On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote:
sounds like one of those pesky auto dialers the simpsons make fun of.
It sure does...
--
Steve
__
You actually need to constantly be alert
and willing to handle things, or life
Hi,
[...]
There should also be a digitally signed version of that file
(cmterm_7920.*.sbn), which the phone probably requires.
nope. no sbn. according to my cisco source the file is not signed.
Funny, that would be the first phone with unsigned firmware.
But I'll double-check after the
On Thursday 15 January 2004 01:30 am, Chandra wrote:
hi i am not talking about * behind NAT. its * outside NAT and GS inside
NAT.
Why leave a host to defend for itself? At least behind a firewall you got some
layers of protection.
--
Steve
__
Jason T. Nelson wrote:
I have already started playing with trying to figure out why Asterisk runs
so badly under FreeBSD, such as eating 100% of the CPU without warning
unload pbx-wilcalu.so, see
http://www.voip-info.org/wiki-Asterisk+freebsd
/O
___
Dear all,
Is * capable to use qmail as a MTA?
If so, how can I set it?
Thanks
Isianto
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hi Robert,
Yes, this fixed the RTP issue for me. Do you need a bug note created on this
???
Cheers
SW
Date: Mon, 12 Jan 2004 14:03:14 -0800 (PST)
Subject: Re: [Asterisk-Users] How to bind RTP when IP alias are configured
From: Robert Hajime Lanning [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Christian,
There are a couple of images at http://snom.com/download/share. We are not
really happy with the latest image yet; hopefully we can fix the remaining
issues in a couple of days. Input appreciated (but no new feature requests
until we have this stuff stable!).
I could not find any image
Hello all,
Is there a way of setting the sound level at which * starts
to transmit silence. It appears when an external call comes in the caller
speaks silently you hear a lot of lost bits as it drops in and out. This only
seems to have been introduced when I upgraded to the latest
Hi
Has anyone managed to connect a GSM modem to Asterisk ?
I have a Siemens M20 Terminal that can do voice/fax/data and want to connect
it. Are there someone on the list that has experience with the Siemens M20
and Asterisk?
Terje
Hi All,
Have just checked kapejod's quadBRI specs and looks wonderful. I am not
an expert on ISDN either, but seems to me that features and functionally
worth the 600 EUR (almost US$600,right??) suggested price.
However, from * stand point it seems --pls pardon me in advance if I am
Hello,
two days ago *NISCC* released a security advisory reference about a
security vulnerability in H323 protocol.
http://www.uniras.gov.uk/vuls/2004/006489/h323.htm
According to Graig Southeren from OpenH323 development team: OpenH323
is affected by SOME of the problems Both asterisk
Hello,
I am trying to use * to handle anonymous ISDN callers.
Something like
exten = 5150/0,1,Congestion
should work but doesn't. Apparently because the ISDN CAPI doesn't
use 0 for callers who don't send their number.
Is there a way to make * identify ISDN callers who use CLIR?
-Walter
--
Walter Doerr wrote:
Hello,
I am trying to use * to handle anonymous ISDN callers.
Something like
exten = 5150/0,1,Congestion
use:
exten = 5150/,1,Congestion
should work but doesn't. Apparently because the ISDN CAPI doesn't
use 0 for callers who don't send their number.
Is
Hi all,
From Andy Powells Getting Started With Asterisk (V 0.1a)
http://www.automated.it/guidetoasterisk.htm
parking.conf file has this number set at 700. I've changed mine to
701 because I was having an issue with Asterisk - although it would
'see'
(looking at the console) I had tried to
Walter Doerr schrieb:
Hello,
I am trying to use * to handle anonymous ISDN callers.
Something like
exten = 5150/0,1,Congestion
should work but doesn't. Apparently because the ISDN CAPI doesn't
use 0 for callers who don't send their number.
Is there a way to make * identify ISDN callers who
Hello,
I think you need to do a little more looking around on the Asterisk
resources and on Google. What you are trying to do is mostly possible if you
have the time, patience and money to follow through with it.
One thing you need to learn is that a great many on this list despise
On Thu, Jan 15, 2004 at 02:45:22AM -0500, Steve said:
On Thursday 15 January 2004 01:30 am, Chandra wrote:
hi i am not talking about * behind NAT. its * outside NAT and GS inside
NAT.
Why leave a host to defend for itself? At least behind a firewall you got some
layers of protection.
By the way, I have an other question, are there any way to
implement MeetMe conference If I haven't zaptel device?
Zhang Peihao
2004-01-15
Yes, there is. Modify the Makefile in the /usr/src/zaptel directory,
ie, change #ztdummy to ztdummy and run
make clean; make install
Find more information
Is * capable to use qmail as a MTA?
If so, how can I set it?
It shouldn't be an issue, as qmail has the standard 'sendmail' binary
included.
Regards,
Andrew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
hello klaus-peter
this sounds great ;
will the phones that are connected to a bri in nt-mode still allow all
isdn-functions (in special : caller id-display)?
thanks...
- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January
hi everybody.
i'm currently looking for a good SIP-softphone. i
tried x-lite but i'm not happy with it's soundquality as there is a very high
noiselevel.
do you know a better softphone?
and would a hardware SIP-phone offer (almost)
the same audioquality as my isdn-phone?
if so what phone
Can skinny work behind NAT? I have a Cisco 7910 using SCCP behind NAT
that has one way audio. The called party cannot hear the calling party
who's using the 7910.
skinny.conf
;
; Skinny Configuration for Asterisk
;
[general]
port = 2000 ; Port to bind to, default tcp/2000
Is there someone here who could offer some assistance to a Linux noobie
in getting an mta configured for a new * server. Off list of course.
Michael
On Thu, 15 Jan 2004 07:14:00 -0500, Andrew Kohlsmith wrote:
Is * capable to use qmail as a MTA?
If so, how can I set it?
It shouldn't be an
Hi ,
I'm having a problem that really bothers me , I have looked for similar
cases but couldn't really
find an answer .
I keep getting messages which says that
-- B-channel xx successfully restarted on span x
and this causes the calls to be disconnected if somone is already using the
channel
Hi ,
I'm having a problem that really bothers me , I havelooked for similar cases but couldn't really
find an answer .
I keep getting messages whichsays that
-- B-channel xx successfully restarted on span x
and this causes the calls to be disconnected if somone is already using the channel
this is answered in the FAQ: http://www.voip-info.org/wiki-Asterisk+FAQ
RTP based protocols (using RTP to transfer the voice in a separate UDP
session) are generally hard to NAT.
On Thu, 2004-01-15 at 13:29, David A. Lauer wrote:
Can skinny work behind NAT? I have a Cisco 7910 using SCCP behind
fxp0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500
address: 00:02:55:30:54:28
media: Ethernet autoselect (100baseTX full-duplex)
status: active
inet 192.168.1.1 netmask 0xff00 broadcast 192.168.1.255
inet6
hi all
for new users, finding asterisk info is unneccesary troublesome. the
asterisk.org page has very little information about the product and
using google for 'asterisk' is like using google for 'linux'. you get
all too many hits that has nothing to do with the product. perhaps the
asterisk.org
Hi -
I noticed this as well a while ago, and spoke with Mark Spencer at Digium.
I think he said it is normal for the channels to occasionally restart
themselves, however I didn't think they were supposed to do it if they are
in use.
Perhaps you should send a message to [EMAIL PROTECTED] and see
Ali,
If
Zap/82 is channel 20 on Span 3, then it looks like it's hanging up before the
channel restarts as this line indicates.
== Spawn extension (inbound, 9009170,
2) exited non-zero on 'Zap/82-1'
Maybe
there is a problem with your agi script.
B
channels only restart when the PRI
Hi!
I am struggling around with * and the spandsp library (app_rxfax) for a couple
of days. I'm trying to receive faxes which come via a SIP gateway.
The rxfax-module answers the call, the reception of faxes, however, still does
not work correctly, the received file is only about 300 bytes of
Hi all!
Can anybody please give me some advice, what is wrong at my first try to
compile Asterisk. I have successfully downloaded the sources from CVS, but
now the next step at zaptel clean; make install fails.
Please have a look at the error-log below.
There must be a fundamental
I have the following configuration:
Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP)
i can register fine and call ringing is working as good. The problem is
i cant hear audio both ways and i get this error:
WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read
Hi all!
Sorry, the error-log in my previous mail was disturbed.
Can anybody please give me some advice, what is wrong at my first try to
compile Asterisk. I have successfully downloaded the sources from CVS, but
now the next step at zaptel clean; make install fails.
Please have a look at the
On my Linux box mii-tool yeilds the following which shows 100mbs full
duplex.
[EMAIL PROTECTED] gford]# mii-tool
eth0: negotiated 100baseTx-FD, link ok
/glen
[EMAIL PROTECTED] wrote:
My ADSL speed is Uplink 128kbit and Downstream 512kbit.
The mii-tool does not tell whether eth0 is in
Hello,
I'm also new to * but I think that this is what we had to do:
You need to make sure that the following packages are installed on your
system:
-OpenSSL-Devel
-Ncurses
-Ncurses-Devel (C++)
-sox
-kernel sourses
-kernel development
-bison
-newt
-newt-devel
-readline
-readline-devel
If you
Have you confirmed that the call is using the ulaw or alaw codec? It
won't work otherwise.
On Thu, 2004-01-15 at 08:13, Peter Bittner wrote:
Hi!
I am struggling around with * and the spandsp library (app_rxfax) for a couple
of days. I'm trying to receive faxes which come via a SIP gateway.
Mine is a small company with 2 incoming analog lines that we use for voice.
One line rolls over to the other if the first is busy.
I started using an */grandstream combo a while ago, and besides a couple of
bugs that I have yet to work out (echo, ringing in the earpiece) its has
been good for the
Ali Mughrabi wrote:
Hi ,
I'm having a problem that really bothers me , I have looked for similar cases
but couldn't really
find an answer .
I keep getting messages which says that
-- B-channel xx successfully restarted on span x
and this causes the calls to be disconnected if somone is already
On Thursday 15 January 2004 08:37, Franz Edler wrote:
Can anybody please give me some advice, what is wrong at my first try
to compile Asterisk. I have successfully downloaded the sources from
CVS, but now the next step at zaptel clean; make install fails.
Please have a look at the error-log
All,
I was wondering if anyone has any experience with the Cisco ATA186 (SIP image) and
outbound faxing with Asterisk. Inbound faxs from PSTN into * and on to the ATA work
fine but outbound faxs receive congestion from *.
I've got packet dumps from both sides and everything appears normal but
Whaaa?? So, to allow 24+ lines of dial in access, how would I configure it?
Would I need to purchase or lease a voice-over-ip box to connect our T1 or
phone lines into? And then from there send the VOIP to the linux/Asterisk
box for recording?
(forgive me, Im new to telephony, but I need to
Someone submitted a bug about wav49 voicemail problems with
the Windows Media Player here
http://bugs.digium.com/bug_view_page.php?bug_id=254
bkw918 changed the status of the bug to resolved because he
could not reproduce the error with his version of Windows Media
Player. I am having the
Hello everybody,
I have the next scenario, working now.
ISDN Network AGW IP 1(*)IP NetworkGW IP 2(*)ISDN
Network B
Now we are looking the avalibility of sending call's from 1 to 2, using
an VoIP Protocol supported by *. But we have the next questions, before
began to do it.
I'm having the same problem.
Warwick
- Original Message -
From: Jim Flagg [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 5:39 PM
Subject: [Asterisk-Users] wav49 voicemail problem with Windows Media Player
Someone submitted a bug about wav49 voicemail
The restarts only occur on idle channels. However, what is interesting is
that according to this log, the *switch* requests a restart on channel 20
(which if my calculations are correct, is the channel that has the call on
it). You can see this because we log the messages on getting RESTART_ACK
you don't have libm (m for math) library ?
Martin
On Thu, 15 Jan 2004, [gb2312] Âí÷ë wrote:
erro cocent:cc -shared -Wl,-soname,libtonezone.so.1 -lm -o libtonezone.so.1.0
zonedata.lo tonezone.lo
/sbin/ldconfig -n .
ln -sf libtonezone.so.1 libtonezone.so
cc -o ztcfg ztcfg.o -lm -L.
hi all
what about this...
I just put together a box on a web shop (komplett.no) that will cost me
NOK ~1850 (¤ 216) plus a small ¤50 drive and cables, so say ¤300. This
consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI
cards (if capijod will finish off the zaptel-driver soon).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gary Franczyk
Sent: Thursday, January 15, 2004 10:37 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question
Whaaa?? So, to allow 24+ lines of dial in access,
I am having problems too Just shy of the 5-second mark in the
test vm.
WMP 9.00.00.3075
Windows 2000 SP4
-Original Message-
From: Warwick Ward-Cox [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 10:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] wav49
Hi!
I am trying to use * to handle anonymous ISDN callers.
exten = 5150/0,1,Congestion
should work but doesn't. Apparently because the ISDN CAPI doesn't
use 0 for callers who don't send their number.
Is there a way to make * identify ISDN callers who use CLIR?
Here is what I do:
exten
I can't reproduce this either, but I do have the gsm codec installed (though
WMP won't play a .gsm file).
I play the wav49 files in Winamp with no issue.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
Pulaski Chamber 2002 Small Business Of The Year
A caller to me was this afternoon detected as a fax machine:
Jan 15 15:31:17 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax
detected, but no fax extension
... and then redirected to voicemail. An extract from extensions.conf is
attached below. Is there any way to stop * even
Me too :(
100% CPU.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of F.G.Testa
Sent: 14 January 2004 20:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] 100% of cpu in an out of the box *
Hi all!
I'm newbie, so here goes my situation:
I have
Hi!
for new users, finding asterisk info is unneccesary troublesome. the
asterisk.org page has very little information about the product and
using google for 'asterisk' is like using google for 'linux'. you get
all too many hits that has nothing to do with the product. perhaps the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Roy Sigurd Karlsbakk
Sent: Thursday, January 15, 2004 11:08 AM
To: Asterisk Users
Subject: [Asterisk-Users] ultra-cheap asterisk box
hi all
what about this...
I just put together a box on a
On Thu, 2004-01-15 at 10:41, Robert Murray wrote:
Hi Mark
Would it be possible to include a way of streaming audio from memory?
For example registering a file type which read from a fifo in memory?
I need this for app_theta. (Cepstral TTS)
I could copy the code from file.c, but I think it
I'm looking to do about the same thing, build very low cost
systems. (I'm looking at putting Asterisk at some
non-profit organizations.) but one thing you can't make
a compromise on is reliabilty. It has to work and keep working
for years to come. I was able to keep the price of a new PC
to
Hi,
Any * users in sweden, particularly in the Malmo or Lund areas? Mail me
off-list, i have some questions :)
Faiz
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
On Thu, 15 Jan 2004, Steve waxed:
On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote:
sounds like one of those pesky auto dialers the simpsons make fun of.
It sure does...
The AT-5000 was Prof. Frink's first patent, and it was
designed to alert children of snow days and such. I
--- [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I have just checked the Openbsd box on the if interface.
fxp0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500
address: 00:02:55:30:54:28
media: Ethernet autoselect (100baseTX full-duplex)
status: active
On Thursday, January 15, 2004 10:42 AM, Iain Stevenson
[SMTP:[EMAIL PROTECTED] wrote:
...
Is there any way to stop * even considering an
incoming
call on a line as a fax call?
Sure, just don't have
exten = fax.
in the same context (or included context).
Iain
--
Don Pobanz
I think this is a MAROR bug in the new dsp.c routines, recompile using the
old dsp stuff by changing the makefile and set OLD_DSP_ROUTINES
- Original Message -
From: Iain Stevenson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 8:41 AM
Subject: [Asterisk-Users]
On Thu, 15 Jan 2004, mattf waxed:
8's
There is a group of Asterisk users that decided to modify the code of
Asterisk to try to make it a predictive dialer, called shady_dial I believe,
but I haven't heard anything about it lately.
http://shadydial.sourceforge.net/
Lots of recent updates
If you don't have a fax connected to * then create and exten:
exten = fax,1,Goto(day,s,1)
I had the same today... :/
Andy
*** REPLY SEPARATOR ***
On 15/01/2004 at 16:41 Iain Stevenson wrote:
A caller to me was this afternoon detected as a fax machine:
Jan 15 15:31:17
It's not what I would want to depend on day in and day out. I know that
you can buy Dell PowerEdge SC400 servers for $299 with HDD, memory, and
either a celeron or p4, depending on what day of the week it is. I'd put
my name on the Dell based solution before the white box solution for the
same
I have been compiling information on this configuration onto the Wiki:
http://voip-info.org/wiki-Asterisk+cisco+FXO
I can call out to the PSTN just fine, but inbound calls all appear in my
default [bogon-calls] context, not in [pstn-incoming]
Can anyone help me locate why?
(Config files are on
Hello,
I have a Hypercom T7P swipe card terminal sitting on a dedicated G711ulaw
port. The Hypercom operates at either 1200 or 2400bps. I get about a 50%
success rate when I try to authorize cards. On this same G711ulaw port, I
have a fax machine with a 100% success rate operating at 9600bps.
how do you spell Teleecooomm again?
[EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gary Franczyk
Sent: Thursday, January 15, 2004 10:37 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Basic Asterisk capabilities
- Original Message -
From: Iain Stevenson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 11:41 AM
Subject: [Asterisk-Users] People detected as fax machines
A caller to me was this afternoon detected as a fax machine:
Jan 15 15:31:17 NOTICE[41997]: File
I get 'Access Denied'... Can it be downloaded zip or tar ball?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: Thursday, January 15, 2004 11:01 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] asterisk.org
Are you wanting to make a pre-built * box, with hardware to connect a
single dial line and one traditional phone, or.. ?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 11:08 AM
To: [EMAIL
Actually he found it in the dumpster after the police threw it out
following a bust! Does anyone want to send a dollar to Mr. Happy?!
-Original Message-
From: C. Maj [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 12:40 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] *
Hi To ALL
i have made an application for billing a traffic but i have strange problem
with free message from Telco provider because when dial the number and Telco
reply The customer have change number... i dont receive a connect so i cant
listen nothing... Yes is right from PRI dont
On Thu, 2004-01-15 at 14:31, Chris Albertson wrote:
I'm looking to do about the same thing, build very low cost
systems. (I'm looking at putting Asterisk at some
non-profit organizations.) but one thing you can't make
a compromise on is reliabilty. It has to work and keep working
for
well, it does say SIMPLEX in the fxp0 flags section. I don't honestly
know if this means it's negotiated half duplex, or something beyond
that 10baseT is capable of running full duplex, although this
requires a NIC capable of is, as well as a switch that can do FD. And
regarding the 1%
Don Pobanz wrote:
On Wednesday, January 14, 2004 5:48 AM, Daniel Bichara
[SMTP:[EMAIL PROTECTED]] wrote:
Hi,
Once a day, * drops all calls (E100P board). Yesterday, I updated *
version to CVS but I got the problem again today. Monitoring log
files,
I found this messages just
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Albertson
Sent: Thursday, January 15, 2004 12:40 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] re hardware requirement - asterisk
I don't think 10BaseT can run full duplex. I
Hi friends,
Could some one recommand a good cdr processing software out there for post
paid billing (invoicing, web-based payment processing) etc.,
Thanks a bunch.
SW
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hello all. I'm new to asterisk and have been using and testing it for about a week
now. My initial hope has been to use it as a sip-h323 gateway to tie SIP H323
based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks.
I am currently running Asterisk 0.5.0 under Redhat 9
Hi,
I'm relativle new to *, so I may be wrong. I build up * from cvs today
(show version: CVS-01/15/04-16:27:36). In an test I use 2 SIP phones
(linphone) to connect to eachother.
The phones are called via the extensions 100 (user 'kwe') and 200 (user
'phone').
I can call from one to another
- Original Message -
From: Zhang Peihao [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 8:51 PM
Subject: [Asterisk-Users] Cooperate with SIP ITSP
Hi All,
When I want use Asterisk as a PBX to cooperate SIP ITSP,
I can not set the caller ID, so SIP ITSP do not
app_festival currently seems to chop the start of sound it plays back -
probably something to do with rtp and maybe the same problem that was
present in voicemail prompt plauback.
Iain
--On Thursday, January 15, 2004 11:16 am -0600 Steven Critchfield
[EMAIL PROTECTED] wrote:
On Thu,
I think that it will be greate to include * inside of a router like ix66
from intertex... 1 GB usb removable flash to record voice mail.and prompts
in the computer..2 fxo...real internal sip server ...internal dns
server..good user interface.. all nat / firewall nightmare ended, no
computers to
On 15/01/04 13:12, Roy Sigurd Karlsbakk wrote:
hi all
for new users, finding asterisk info is unneccesary troublesome. the
asterisk.org page has very little information about the product and
using google for 'asterisk' is like using google for 'linux'. you get
all too many hits that has nothing
Hi All,
I have a e100p that is not receiving any interrupts. My /proc/interrupts
look like
CPU0
0: 87288 XT-PIC timer
1:104 XT-PIC keyboard
2: 0 XT-PIC cascade
8: 1 XT-PIC rtc
10: 814092
fxp0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500
address: 00:02:55:30:54:28
media: Ethernet autoselect (100baseTX full-duplex)
status: active
inet 192.168.1.1 netmask 0xff00 broadcast 192.168.1.255
inet6
It's spelled MCI, WorldCom, Sprint, T-Mobile... All the same except for
the billing and the twists and turns of the contract. Whatever happened
to POTS (i.e., Bell System.)
An Old ATT 4A/ETS and ESS/S7 Craft
==
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Has anyone played around with QoS or TOS relative to * and sip phones?
I was just doing a little real-time research and noticed our C7960's
mark IP packets with low delay and high throughput (presumably due
to tos_media: 5 in the SIPDefault config file), and rtp packets flowing
from asterisk
This is not directly * related, but could be. My company is
using a VoIP conferencing solution that is suffering from
developer neglect.
I've considered trying to leverage *, and our internal developers
can build the management interfaces. If that plan is not accepted
by management, I need to
Oooops... A little jump ahead. It asked for sign on etc... Got it now,
mucho thanks and understanding my slow brain... :-}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Mynatt
Sent: Thursday, January 15, 2004 12:39 PM
To: [EMAIL PROTECTED]
Are there any cheap SIP phones (like the Grandstream
for example) that support power over ethernet?
What is necessary to support SIP phones in a
Cisco Call Manager environment?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I am real close to finalizing my hardware selection for my Asterisk test
machine. I am going to use the following hardware:
Dell 400SC w\Red Hat 9.0
1 - 4 Port TDM40B Card (FXS)
3 - Wildcard X100P Cards (FXO)
Are there any known conflicts using this setup in this machine? I will be
are you running safe_asterisk ?
If so try to modify safe_asterisk ... CONSOLE=yes to CONSOLE=no
or if not
list all the asteirsk threads 'ps -axum | grep asterisk'
find the thread that takes the most CPU and connect with gdb
gdb /usr/sbin/asterisk pid
and do 'bt'
and post the last few lines
Sipura recommended disabling the echo cancellation on the SPA-2000 for
modem pass-through. It does help although still not 100% success rate.
Stephen
-Original Message-
From: Christopher J. Wolff [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 12:14 PM
To: [EMAIL
I have a user, running CVS a/o 11/23/03, who has complained about
phantom messages showing up days or even weeks after she has deleted them.
So I asked her to let me know when it happened again, and she called a
few minutes ago.
The directory listing below shows a listing of the
That's why I stoped using app_festival and instead use the Festival
text2wav program to generate a .WAV file and use app_playback to stream
the audio to the user.
On Thu, 2004-01-15 at 13:41, Iain Stevenson wrote:
app_festival currently seems to chop the start of sound it plays back -
On 15/01/04 19:39, Jesse Peterson wrote:
#0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72
#1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504
Do you experience the same problems when you use the other (bundled)
h323 driver?
Hi,
I am a newbie in Asterisk as well, intending to use it in a similar way as
you are, communicating with AS5300 as well as other gateways including
MAXTNT.
I have had similar, but yet different experiences than yours.
1. Asterisk does crash with the number of calls, but in my case, about or
Referring to my previous post about degradation of voice quality when
having more than 2 connection.
The actual route is:
pc xlite - local asterisk box - iaxtel - local asterisk
I have tried out a different situation:
pc xlite - local asterisk box - iaxtel
and the second connection
pc xlite
1 - 100 of 151 matches
Mail list logo
