them that is at fault.
I have a channel bank (TA750) and a PRI with 30 channels connected to a
TE405P, in the channel bank I have a extension to a fax machine, but it
doesn't work to send or receive fax.
There are any advice ?
Kind regards,
Miguel
___
Asterisk-Users
Hi
ne1 know if there has been recent (over easter) cvs changes to what happens
when nat =yes especially in relation to sip
some things seem to work for me that didn't before ;)
thanks
walt.
_
Express yourself instantly with MSN
I have prepared two new, not-final yet, releases of asterisk-oh323:
- 0.6.6-pre1 for Asterisk stable
- 0.7.2-pre1 for Asterisk CVS HEAD
They can be found at:
http://www.inaccessnetworks.com/projects/asterisk-oh323/download
Please try them and report problems at the bugtracker of
the channel driver
Thore a écrit :
Hi !
This work well for incoming calls, but not for outgoing call.
Those i call get the wrong number in the display.
Thanks to reply to list, not private
Thore
- Original Message - From: administrator tootai
[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Hi,
on Alias the badguy(tm) on the phone usually sound like Darth Vader thanks
to some cool device from marshall :-)
Is something like this possible with asterisk, or, asked a little more
generic,
can i somehow pipe an rtp-stream to an application via STDOUT and read
it back via STDIN?
Hello,
i am running suse linux enterprise edition of kernel version
2.6.5-7.97-smp, i have latest stable asterisk zaptel asterisk stuff
compile fines i have TDM400P card with 1FXS and 3FXO modules, every
time i probe with modprobe and issue ztcfg -vv commandit shows the
following errors:
also
why not using a IAX phone, is running great on OS X
http://iaxclient.sourceforge.net/iaxcomm/
··
Adrià Vidal
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To
Hi,
Im the astertest guy.
If you are referring to the graphs on page 41 - 42, please note that
those are done on a embedded via 800mhz cpu and not on a system similar
to yours. So i'm pretty sure you can do more than 10 speex encodings at
the same time. (also some things changes since we did those
Hi!
I have just installed Redhat 9 and Asterisk to my computer, and now i have
problems with my non-zaptel Card, I don't know how to set it up since all
instructions are for digium's hardware.
I have searched from the Internet for hours now, can you help me to understand
all this HFC-s thing and
I like your idea, Ill play with it for a while and see what comes out.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Martes, 29 de Marzo de 2005 12:36 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I guess I should have included more information to get assistance after
reading what I posted,
I am running MF not SS7, the local Telco does see me sending the appropriate
info but not in the correct protocol according to them. They say it looks
more like I am dialing the dial code and CIC code
On Tue, 29 Mar 2005 11:33:16 +, Andreas Anderson
[EMAIL PROTECTED] wrote:
Hi,
on Alias the badguy(tm) on the phone usually sound like Darth Vader thanks
to some cool device from marshall :-)
Is something like this possible with asterisk, or, asked a little more
generic,
can i somehow
On Mon, 28 Mar 2005, Jason Miller wrote:
Has anyone ever setup Asterisk to pass Feature Group D access while using a
CIC code for outbound calls? If so can you please email the configuration
you have done? I have tried to get this up and running but with no luck. I
have also contacted support and
Thank you for your story Paul, nice work with the dialplans!
I have one question, so you say that for server 2, asterisk is behind nat
and you have sip clients inside and outside the nat. Which ports are you
forwarding to asterisk from your firewall and in the case of sip clients
outside nat,
Hi!
firefly(iLBC) works over dial-up connection. also u can use x-Lite (GSM/iLBC).
C ya,
Madhawa
On Tue, 29 Mar 2005 16:26:13 +1200, Matt Riddell
[EMAIL PROTECTED] wrote:
Kerry Garrison wrote:
This is what I get:
speex - - - - - - - - - - -
Do you remember what you actually changed to make it work cause that is the
same switch that I am dealing with myself if I am not mistaken.
Thank you,
Jason Miller
From: Dave Weis [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
I've been trying to setup SMS on asterisk - would be useful to have for
things like server outages, email from important customers, etc.
I can send SMS with no issues, although I have to send it over the Zap
line.. none of the VOIP providers will route the call. It arrives on my
mobile
I am looking for a step-by-step on adding H323 to [EMAIL PROTECTED]
So far I have installed [EMAIL PROTECTED], upgraded to the CVS-HEAD and followed
instructions according to voip-info and this list's archives. I keep getting
critical errors on compilation of H323, both Open 323 and OH323.
Has
Greetings to
everyone!
I am new to
Asterisk and ISDN modules so i tried to follow many of the
articles about
capi, bristuff, mISDN and so on.
Now I am working
in mISDN but every way I try I have compiling errors!
The HFC PCI card
is a Digi Datafire Micro V.
The bristuff give
me the
Does anyone on-list have any advice about the version 2.2 firmware for
the Zultys 4x5 phone? It has a new gateway mode that is supposed to
direct calls on the analogue line to the PBX for VM, etc. Zultys has
not yet presented any docs, and the keep refering me to the reseller
who doesn't know
For all my PBX installations I want to have Fail Over on the main incoming
PSTN line so that a power outage does not leave the offices stranded. Is
there any commercial solution to this? I would rather a finished product
than a home soldering project.
Chris Mason
[EMAIL PROTECTED]
Box 340, The
On Tue, March 29, 2005 3:48 am, Jan Henkins said:
it would be nice to know wether Voicetronix products are on the
trusted list or not.
I've got an OpenSwitch6 working in a development (soon to be production,
fingers crossed, box); never had such a lockup. Have you sent any queries
to [EMAIL
Title: No D-channels available!
Hi all,
I´ve ran into a problem regarding D-channels. I have setup Asterisk (CVS-HEAD-03/21/05-16:41:57) on RH Fedora Core 3(2.6.10-1.770_FC3) I also have a digium wcte11xp card for connetivity to the PSTN(E1). When I start zttool i see that Current Alarms
I'm having an issue sending DTMF to cisco
dialing this extension I should hear the dtmf tone
RTP playload 101 has been sent to the cisco phone, but no audio.
in the dialplan
exten = 8603,1,Answer(1)
exten = 8603,n,sipdtmfmode(rfc2833)
exten = 8603,n,SendDTMF(1|100)
exten = 8603,n,hangup()
sip.conf
Hi,
Look at this page page http://www.junghanns.net/asterisk/downloads/
and get the latest version of bristuff which should be :
http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC7k.tar.gz
This package is specific for BRI adapter using the cologne chipset. It
works great !
Best
Hello Paul,
On Tue, 2005-03-29 at 08:04 -0500, Paul Dugas wrote:
I've got an OpenSwitch6 working in a development (soon to be production,
fingers crossed, box); never had such a lockup. Have you sent any queries
to [EMAIL PROTECTED] Ben's been very helpful when I've had
troubles. He's
There's many solutions.. One being www.voiceguard.com I think might be what
you want.
- Original Message -
From: Chris Mason [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, March 29, 2005 8:01 AM
Subject: [Asterisk-Users] Fail over
For all my PBX installations I
Just Look at the missed calls screen and exit, the counter will clear.
Not sure on an IP300, but on an IP600 fastest is to press down arrow
then left arrow.
On Mar 28, 2005, at 6:32 PM, Paul Hales wrote:
1. You can set up items in the Digitmap (under SIP conf) to know when
a number is
On Mon, 2005-03-28 at 15:04 -0800, Jay Ray wrote:
Hi,
I am running Asterisk on Fedora Core 3. I am trying to use DDD to
debug Asterisk. However, when I attach the debugger to the Asterisk
Process, the Asterisk CLI promt hangs. Does not give any output, and
Asterisk stops processing
On Tuesday 29 March 2005 14:08, Rikard Westlund wrote:
[...]
When I start Asterisk(asterisk -vc) I get this:
Mar 29 15:02:15 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No
D-channels available! Using Primary on channel anyway 16! == Primary
D-Channel on span 1 down
[...]
I'll
Matt.
I gave your ideas a try and made it work with a twist. Use a macro but...
Here is the good part, call the macro from a call file using application,
passed parameters like name of the sound file, telephone to call, etc.
Voila! Works great!
Thx for the hints Matt.
-Original
Nope! that I have checked.
Rikard
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bob Goddard
Sent: den 29 mars 2005 15:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No D-channels available!
On Tuesday 29
No, that's a service, or at least I think it is, the sales garbage obscures
what it really is so who knows.
What I need is a little box that diverts calls if the PBX goes down.
Chris Mason
[EMAIL PROTECTED]
Box 340, The Valley, Anguilla, British West Indies
Tel. (264) 497-5670 - Cell: (264)
As soon I do a reload I see contant ringing like this
on the CLI:
-- Zap/14-1 is ringing
-- Zap/23-1 is ringing
-- Zap/22-1 is ringing
-- Zap/20-1 is ringing
-- Zap/19-1 is ringing
-- Zap/14-1 is ringing
-- Zap/23-1 is ringing
-- Zap/22-1 is ringing
-- Zap/20-1 is ringing
-- Zap/19-1 is ringing
-Original Message-
From: Zoa [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 29, 2005 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Erratic CPU load
Hi,
Im the astertest guy.
If you are referring to the graphs on page 41 - 42,
On Tue, 29 Mar 2005, Jason Miller wrote:
Do you remember what you actually changed to make it work cause that is the
same switch that I am dealing with myself if I am not mistaken.
Approximately line 1673 in chan_zap.c, it looks like this:
if (p-sig == SIG_FEATD) {
Could anyone explain to me what is the difference between Call-ID and
UniqueID of SIP calls, please?
Which one could be used as an ID to trace, for example, the status of a call
with Manager API and PHP?
Thanks,
Alex
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Asterisk-Users mailing list
On March 29, 2005 08:40 am, Richard Reina wrote:
This goes on continuously and no phones are ringing.
I am using a digium T1 card and ADIT 600.
Do you have the Adit600 configured correctly? It's not stuck in a test mode
or anything?
-A.
___
On Tue, 29 Mar 2005 06:31:11 -0600, Jason Miller [EMAIL PROTECTED] wrote:
Do you remember what you actually changed to make it work cause that is
the
same switch that I am dealing with myself if I am not mistaken.
Thank you,
Jason Miller
From: Dave Weis [EMAIL PROTECTED]
Reply-To: Asterisk
Hello David,
I've spent many hours to make my 2 Fritz PCI v2 work with Asterisk :-)
I was not able to make them work with the fcpci drivers (even with
custom driver modifications).
The solution was to use mISDN (with chan_capi) instead of fcpci.
You have a guideline at
On Tuesday 29 March 2005 14:40, Rikard Westlund wrote:
Nope! that I have checked.
1. Double check
2. Change the D channel to be 24 and retry
3. Cycle all channels through all possibilities.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bob Goddard
Hey Everyone,
I bought a Cisco 7970 Color IP phone. I wanted to reset it back to
factory defaults. I went through the sequence of holding down the pound
key when the unit is powereing on and then when the sequence changes to
press 123456789*0#. The phone seemed to do something different after
Basically, I'm forwarding the standard Asterisk ports:
tcp 5060
udp 5060
udp 4569
udp 5036
tcp 5038
udp 5038
udp 1:2
I'm not sure that i needed both tcp and udp on the mgmt port 5038, but what
the heck. :)
In sip.conf:
externip = xx.xx.xx.xx
localnet=192.168.1.0
In the sip client
Thanks Dave,
I will strip out the macro and give it a try. It did appear from the
log that the macro was being called and the last command was rxfax.
The sending fax shows that it is getting connected, and then fails to
send the fax with a message that the receiving fax did not respond and
no
Do you have the Adit600 configured correctly? It's
not stuck in a test mode
or anything?
I have no idea if it's configured correctly. We just
kind of hooked it up when the install was done a
couple months ago.
-A.
___
Asterisk-Users mailing
Mike Miller wrote:
I'm not sure where to start even -- It seems that the problem is with
the response to the digest authentication, but I'm not sure how to fix
that. The log below is from linphone, but I see the exact same thing
with kphone and xten from a indows box as well.
You are right
On Tue, 29 Mar 2005 09:40:08 -0400, Chris Mason [EMAIL PROTECTED] wrote:
No, that's a service, or at least I think it is, the sales garbage obscures
what it really is so who knows.
What I need is a little box that diverts calls if the PBX goes down.
The Sipura 3000 does this. That is what
Stewart Nelson wrote:
I never get such good support for commercial software, even on high-end
packages that charge an arm and a leg for maintenance.
Many thanks to Mark, Kevin, and the Asterisk team.
Thanks for the kind words, we appreciate it!
___
Any problems with RTP or voice just on one side?
So as long as you use some STUN server, the RTP packets have the right IP.
Did you install your own stund or are you using a public one?
You didn't have to use SER at all right?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I am using the manager API for "show channels".
If I have a multi line phone extenstions 510 - 515
and 510 has a call on hold and 511 has a call on hold
and I am answering 512 the manager API show channels
doesnt seem to tell me that 510 and 511 are on hold?
They are reported as Up.
How do I
Alex wrote:
Could anyone explain to me what is the difference between Call-ID and
UniqueID of SIP calls, please?
Which one could be used as an ID to trace, for example, the status of a call
with Manager API and PHP?
The Call-ID is internal to the SIP protocol, and not exposed inside
Asterisk (or
Since signalling info is carried in the A/B bits which is how * talks
to the Adit regarding the state of each channel, any framing misconfig
or timing misconfig will cause this.
Perform a print config on the adit and closely compare with
zapata.conf and zaptel.conf
On Mar 29, 2005, at 8:02
Title: Partially receiving a fax
Hello everybody,
I've succesfully installed spandsp and libtiff4 on my Debian linux platform. I want to receive faxes on my Asterisk server through a tdm10b PCI card. But when I send a fax to Asterisk I get following output from Asterisk and I only receive
Hello
Just installed fresh Debian testing box, checked out Asterisk and others
from CVS stable (-r 1.0), and now trying to 'make install' in Asterisk.
I get this error:
if [ -d CVS ] ! [ -f .version ]; then echo CVS-v1-0-03/29/05-15:19:53
.version; fi
gcc -g -o asterisk -Wl,-E io.o sched.o
WHen you say cannot communicate you mean it keeps giving you a busy
signal when you try and dial?
and could you post ur sip.conf along with the messages asterisk prints out.
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I get this error:
if [ -d CVS ] ! [ -f .version ]; then echo
CVS-v1-0-03/29/05-15:19:53
.version; fi
gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o
config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o
ulaw.o alaw.o callerid.o fskmodem.o image.o
The fact is, there is not ONE sip or iax softphone that is as easy to
use as skype for the average user. The sad thing is it doesn't have
to be that way.
Spend the $100 and get her a IAXy that's pre configured to your local
Asterisk server. Then she can use an analog phone to call you for
Ok, that was straight from the wiki. Still does not work, I tried it
from the iax.conf, etc files and it works just fine. I even tried
terminating/placing calls on the same server with realtime and it works
fine. Is realtime broken? Is there anything else I can test with?
Thanks, Matt
You need to have your openssl development package installed. It's trying
to link to librairies that are not availables.
Fred Blaise wrote:
Hello
Just installed fresh Debian testing box, checked out Asterisk and others
from CVS stable (-r 1.0), and now trying to 'make install' in Asterisk.
I get
Hi ALL,
I have downloaded app_chanspy.c and chanspy_sounds.tgz.
But I haven't found any instructions on how to compile and where to untar these files...
I tried to put the .c file on asterisk-src/apps and remake asterisk, but it seems it was not enough...
Thank you!Dov
No, that's a service, or at least I think it is, the sales garbage obscures
what it really is so who knows.
What I need is a little box that diverts calls if the PBX goes down.
FYI, the topic has been discussed previously on the list, and the
problem that you're trying to address is far
Has anyone come up with a way to get power to a TDM400P card installed
in a Dell PowerEdge 1750?
Thanks,
Adam
The contents of this email message and any attachments are confidential and are
intended solely for addressee. The information may also be legally privileged.
This transmission is
Adam Robins wrote:
Has anyone come up with a way to get power to a TDM400P card installed
in a Dell PowerEdge 1750?
I've not tried, but based on what I see in my 1750s, I would say 'good
luck'. There are no drive power connectors anywhere, and you can't steal
power from a fan connector because
Has anyone come up with a way to get power to a TDM400P card installed
in a Dell PowerEdge 1750?
The TDM card only needs the external power connector if fxs modules
are installed. The fxo modules don't use it that power.
If fxs modules are present, only the 12 volt lead is used. Therefore
I thought the TDM was broke on 1750's...?? I could never get passed
that NMI issue.
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 29, 2005 9:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dell 1750
My backup plan is to use an Adtran Express 3000 to analog and then a
Digium card but I'm not sure I can preserve the signaling for the
centrex features. I guess that's a cheap way to try this if I can't find
a reasonably prices ISDN card.
Brian
On Mon, 2005-03-28 at 15:52, Kevin P. Fleming
Matt Schulte wrote:
Ok, that was straight from the wiki. Still does not work, I tried it
from the iax.conf, etc files and it works just fine. I even tried
terminating/placing calls on the same server with realtime and it
works fine. Is realtime broken? Is there anything else I can test
with?
On Tue, 29 Mar 2005 07:37:24 -0700, Kevin P. Fleming
[EMAIL PROTECTED] wrote:
Mike Miller wrote:
I'm not sure where to start even -- It seems that the problem is with
the response to the digest authentication, but I'm not sure how to fix
that. The log below is from linphone, but I see the
I'm hoping Asterisk can help me solve an unusual problem.
I need two SIP endpoints (VoiceXML gateways) to transfer DTMF tones to
each other. Both of them can detect DTMF according to rfc2833.
However, one of them (host2) must generate DTMF inband.
Happily, I came up with the following sip.conf
Mike Miller wrote:
Based on what you wrote -- I'm using type=friend, not type=peer. This
should be ok, though, correct? (As friend == peer + user, right?)
Yes, type=friend is fine.
sip.conf:
[general]
context=default; Default context for incoming calls
realm=192.168.1.100; Realm for digest
Has it been updated for AMP 1-10-007a?
Or manual update is required?
Thanks
Robert
Btw: great work!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, March 28, 2005 8:36 PM
To: asterisk-users@lists.digium.com
On Tue, 29 Mar 2005 09:01:37 -0700, Kevin P. Fleming
[EMAIL PROTECTED] wrote:
Mike Miller wrote:
Based on what you wrote -- I'm using type=friend, not type=peer. This
should be ok, though, correct? (As friend == peer + user, right?)
Yes, type=friend is fine.
sip.conf:
[general]
Short of finding somewhere to tap 12v off the board that 1) would'nt
make the danged thing beep and 2) voiding the warrantee cdrom??) , I'd
just juryrig an external 12v supply along the lines of
http://www.soekris.com/PowerAccessories.htm.
I'm assumong the tdm400p only taps the 12V for RI and
Le mardi 29 Mars 2005 18:13, Parker, Blake (MIS) a écrit :
What is the command to create a new voicemail box?
addmailbox in /asterisk_directory/contrib/scripts
Blake
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yes
--- Matt [EMAIL PROTECTED] wrote:
Web Meetme is now installed by default and the
meetme2 application is no longer needed.
What does this mean exactly? Does this use the
regular meetme as
opposed to the meetme2 we had to setup before?
On Mon, 28 Mar 2005 17:35:37 -0800 (PST),
Mike Miller wrote:
1.0.6 from an ubuntu package. I'd also tried a version compiled from
source, but with the same results.
I tried taking out username, but it didn't help.
OK, then we need a _full_ log, with:
- sip debug
- set verbose 255
- set debug 255
There should be (at least) a message on the
Anton Krall wrote:
Any problems with RTP or voice just on one side?
So as long as you use some STUN server, the RTP packets have the right IP.
Did you install your own stund or are you using a public one?
You didn't have to use SER at all right?
Setting nat=yes does pretty much the same as a STUN
I'm sure some users would use it. Once it's done post
it and I'll add it to [EMAIL PROTECTED]
Also is it possible to do it without MeetMe2? The new
WebMeetMe from Areski uses the normal conferencing app
and this is much cleaner and simpler than meetme2.
Also is it posible to do it without
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Latham
Sent: Sunday, March 27, 2005 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] small qos switch
I heard a great solution at Linux
No. But 0.8 will be out soon with AMP 1-10-007a and
some other fixes and features.
--- Robert Augustyn [EMAIL PROTECTED] wrote:
Has it been updated for AMP 1-10-007a?
Or manual update is required?
Thanks
Robert
Btw: great work!!
-Original Message-
From: [EMAIL PROTECTED]
Does the backup feature preserve enough info so that I won't have to
rebuild all of my extensions/etc?
JD
[EMAIL PROTECTED] wrote:
yes
--- Matt [EMAIL PROTECTED] wrote:
Web Meetme is now installed by default and the
meetme2 application is no longer needed.
What does this mean
Just a follow-up to my message.
I hope I didn't come off as negative about voipconnection. They're a
great crew over their, and they defintely know their stuff :) Give them
a look because I think you'll be happy
Andy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
While researching Areski's new Web-MeetMe management gui,
I found some odd (from what I expected) behaviour). Using
the CLI to set un/mute status works but does not update the
flags, or so it appears.
Starting with a fresh conference (1 user)
*CLI meetme list 3456
User #: 1 Channel: OH323/R61
Incoming however just isn't working. I've got a nice list of numbers
from which SMS messages come:
snip
You are sending the extra digit to say which mailbox the message is
for, right? In this country, if you do not send that digit, it will
try to vocalize the message during the calls.
On Tue, 2005-03-29 at 16:11 +0200, Oga wrote:
I've spent many hours to make my 2 Fritz PCI v2 work with Asterisk :-)
I was not able to make them work with the fcpci drivers (even with
custom driver modifications).
The solution was to use mISDN (with chan_capi) instead of fcpci.
You have
Any possibility to support a zero extension and operator extension
automatically in the Auto-attendant?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, March 29, 2005 9:33 AM
To: Asterisk Users Mailing
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