Anders Svensson wrote:
We use Grandstream GPX2000 for this. It works ok. Support 11 lines in basic.
Anders
I also use this phone, have read about the 11 lines, but how does one
'manage' these lines? The first 4 are easy, you have buttons for that,
but how can you use the 'others' ?
Kristian Kielhofner ha scritto:
Or you can keep using the phones with SIP and use sip_notify. I think
Ciscos support it.
In my last try it was not doing it on cisco sip phones.
Sergio
___
--Bandwidth and Colocation provided by Easynews.com --
I'm wondering if there's anyone out there who has successfully gotten an
SPA-3000 to register, as its documentation would indicate, on both ports
5060 (for standard client FXS service) and 5061 (for the purpose of
originating calls via SIP from the PSTN interface on the box).
I can get one or
René Enskat [Teamware GmbH] ha scritto:
-- Executing Set(SCCP/1000131-0006, Language()=de)
edit your sccp.conf and in the general section set
language=de; Default language setting
Sergio Chersovani
___
--Bandwidth and
You can use the speeddial buttons. They are configurable
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristof Hardy
Sent: den 13 december 2005 09:17
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IP
Brian Capouch wrote:
I'm wondering if there's anyone out there who has successfully gotten an
SPA-3000 to register, as its documentation would indicate, on both ports
5060 (for standard client FXS service) and 5061 (for the purpose of
originating calls via SIP from the PSTN interface on the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all,
currently i running * 1.0.9 with chan_capi 0.3.5
my first problem is:
in incoming call, when BCHAN is full in contr1 incoming call on contr2
are not answered with error :
chan_capi.c:1953 in capi_handle_msg: received a call waiting
Dear All,
Never Mind, I have solved the problem. It seems that you should clear the buffer for any 'waiting' response or else you will be getting an empty '200 result=1' response. So be sure to read, before you write in php agi script to ensure that you will get a proper response.
Regards,
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Dec 12 18:03:33 WARNING[7237]: loader.c:325 __load_resource:
/usr/lib/asterisk/modules/cdr_addon_mysql.so: undefined symbol: ast_load
Dec 12 18:03:33 WARNING[7237]: loader.c:554 load_modules: Loading module
cdr_addon_mysql.so failed!
I
Hi :)
I have an A104 and wondered if other owners could confirm the strange
behaviour I'm seeing.. it's best seen on an idle system, thus
eliminating asterisk or other factors..
Very simply, just let 'vmstat 1' run for a few minutes and watch the
output, specifically the 'sy' column...
On the
Hi!
currently i running * 1.0.9 with chan_capi 0.3.5
Try chan_capi-cm instead and see if it helps.
Cheers, Philipp
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Hi!
Are there any Windows-based softphones (SIP or IAX based) that support
the new Hint system in Asterisk 1.2? I don't mind evaluating commercial
options, if they're available.
Try the SNOM softphone:
http://www.snom.com/snom360softphone.html
The only other softphone I am aware if is
Hi all,
I have really very serious problem. I installed G.729 and G.7231 from
the Intel. And I got it is registered with asterisk.
Registered translator 'g723tolin' from format g723 to slin, cost 1
Registered translator 'lintog723' from format slin to g723, cost 7
Registered translator
Hi,
Yesterday was the first day my call center operated
under Asterisk 1.2.1.
At the end of the day, I ran a "show queue
queuename" and saw that the value of abandoned calls was
45.
This morning, after updating my database with data
from queue_log file, I saw, through Asterisk Guru Queue
Hi,
I use Asterisk 1.2. My configuration is:
ooh323.conf:
[general]
port=1720
bindaddr=0.0.0.0
allow=all
context=office
tos=lowdelay
iax.conf:
[general]
disallow=all
allow=gsm
bindport=4569
bindaddr=0.0.0.0
codecpriority=reqonly
language=en
jitterbuffer=yes
tos=lowdelay
[test]
type=friend
as I though this
was disabled with canreinvite=no.
denwa*CLI
-- Executing Goto(SIP/10.129.46.102-0853ec38, sip|1000|1) in new stack
-- Goto (sip,1000,1)
-- Executing SetVar(SIP/10.129.46.102-0853ec38,
CALLFILENAME=000-20051213-110514) in new sta
ck
-- Executing GotoIfTime(SIP/10.129.46.102-0853ec38
Mojo Jojo wrote:
Anyone using one of these as a QOS device in an Asterisk environment?
If so, does it work well?
No, I don't use one of these myself. However...
Do you know what exactly it prioritizes? SIP only? IAX?
...during my recent DCE course, this product (or one extremely similar
to
tir, 08,.11.2005 kl. 17.08 +0100, skrev Olivier Perrin:
According your conf, you are in France, so i answer in french :-)
That's really not very polite, since most people on this list won't
understand a word you're saying. Other people read this list too, you
know...
--
Henning Kilset Pedersen
Hi All
I am having various problems that I am convinced is NAT related.
I have a Vega box on public IP talking to an Asterisk box on a public IP
address. Calls from the Asterisk to the Vega and back are fine. I have 2 VoIP
phones in a NAT network registered to the Asterisk box.
The problems I
Hello,
Can you post what firmware your board is and what wanpipe driver
version you are using?
We do up to 50 concurrent recordings on our systems and they do not
have recording issues. We use MegaRAID 320-1 cards as well.
MATT---
On 12/13/05, Gavin Hamill [EMAIL PROTECTED] wrote:
Hi :)
I
Hi Asterisk Users,
i have a bristuffed-0.2.0-RC8q Asterisk 1.0.9 System running on a Debian
3.1. With a quadbri card installad, wich is running on the bristuff drivers.
Everything seems to be fine so far.
but now i wanted to use the g.729A Codec. I bought 5 licences and
installed them:
Hi all!
I have got a bit strange output from iax2 show channels:
Med venlig hilsen
ComX Networks A/S
Dmitry Zhukovski
System developer
ComX Networks A/S
Naverland 31, 2
DK-2600 Glostrup
Denmark
Phone: +45 70 25 74 74
Fax: +45 70 25 73 74
Web: www.comx.dk
Dmitry Zhukovski
Direct:
One of my things also does very strange things, does somebody know what
could be wrong with those things ?
Maybe the other guy (you know, the one with the hair and the two or less
eyes and two legs) could help me ?
Please, at least give us some info... What are you referring to ?
Zoa
Dmitry
Hi all!
Sorry for last message.
I have got a bit strange output from iax2 show channels:
x*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx)
Lag Jitter JitBuf Format
(None)xx.xx.xx.xx x 1/00318
On Tue, December 13, 2005 13:47, Dmitry Zhukovski said:
Hi all!
I have got a bit strange output from iax2 show channels:
Med venlig hilsen
ComX Networks A/S
Dmitry Zhukovski
System developer
Adding some info might be helpful?
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0
On Tue, 2005-12-13 at 07:24 -0500, Matt Florell wrote:
Hello,
Can you post what firmware your board is and what wanpipe driver
version you are using?
Hi Matt :)
I've already been through all this with Sangoma's support - just looking
for external opinions from real-life installs - so thank
Got a really wierd problem her. Maby it's a bug.
But before i report it, i'll try my luck here.
I have one asterisk server on public ip.
I have two identical hardphones on two different LAN's. The firewall are
different.
Both are configured in asterisk with nat=yes and qualify=yes.
For one
Brian Capouch wrote:
I'm wondering if there's anyone out there who has successfully gotten an
SPA-3000 to register, as its documentation would indicate, on both ports
5060 (for standard client FXS service) and 5061 (for the purpose of
originating calls via SIP from the PSTN interface on the
In article 3bf71fa80512121816u6928839cg2dfcf14d3ffb2c04
@mail.gmail.com, [EMAIL PROTECTED] says...
I believe you are missing 2 variables in your conf file:
table=cdr
(the table your cdrs should be stored)
sock=/var/lib/mysql/mysql.sock
(the location to your mysql.sock)
I didn't use
Instead of hostname=localhost, it would be hostname=IP address of MySQL server.On 12/12/05,
Innocent Evil [EMAIL PROTECTED] wrote:
I was also following this thread.
Would anybody please tell, what would be configuration file if mysql is a different machine than asterisk box?
Thanks,
--You
On Fri, Dec 09, 2005 at 03:23:31PM -0500, Tom Rymes wrote:
On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi:
i added these two lines to my general context ,but
nothing happened the same result the sound came in one
way for 3 seconds and stopped but it didnt hangup.
--- Jeffery
On Tue, 2005-12-13 at 12:59 +, Gavin Hamill wrote:
[snip]
What kind of CPUs are you using? Also, single or dual (or a single with
hyperthreading ?) What onboard L2 cache do they have? My last hope is to
try a P4 machine with 1MB cache, since the others I've used have 512K..
They're all
Karl O. Pinc wrote:
I foolishly made this patch against the zaptel 1.2
branch rather than trunk, although I did check that
the trunk has the problem. It'll probably apply
This script is completely unnecessary on Debian; just add the modules
you wish to load into /etc/modules and they
Douglas Garstang wrote:
Then again updates are sent to the master DNS server, which filters them
down to the slave DNS server, and you do queries to the slave... might take a
few minutes to become effective.
The bigger issue would be caching on the client ends; unless you set the
TTL
It seems that version 1.2.1 has broken Asterisks
ability to use realtime in the voicemail.conf file.
It appears that the statement:
switch = Realtime/@
is not read properly by Asterisk.
-- Executing
Voicemail(mailto:Local/[EMAIL PROTECTED],2,
mailto:[EMAIL PROTECTED])
Dec 13
On Tuesday 13 December 2005 01:27, Jason Frisch wrote:
I see. How would I go about checking such conflicts (for the future)
With the old NIC in and everything running normal, type cat /proc/interrupts
/tmp/ints-oldnic.txt
Now with the new NIC in and everything running normal, type
cat
On Tuesday 13 December 2005 06:20, Mario Evangelista-Silva wrote:
Verify communication between protocols. SIP ou IAX2.
I get it with both protocols, but it's far more infrequent... one call in a
hundred maybe. I've verified (with IAX2 at least) that both sides are seeing
each other's
On Tuesday 13 December 2005 02:11, Chris Mason (Lists) wrote:
At sixty concurrent calls, you are not looking at a small embedded
machine. Rack mount dual P3 or P4 in a small form factor I could see. I
have to wonder if a CF card based system can be adequate for this kind
of work, I have tended
Joseph Rothstein wrote:
It seems that version 1.2.1 has broken Asterisk's ability to use realtime in
the voicemail.conf file.
It appears that the statement:
switch = Realtime/@
is not read properly by Asterisk.
(Could you use a little more whitespace next time? G)
What does
Thank you Traci,I put this two variables in my .conf file and it works!!!Well, It seems that this variables are not necessaries in old versions, but in newest ones.Thank you again,Juanjo
I believe you are missing 2 variables in your conf file:table=cdr(the table your cdrs should be
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William K.
Volkman
Sent: Monday, December 12, 2005 9:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Turning off hardware echo can on TE411P
Hello,
On Mon,
I didn't write this below. I replied with a blank line by mistake.
I am truly sorry if you were confused by that.
-Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, December 13, 2005 8:16 AM
To: Asterisk Users Mailing
Hi
We have a call queue setup with several agents using agentcallbacklogin.
If one of the agent is logged in and is talking on the phone with another
employee the queue application doesn't see that the phone is busy and
continues to forward incoming calls to him.
Since the agent cannot
Hi, i just figured out, that there is also a problem by going in a
conference with the sip phone that runs the g729a codec.
Could it be, that i have timing problems? I don´t have digium hardware
installed, but i have ztdummy:
asterisk3:/etc/asterisk# lsmod | grep ztdummy
ztdummy
Jason Brashear wrote:
OK, so is there a way to have hardware echo canceling and have DTMF
digits go out correctly? We bought the expensive hardware echo
canceling card however it appears that we have to have vpmsupport=0
in order to get DNIS digits correctly (see my thread about ADIT and
DNIS
I have a Snom 190 and setup two lines one for the local Asterisk and the
Other for a remote asterisk. I can see that both likes register and in the
web interface say they are ok. My problem is that line 1 takes precedence.
I am not sure how to use line 2.
If I go to the main setup page in the web
Yeah I just got in a 301 to test and I can configure a key (for example
in sip.cfg key.IP_300.2.function.prim=Messages/ and then when I hit
the line 2 key it drops me right into VM (since I have that configured
too)
Still playing around, I noticed that if you get the soft keys (the menu
ones
FC4, Asterisk 1.0.9 and SjPhone softphone. On CLI I get this message
every 20 sec.
# Testing 10.0.0.203 with 10.0.0.0
10.0.0.203 is the IP of softphone and 10.0.0.0 is the network defind in
sip.conf. Asterisk server is on 10.0.0.26 address.
Why do I get this message?
sip.conf
[general]
On Tue, 2005-12-13 at 14:32 +0100, Patrick wrote:
Been a while since I used Asterisk on a Dell box but I remember I had to
turn off HT. Have you tried that?
For sure, I've tried HT on and off, with 2.4 and 2.6 kernels :)
or booting the kernel with noht. On Dell boxes I have also seen some
Or you can treat everything as a 10 digit number retaining in a variable
whether the user dialed one or not
exten = _1NXXNXX,1,SetVar(ONPRESSED=TRUE) *** skip this step if you
don't care whether the one was pressed in any of your dialplans
exten =
'searchcontexts=yes'
added to my voicemail.conf file solved the problem.
Joe
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
I want to put a * server in front of our legacy phone system. Currently
this legacy system is connected to the CO with an ISDN PRI interface.
With a dual PRI card in the * server can I only pass thru a certain
number of channels to the legacy phone system and then leave the other
half of the
Matt Burleigh wrote:
The legacy phone system would only be able to use, for example, channels
16-23 from the ISDN PRI coming out of the * server.
You cannot make it invisible, because the D-channel cannot be shared.
However, PRI channels are allocated dynamically, so doing what you want
to
This same thing happened to me last night, I'll have to try this out and
see if it works for us too :)
Aaron
Joseph Rothstein wrote:
'searchcontexts=yes'
added to my voicemail.conf file solved the problem.
Joe
___
--Bandwidth and Colocation
Please correct me if I am wrong, but if a SIP call goes unanswered,
shouldn't the proper response be a '408 Request Timeout', and not a 403
Forbidden?
Anyone care to comment?
Joe
___
--Bandwidth and Colocation provided by Easynews.com --
What does this error mean when running fxotune on my TDM04B
could not fill input buffer on channel 2
Thanks
--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Phone: 978-203-3848 x205
Fax: 518-631-2856
We just posted an updated guide to the SPA-3000 a few days ago. The example
uses AMP but all the settings are there:
http://voipspeak.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Tuesday, December 13, 2005 5:04 AM
To: Asterisk
Just curious what everyone (as in, the people that have read it or use it)
thinks about the O'Reilly Asterisk book. I'd really like to delve into the
nitty gritty of Asterisk, but I'm getting kinda tired of swimming through
forums and Google results. I've been reading the wiki off and on for
Yes, it is correct. The best way to handle this problem (on 1.2) is to
pause the agent before the outbound call and the unpause him when he's
done.
Yours
l.
On Tue, 13 Dec 2005 15:20:56 +0100, Patrick Fortin [EMAIL PROTECTED]
wrote:
Hi
We have a call queue setup with several agents
I want to put a * server in front of our legacy phone system.
Currently
this legacy system is connected to the CO with an ISDN PRI interface.
With a dual PRI card in the * server can I only pass thru a certain
number of channels to the legacy phone system and then leave the other
half of
Every few days our receptionist's phone will not take calls on one of
the extensions. We have an extension 118 going to the first two lines of
her phone and extension 101 going to the other. If we try to dial 118 it
goes to voicemail even though she is not on the phone. Asterisk is
thinking
On Tue, 13 Dec 2005 09:45:09 -0600
Ross C [EMAIL PROTECTED] wrote:
Just curious what everyone (as in, the people that have read it or
use it) thinks about the O'Reilly Asterisk book. I'd really like to
delve into the nitty gritty of Asterisk, but I'm getting kinda tired
of swimming through
Thanks for the responses. I guess the next step is to get a Digium
TE210P. Are there any other 2 port PRI cards anyone would recommend for
*?
--
Matt Burleigh
Senior Systems Engineer
Enterprise Integration, Inc.
eiisolutions.com
703-236-0790
-Original Message-
From: [EMAIL PROTECTED]
We use all Asus motherboards now, with single P4 processors(some with
512k, 1024k and 2048k L2 caches) We run most of them with HT on, no
issues there.
Also, if you are using the on-board RAID, it's not really a complete
LSILogic RAID, They(LSILogic) won't support it because Dell does
Philipp von Klitzing wrote:
Hi!
currently i running * 1.0.9 with chan_capi 0.3.5
Try chan_capi-cm instead and see if it helps.
Cheers, Philipp
compiling 0.5.4 when there was more than 2 call i got :
ERROR[6060]: chan_capi.c:2324 capi_handle_connect_indication: received
a call
Hello everyone,
I am trying mISDN driver with asterisk 1.2.1 but when i call from SIP to
mISDN and from mISDN to SIP, the caller ID appears always with a leading
0 (0X). I think the problem is with nationalprefix.
How can I remove that zero
Here is my config.
[general]
debug=0
It looks like http://bugs.digium.com/view.php?id=5266 is the problem here.
My CDR shows as not answered for the tool free number.
The local number answers and call forwards.
Questions:
It says it was committed on 10-04-05. How do I know which versions that was?
I am currently running:
asterisk
Matt Florell wrote:
Hello,
Need some more information here:
- hardware specs (including what kind of hard drives?)
The Asterisk server is a Dell PowerEdge 6850 with the following specs.
Please note that we are NOT recording to the hard drive. We are
recording to a RAM disk as detailed here
I was wrong.
This patch is for channels/chan_zap.c
I have been hesitant to go to 1.2.1 without config testing.
Should I have any negative issues going from 1.0.9 to 1.0.10? ( I have to see
if the changes are in the 1.0.10 version of
channels/chan_zap.c)
--
--
Steven
It looks like
Are you trying to register both lines to the same user account in *?
That wont work, a user can only be registered once at any time.
Kerry Garrison wrote:
We just posted an updated guide to the SPA-3000 a few days ago. The example
uses AMP but all the settings are there:
The book is a great *starting* point, IMHO. If you've spent a
considerable amount of time reading other sources, you probably won't
find much new information in the book. OTOH, you may find that its
organized approach helps consolidate what you've read. And if it clears
up a couple of key
Hey all - I'm sure this has been done before, but I'm curious about how well
it works.. Typically we have all our servers setup for dual fast/gig
ethernet failover... I.e. bond0 slaves eth0 and eth1 and fails over between
the two. This together with dual p/s and raid1'd(at least) drives provides
On 12/13/2005 07:32:10 AM, Kevin P. Fleming wrote:
This script is completely unnecessary on Debian; just add the modules
you wish to load into /etc/modules and they will be loaded at boot
time.
FYI the list. Using debian with linux 2.6 you don't do anything,
the requsite module
wath is the list of isdn cards supported by asterisk?
anybody have the list or the link about that?
__
Visita http://www.tutopia.com y comienza a navegar más rápido en Internet.
Tutopia es Internet para todos.
___
On 12/13/2005 09:45:09 AM, Ross C wrote:
Just curious what everyone (as in, the people that have read it or use
it)
thinks about the O'Reilly Asterisk book.
I am just getting started. The book works for me.
My gripe is the license. I can't submit improvements
where I ran into gotchas, so I
We made a review of it a while ago, if you wonder if you will like it,
why not download the pdf and have a look for yourself ?
http://www.asteriskguru.com/review.php
Zoa.
John Biundo wrote:
The book is a great *starting* point, IMHO. If you've spent a
considerable amount of time reading
stéphane plichon wrote:
Hi all,
currently i running * 1.0.9 with chan_capi 0.3.5
my first problem is:
in incoming call, when BCHAN is full in contr1 incoming call on contr2
are not answered with error :
chan_capi.c:1953 in capi_handle_msg: received a call waiting CONNECT_IN
but if
Matt Burleigh schrieb:
Thanks for the responses. I guess the next step is to get a Digium
TE210P. Are there any other 2 port PRI cards anyone would recommend for
*?
Yes - there are 2xPRI cards based on the CologneChip HFC-E1 chip and the
A102u cards from the Canada based manufacturer SANGOMA:
What codec are the calls? What codec are you recording in?
I would try some non-Dell hardware, I would also try a less bloated
Linux Distro, something like Slackware, just to see if that had any
effect. And make sure you use the megaraid2 linux drivers.
MATT---
On 12/13/05, Matt Roth [EMAIL
Does anybody have a Tellabs manual for:
* 253c shelf. the complete model number is: 81.0253c
* 2572 Echo Canceller card, complete model number is: 81.2572
I know the wiki has got lots of info on it, but I'm trying to get the
original docs from Tellabs.
Thank You
Ross C wrote:
Just curious what everyone (as in, the people that have read it or use it)
thinks about the O'Reilly Asterisk book. I'd really like to delve into the
nitty gritty of Asterisk, but I'm getting kinda tired of swimming through
forums and Google results. I've been reading the wiki
Hi Everyone,
I'm trying to get chan_misdn working with asterisk. Currently I'm using
two seperate * boxes with chan_capi and one AVM Fritz card per box and
I'd love to get one box doing the job (plus I'm hoping that echo
cancellation is better in chan_misdn).
I have this error when I start
Hi all,
I'm currently involved in a project where the meetme application is used
extensively forbridging calls between an operator and 2 or more parties. One
of the features that we require is the ability to pass DTMF signals from any
party in the bridge to a pre-specified bridge connected
We are running a number of hosted SIP-only PBXs, and
we do have memory problems with some of them.
The servers have typically 512 MB RAM, and in some of the servers
the Asterisk usage goes up from a couple of percent (at restart in the morning)
to more than 82% after periods with a high
On Tue, 13 Dec 2005, stéphane plichon wrote:
stéphane plichon wrote:
Hi all,
currently i running * 1.0.9 with chan_capi 0.3.5
my first problem is:
in incoming call, when BCHAN is full in contr1 incoming call on contr2
are not answered with error :
chan_capi.c:1953 in
On a recent install
of ast 1.2 (b1) I noticed something strange in the CDR records (in mysql).
The caller ID name and number contained extra quotes for calls outbound
(inbound was fine).
Below is an example
of the extensions.conf excerpt, and an excerpt from my sql. Can anyone
explain how
Hello
list!
I had a problem
while trying to build asterisk-addons, but noticed some paths specified in the
Makefile didn't fit my system.
So I modified
Makefile for it to look for MySQL includes and libs on the following
locations:
/usr/local/mysql/include/mysql
/usr/local/mysql/lib/mysql
I'm looking for iax2 wifi phones, do you know where i can buy them?
Thanks
Mario
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Matt,
The calls are u-Law. The format of the recordings is PCM. Is this
correct to prevent transcoding the recording? We've noloaded all other
codecs, so I don't believe that transcoding is occurring. I've only
ever seen "show translation" generate the following output:
immlx15*CLI show
I have a 253A manual out of the big three manuals I have, but not the
echo canceller.
- James
C F wrote:
Does anybody have a Tellabs manual for:
* 253c shelf. the complete model number is: 81.0253c
* 2572 Echo Canceller card, complete model number is: 81.2572
I know the wiki has got lots of
Hello list,
I have the following problem.
The behavior of music on hold is not constant on my
queues... Sometimes it plays well, sometimes it becomes mute in the middle of
the wait and sometimes it doesn't even start.
mpg123 is installed on my server.
Is there something I am missing???
Can you please email it to me? off list.
What are the other 2 manuals?
Thank You
On 12/13/05, James Armstrong [EMAIL PROTECTED] wrote:
I have a 253A manual out of the big three manuals I have, but not the
echo canceller.
- James
C F wrote:
Does anybody have a Tellabs manual for:
*
Steve Totaro wrote:
What are you doing in between making changes and testing the changes?
After changing settings I reboot system! Really. :)
Because other actions have no effect. Also reboot, too..
Thanks,
Steve
Just a couple guesses on things to try.
Zapata.conf
1. Changing switchtype
Hi all,
In order to fix my problem with music on hold I
would like to test format_mp3, that comes with asterisk-addons
package.
For that, the wiki says "Be sure to remove mpg123 from your system (this
may attribute to 'Request to schedule in the past!?!?!' messages). Now you are
set! "
Hello,
To see if it's somehow the recording throughput that's the problem I'd
suggest trying recording in GSM just as a test and see how that is.
As for the hardware, just try a machine with no Dell parts in it. I've
talked to many Asterisk users who's problems went away when they
switched to
On Tue, Dec 13, 2005 at 05:12:44PM -0200, Dov Bigio wrote:
For that, the wiki says Be sure to remove mpg123 from your system (this may
attribute to 'Request to schedule in the past!?!?!' messages). Now you are
set!
How do I uninstall mpg123?
How did you install mpg123? If you installed
Kerry Garrison wrote:
We just posted an updated guide to the SPA-3000 a few days ago. The example
uses AMP but all the settings are there:
It was exactly that example that I was using to start with.
Using the setup just as below, I get the following error:
chan_sip.c:10823
If I'm in a meetme conference, what would I need to do to have some
call files make calls and connect them into the meetme conference with
me?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
I'd recommend the Digium dual port cards - generation 2 card are excellent and the support we receive superb. And it supports Digiums support and development of Asterisk - Sangomas contribution is token if any.Rob
On 12/13/05, Christian Victor [EMAIL PROTECTED] wrote:
Matt Burleigh schrieb: Thanks
On Tue, Dec 13, 2005 at 05:12:44PM -0200, Dov Bigio wrote:
For that, the wiki says Be sure to remove mpg123 from your system (this
may attribute to 'Request to schedule in the past!?!?!' messages). Now you
are set!
How do I uninstall mpg123?
How did you install mpg123? If you installed
1 - 100 of 154 matches
Mail list logo
