It might be nice if safe_asterisk (or some part of asterisk) e-mailed a
backtrace of the last asterisk .core file to Digium so they can see what
causes Asterisk to core dump. I've not had asterisk crash in that way,
but it might be nice for Digium.
On Tue, 6 Apr 2004, Mark Spencer wrote:
Eric Wieling wrote:
On Tue, 2004-04-06 at 12:29, Brian Rathman wrote:
Does anyone know how I can pass dtmf digits from a SNOM 200 to a cisco
AS5300 with * in the media stream. Unfortunately, the only way I can get the
calls to connect is with t or T at the end of the Dial() statement and then
...or the problem is, as hinted, that Asterisk sends a short dtmf.
This is a bug in 7.2 release.
It's setting the duration of RFC2833 digits to as low as 30 msec which is
definitely not enough.
To fix, change the following in rtp.c, function ast_rtp_senddigit()
/* Make
Hi,
-Original Message-
Now, of course, any time you put a call home feature in,
there are people who will be concerned about privacy.
Clearly it will be able to be disabled, but I want to run my
idea about deployment by everyone here and see if you guys
had some ideas. The
Marian Durkovic wrote:
...or the problem is, as hinted, that Asterisk sends a short dtmf.
This is a bug in 7.2 release.
It's setting the duration of RFC2833 digits to as low as 30 msec which is
definitely not enough.
To fix, change the following in rtp.c, function ast_rtp_senddigit()
Hi all,
i have an ISDN Phone connected to an HFC-S based card, all works fine but is i
call the Phone from a SIP User Agent or over PSTN Line the Phones Display
shows the correct CallerID but with a leading 0 .
I cant find this in the config files, how can is solve this?
Dialing Out with the
Hi
a) The idea itself -- is it a good one or is it stupid?
great idea. could be very useful if you don't have much time
to track/test cvs version and/or the bugtracker
b) The way to make it deployed without sneaking a call home in on
anybody that doesn't want it?
make it off by default,
I'd like to give this one 10 thumbs down. IMHO a bad idea, a nasty little bad idea..
evil, spawn of Satan. If this were implemented the first job of a new update would be
to rip it out and flush it down the nearest toilet.
I can only wait until we see M$ like activation implemented... oh the
On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote:
download ethereal and take a peek at the packets on the wire. Without
something like that, no one is really going to be able to help you.
Do you mean then that my SIP trace displayed at kphone looks otherwise OK --
that the REGISTER
On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote:
download ethereal and take a peek at the packets on the wire. Without
something like that, no one is really going to be able to help you.
Do you mean then that my SIP trace displayed at kphone looks otherwise OK --
that the
Alex Lopez wrote:
What is the difference b/w
USE_MYSQL_FRIENDS=1
and
USE_SIP_MYSQL_FRIENDS=1
Not sure ;)
Am I to think that this replaces the entrys in sip.conf for the
registering clients??
Yes
If so, I am hosed as I cannot get a ATA-186 to register via MySql, but
if I leave the config in
On Wednesday 07 April 2004 09:24, Richard Airlie wrote:
On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote:
download ethereal and take a peek at the packets on the wire. Without
something like that, no one is really going to be able to help you.
I thought I'd chime in here with a
Interesting idea, but needs some refinement...
Very few Asterisk installations are alike. I have a couple of FreeBSD asterisk
installations without any zaptel stuff, without ISDN, without MGCP, Skinny and
a lot of other modules stripped out. If any of those modules have a MAJOR bug,
it's not my
Hi all,
In myAsterisk setup, incoming calls through
Cisco PSTN gateway to Asterisk extensions sounds work fine. All calls can be
terminated properly after hangup. However, when callswere forwarded to
voicemail, after recording hangup the PSTN callsand ciscoFXO
port remained
Hi,
Asterisk either need to know when the remote caller ends his call,
or it must detect the silence.
Simplest solution is to activate silence detection, see
voicemail.conf.
You may need to do some testing to get the proper
silencethreshold setting.
Also search the archive, this is a often
Title: indications.conf for Portugal
Does someone have the settings for 'indications.conf' in Portugal?
Thank you,
Pedro Goncalves
--
Pedro Goncalves
PT Inovação SA - Pólo do Porto
Largo de Mompilher, 22 - 4º
First pass through the trace indicates all udp packets originating from
194.200.209.137 have incorrect checksums. However, the asterisk machine
acknowledged the initial register packet with a 100 Trying, therefore
it must be ignoring udp checksums. (Still curious why incorrect checksums
are
I got this compiling the new cvs code ...
any idea ?
Tnx !
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
-DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/07/04-11:28:50\
Morning Asterikians,
I've just got my nice shiny quadBRI card, and it seems to be working
very well - except for one little issue - CallerID.
The card is currently connected to an ISDN2e line in P2P mode, and an S0
adapter on our existing alcatel PBX. The S0 connection recieves callerID
and
I've just got my nice shiny quadBRI card, and it seems to be working
very well - except for one little issue - CallerID.
The card is currently connected to an ISDN2e line in P2P mode, and an S0
adapter on our existing alcatel PBX. The S0 connection recieves callerID
and displays it correctly
Hello * users
I try to get SIP flashhook transfer to work properly in my setup. The
problem is that when I flashhook and then dial another extension I get some
really garbled sound in the end I flashhook from. The remote can hear me
just fine, I have threewaycalling=yes and transfer=yes in my
Linus Surguy wrote:
I've just got my nice shiny quadBRI card, and it seems to be working
very well - except for one little issue - CallerID.
The card is currently connected to an ISDN2e line in P2P mode, and an S0
adapter on our existing alcatel PBX. The S0 connection recieves callerID
and
Hi,
just to repeat my previous post (and trying to find a solution):
Setup is * behind NAT.
I can use FWD (time service, echo server) without problems when I add
this to sip.conf:
externip=a.b.c.d; a.b.c.d is the IP of the router (Linux/Nat)
outside_addr=a.b.c.d
My ICH however now
Yes!,
The latest CVS has fixed this problem.
Thanks for the help.
Sean
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, April 06, 2004 6:53 PM
Subject: Re: [Asterisk-Users] voicemail-hangup issue
I have a small * installation with 2
Andreas,
below is my partial sip.conf (which is relevant for fwd)
this works for me.
jakob
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
nat=yes ;
externip = myhost.dyns.net ; Addr put in SIP messages if we're behind
a
excellent idea.
eliot
On Tue, 2004-04-06 at 23:31, Mark Spencer wrote:
I've been considering the nature of Asterisk, its security, the bug
tracker, and more... And i've come up with an interesting idea: A
message of the version. The idea is that Asterisk has a compile
time
32-bit unsigned
Mark Spencer [EMAIL PROTECTED] wrote:
I've been considering the nature of Asterisk, its security, the bug
tracker, and more... And i've come up with an interesting idea: A
message of the version. The idea is that Asterisk has a compile time
[...]
a) The idea itself -- is it a good one or is it
Mark Spencer wrote:
Any feedback on:
a) The idea itself -- is it a good one or is it stupid?
Now this is just my views. No I do not feel we need to be sending any
information back unless we want to. Like someone else said a sub job that is
turned off by default. My preference would be no
Robert Hajime Lanning wrote:
quote who="Andrew Thompson"
I regret that I've only used MeetMe a few times, and only up to two users.
Well, the problem with giving general stats, is that it REALLY depends on the
exact environment.
Key points: (on a server dedicated for
I wouldn't want a call home feature that is enabled by default.
I think it would be great though if * had some ability to update itself.
Perhaps via a CLI command, as others have suggested. Something similar to
RH's up2date would be great in my opinion.
Anyway, thats my 2 cents.
Sean Rodger
Andrew Kohlsmith wrote:
Are there open problems/issues with iax2 and jitter (quality)?
Just upgraded to today's dev cvs about an hour ago, and it seems the iax
conversations are lower quality then a month or two ago. iax2 show firmware
says version 13. (Test call
Hi all,
Has anyone got any experience with hooking Asterisk up with a
Siemens EWSD 13 switch over a E1/PRI ?
We're located in Belgium (Europe) and one of our telecom partners
uses this switch.
We connected one of our TE410P ports with their switch, but the status
light on the TE410P card keeps
See :
http://sourceforge.net/mailarchive/forum.php?thread_id=4096021forum_id=293
80
SourceForge reports invalid forum
-A.
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I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite
some time, to no avail. I've googled, I've tried loads of configurations, I've
rewired phone lines, and still I am not winning the battle.
Here's my config.
PRI-T400P-Asterisk-T400P-Adtran 750(L36
Andy Powell Wrote:
I'd like to give this one 10 thumbs down. IMHO a bad idea, a nasty little
bad idea.. evil, spawn of Satan. If this were implemented the first job of
a new update would be to rip it out and flush it down the nearest toilet.
Just curious, but why does it strike you as such a
On Wed, 7 Apr 2004, Steve Kann wrote:
If people are looking for a higher-capacity conferencing application,
take a look at app_conference, in the iaxclient (on sourceforge) CVS.
I haven't really benchmarked meetme, but I _think_ that app_conference
might be able to beat it. Certainly in
Bisker, Scott (7805) wrote:
I've been trying to get a Win 2000 RAS server working with my
asterisk PBX for quite some time, to no avail. I've googled, I've
tried loads of configurations, I've rewired phone lines, and still I
am not winning the battle.
Here's my config.
Title: IAXTel toll-free gateway
Is anyone else having trouble placing toll-free calls though IAXTel lately? Mine just stopped working yesterday, yet I seem to be able to make 1-700 calls.
-brian
1-700-676-3830
Is it possible to detect the attempt to dial to unallocated (unassigned)
numbers ? Currently I cannot distinguish the error from no-answer.
There is a extension with priority n + 101 but it is not used.
The dialplan extension looks like:
Dial(CAPI/${num1}:B${num2},30,T)
If I use lowercase
Hi all,
We keep getting these and all the calls between these two asterisk boxes get
dropped. what is going on here, I have been trying to solve this problem on
my own but maybe I don't have the trunk setup right. also I have posed the
output of my full log of the machine with the zap
Hi,
use prilocaldialplan=local in zapata.conf.
--
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
I have a system with no Digium hardware in it (two others with 2 X100P
cards in each of them as well). I'm interested in using MeetMe in the
one without the hardware (it works great in the ones with the
hardware). I can't use ztdummy, because the system has usb-ohci
drivers, rather than
Today I updated my asterisk cvs, I would love to set up mysql support
for voiceboxes.
Here is what has happened (by now):
cvs does not compile as downloaded, module chan_oss reports an error;
if I compile with -i all works fine except for chan_oss.
Ok, time to set
USE_MYSQL_VM_INTERFACE=1 in
Is anyone else having trouble placing toll-free calls though IAXTel lately?
Mine just stopped working yesterday, yet I seem to be able to
make 1-700 calls.
It's up/down/etc rather frequently, so no surprise. Good thing it's not
a paid service or we'd all have an issue. Consider it as a
Four or five analog lines can be done with a single computer so no channel
bank is needed. If you need 6 or more than there is also the choice of using
two machines and IAX.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Hajime
Lanning
Sent:
Same as mine. Do you know off the top of your head what firwmare you're using? Also,
what RAS card do you have on your PCAnywhere side?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ariel Batista
Sent: Wednesday, April 07, 2004 10:39 AM
To: [EMAIL
On Tue, 2004-04-06 at 23:31, Mark Spencer wrote:
Any feedback on:
a) The idea itself -- is it a good one or is it stupid?
I like the idea of being able to see what updates/fixes are available
vs. the code that I'm running. I think this would definitely be helpful
to me.
b) The way to make
Hi.
another (stupid) thing.
don't call that function motv. motv is a name
for another opensource project.
Matteo.
--
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web : http://www.espia.it
Phone : +39 02 70633354 - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Bisker, Scott (7805) wrote:
Same as mine. Do you know off the top of your head what firwmare
you're using? Also, what RAS card do you have on your PCAnywhere
side?
I have firmware L36. Ras card is a Digikey 4 port board on one NT server
and others are using the normal serial ports on the
Which brand of card did you get?
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
http://www.dreamtime.net
http://www.emailblaster.us
Corporate Office
101 California Street, 22nd Floor
San Francisco, CA 94111-5802
Voice - 877-203-9308
Fax - 310-943-2606
Dreamtime is your global choice for
There are several ways to know what changes in Asterisk's CVS.
This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly
up to date CVS changelog summary information.
You can also sign up for the Asterisk-CVS mailing list at
http://lists.digium.com/mailman/listinfo/asterisk-cvs
Title: Message
Then
your firewall is closing the return RTPport to fast. Check for the
latest firmware and also make sure the SPI (stateful packet inspection) is
turned off if you router has that option. Otherwise you may have to give
up and fall back to port forwarding.
quote who=John Vogel
Four or five analog lines can be done with a single computer so no channel
bank is needed. If you need 6 or more than there is also the choice of using
two machines and IAX.
Talk about port density issues. So, if he really needs all 12 lines, then he
needs 3 PCs? (He
Hi!
I am looking for a software that can work as h.323 - sip gateway other than
asterisk and free. Someone can help me?
Thanks.
Mireia
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Hello Asterimaniacs,
Finally, I went out with that... sorry I had lot of work and not enough courage
to work at night ;)
Well, mysql and postgresql now work well for me and I have put some order in the
code.
Just enjoy it, I m waiting for the feedbacks ;)
Wow... talk about a detailed response; thanks!
In our situation, we've got a T-1 voice PRI from Allegiance Telcom. For
the benefit of those of us who aren't as in the know as you are (and
who have no affiliation with a CLEC), is there a way to be able to
control what gets sent out as our name
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
John Vogel wrote:
| Four or five analog lines can be done with a single computer so no channel
| bank is needed. If you need 6 or more than there is also the choice of
using
| two machines and IAX.
I assume you would be using 4 or 5 X101Ps (or
Hello,
As you can see are pri is being reset every 60 minutes! Is there a way to
stop this?? Is it a Zapata configuration problem?
We have a * box with a single port T1/pri card installed.
Thanks
lach
Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 2 successfully restarted
on span 1
Apr
Try Vovida's Vocal, i think it does it.
Mireia Munoz de jesus wrote:
Hi!
I am looking for a software that can work as h.323 - sip gateway other than
asterisk and free. Someone can help me?
Thanks.
Mireia
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The card is an ASUSCOM ISDNLink PCI (passive) and the circuit is from
Qwest (in the US). I will be using this circuit only for voice (I'm
doing this because of the poor quality of my POTS lines).
I've compiled Hisax (as a module) into my 2.4.25 kernel, and with
'modprobe hisax type=35
Hi,
If you didnt do it yet I'd suggest you start with simpliest thing which is
making loop on the cables. And testing the status. Simple RJ45 plug with
10cm of 2 pairs of cable crimped by 1,2 on 4,5.
When you make loop T410P goes green. EWSD also will see a loop.
Check cables and loops and when
I've got Asterisk loading 100,000+ extensions in extensions.conf. This
process is taking a little upwards of 10 minutes to complete on each of my
dual 3.2Ghz HP DL380 with SuSE Linux Enterprise 8 boxes.
Although asterisk creates child processes, it appears that it is only using
a single processor
I don't have a PocketPC PDA, mine is Palm. But regardless, I don't see
what all the Hype is with Skype. It is a closed protocol and highly
platform-restricted product. Sure the concept of a peer-to-peer phone
network is interesting, but if not everyone can connect to it, what is
the point? If
Eric Wieling wrote:
There are several ways to know what changes in Asterisk's CVS.
snip
You can also sign up for the Asterisk-CVS mailing list at
http://lists.digium.com/mailman/listinfo/asterisk-cvs
I've signed up for the cvs mailing list and have been stockpiling the
messages.
I
I cannot help you with the .conf files on the * as I am brand new to the
* and in the process of compiling the software now.
I do know this.. You have to make sure the the generic name IE
(information element ) is being populated in the outbound ISDN setup
message to Allegiance. If you have a
Does Asterisk work with Lucent or any other PBX phone systems
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Hello Lach-
I think B channel restarting is a normal occurrence, although I thought it
was supposed to be more often than once every 60 minutes.
The channels are not supposed to be restarted if they show in use, so this
should be a transparent occurrence.
Are there any problems that this is
Hi All
Just got a new quadBRI card and connected one port to our Old PBX.
When I make a call from a sip phone to a phone number the phone rings, I hook up, and
the call on the
sip phone allmost imidialely disconnects, after a fiew seconds the real phone
disconnects too.
Here is a trace:
--
i'm new with asterisk. i currently have 1 fxo port. my phone
line connected to the fxo is capable of 3 way calls using flash.
i'm thinking of a callback and then 3 way call from asterisk. is
this possible?
1. phone1 calls asterisk thru zap/fxo. asterisk gets callerid of
phone1.
2. asterisk will
osx wrote:
Hello,
As you can see are pri is being reset every 60 minutes! Is there a way to
stop this?? Is it a Zapata configuration problem?
We have a * box with a single port T1/pri card installed.
This is an expected and desired behavior.
Jeremy McNamara
I am getting this too under RH9.
Sean
- Original Message -
From: Alessio Focardi [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 07, 2004 9:04 AM
Subject: [Asterisk-Users] errror compiling asterisk from cvs
I got this compiling the new cvs code ...
any idea ?
Tnx
Title: RE: [Asterisk-Users] Lucent Phones
Absolutely, it can be a little tricky but its definitely doable. Check out the info I wrote on the wiki, as well as peoples posts here for more information on hows its done.
Matt
-Original Message-
From: James Moran [mailto:[EMAIL
I'm timing off my PRI from Verizon as well. This is mind boggling. All my Fax
machines are fine. The modems connect, but drop the calls after about 1-2 minutes
regardless of busydetect.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ariel Batista
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Subject: [Asterisk-Users] Lucent Phones
Does Asterisk work with Lucent or any other PBX phone systems
Sure. You can use Asterisk as a VoIP gateway to your existing legacy PBX.
You can't plug Lucent's
Hi Lach,
this looks like normal behaviour to me. Most of the equipment I use issues a
restart upon initial physical connection (bad equipment can cause problems
when it doesn't do this) and then several times per hour thereafter. Once
every hour seems infrequent but I guess that this is down to
One thing that the BSD open source operating system projects do, and
many other projects for that matter, which Asterisk does not seem to do, is
put CVS ID tags in the source files of the package itself. If ID tags were
put into the source files, and even embedded in strings so that
quote who=Darren Sessions
I've got Asterisk loading 100,000+ extensions in extensions.conf. This
process is taking a little upwards of 10 minutes to complete on each of my
dual 3.2Ghz HP DL380 with SuSE Linux Enterprise 8 boxes.
Although asterisk creates child processes, it appears that it
My FWD and ICH through NAT work just fine (for outgoing calls) depending
on the setup I choose. The setup is just mutually exclusive.
FWD needs:
externip=a.b.c.d; a.b.c.d is the IP of the router (Linux/Nat)
outside_addr=a.b.c.d; as per your config this is optional
ICH doesn't need
Bisker, Scott (7805) wrote:
I'm timing off my PRI from Verizon as well. This is mind boggling.
All my Fax machines are fine. The modems connect, but drop the calls
after about 1-2 minutes regardless of busydetect.
That was our exact problem with Sprint when we had there T1 line. We
decided
Brian == Brian Cuthie [EMAIL PROTECTED] writes:
Brian Is anyone else having trouble placing toll-free calls though
Brian IAXTel lately? Mine just stopped working yesterday, yet I
Brian seem to be able to make 1-700 calls.
I'd suggest using enum lookups on freenum.org instead.
Cf:
Alessio Focardi wrote:
I got this compiling the new cvs code ...
any idea ?
Tnx !
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
We have one other error (twice today) we get Out of trunk data space on
call number , dropping
How do I determine what is causing this error? we have a point-to-point
T1
between 2 * boxes, with 3 phone in the remote office. I have no idea how
the trunk could be out of space.
The
Eric Wieling wrote:
There are several ways to know what changes in Asterisk's CVS.
This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly
up to date CVS changelog summary information.
You can also sign up for the Asterisk-CVS mailing list at
I am trying to setup the call manager and I configured the manager.conf
file.
When I try to telnet 0.0.0.0 5038
It says trying 0.0.0.0
Connected to localhost
Escape character is '^]'.
Asterisk Call Manager/1.0
Then I type
Action:Login
I'm having the same kind of issues. We get the out of trunk data space
error consistently during conference calls between asterisk servers.
And occasionally on regular iax calls. Also while we're on a conference
call it seems to cause other calls going out through iax to fail and
also give
Hi
Il mer, 2004-04-07 alle 20:28, Bartosz Jozwiak ha scritto:
hello,
Is it possible to make attendant transfer (not blind) with ZAP channels ?
sure. just press the flash key on the phone (also known as the 'R'
key, at least in EU), you will hear the dialtone, while
the caller is put on hold.
Hi.
try adding a whitespace between ':' and the command.
Eg.
action: login enter
blah
blah
Matteo.
I am trying to setup the call manager and I configured the manager.conf
file.
When I try to telnet 0.0.0.0 5038
It says trying 0.0.0.0
Connected to localhost
I'd like to jump in here because we're also experiencing the out of
trunk data problem. So is this the only thing that causes the out of
trunk data error? Because we are running iax between the boxes and both
boxes have trunk=yes in the iax.conf entries and there is a zaptel
device in both.
Hi
I am also having jitter trouble on IAX2, and I can vouch
that the jitter buffer is busted.
On Wed, 07 Apr 2004 09:56:01 -0400
Steve Kann [EMAIL PROTECTED] wrote:
Andrew Kohlsmith wrote:
Are there open problems/issues with iax2 and jitter
(quality)?
Just upgraded to
Hi,
I presently have 6 PRIs of IVR traffic that I am planning to migrate from
Dialogic on SCO-Unix to Digium-Asterisk on Debian. Here is the general
description of the traffic in question :
- IVR system, 138 PRI channels (6 PRIs, multiple D-channel)
- Some traffic from TV ads, so all traffic
On Wed, 7 Apr 2004, Jain, Sonal wrote:
I am trying to setup the call manager and I configured the manager.conf
file.
When I try to telnet 0.0.0.0 5038
It says trying 0.0.0.0
Connected to localhost
Escape character is '^]'.
Asterisk Call Manager/1.0
What about the Partner phones and TDM400?
You can't plug Lucent's (Avaya's) DCP, MLX, or ATL phone sets into an
Asterisk box -- the protocols are all proprietary.
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I am planning an * install at my business. It will be replacing an
existing Toshiba system (I think it is a 424dk). I was wondering if
anyone knows of a way for me to use my existing Toshiba phones to
connect to *. I would rather not have to spend the $15,000 to replace
all of my phones, but I
dido
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, April 07, 2004 2:41 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with IAX2?
Hi
I am also having jitter trouble on IAX2, and I can vouch
that the
I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone
are registered by the below information
*CLI sip show peers
Name/usernameHost Mask Port Status
2002/2002192.168.22.199 (D) 255.255.255.255 5060 Unmonitored
2001/2001
Hey All,
We are using Asterisks as a voicemail only application, and
so far all is great. (Excellent product!)
However, I do have one question that I am hoping you might
be able to help me with.
In our asterisk application. When our customers call
*55 (our dialplan code to check
I just updated to latest cvs and the problem remains. I did also notice
that when the call coming in on the queue is through a Zap line (from an
adtran 750 to an x100p) it logs the following just before the warnings
below:
pr 7 14:21:21 VERBOSE[60194841]: -- SIP/hrutter-432b answered
Greetings,
Does anyone have any experience in getting dialpad.com working with *
They use a proprietary softphone but also have facility for cisco
ata-186 and Sipura SPA-2000.
Before I go off and investigate, I though I would check and see if
anyone has any experience with them
Thanks, Craig
I'm going to leave most of what you said alone, I understand you point
and it's your point to make.
However I will make a small comment about
I don't need hotlist functionality,
if I dial their number and they aren't on, I get a busy
reorder signal. No big deal.
Presence based
On Wed, 2004-04-07 at 17:20, Eric Wieling wrote:
There are several ways to know what changes in Asterisk's CVS.
This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly
up to date CVS changelog summary information.
You can also sign up for the Asterisk-CVS mailing list at
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