hi guys,
i have a problem with 1.8 branch no matter which release of 1.8 i'm
using. i can't make any sip calls, this is the error message i get on
each call:
[Jan 18 19:02:15] ERROR[1698] rtp_engine.c: No RTP engine was found.
Do you have one loaded?
[Jan 18 19:02:15] ERROR[1698] chan_sip.c: Got
if callers hangs up.
is there anything i should to to avoid this or it's a bug.
thanks,
paradise dove
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I'm using AGI and set AGISIGHUP=no
to make it keep on running on channel hangup
On Fri, Aug 8, 2008 at 10:24 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
You are using AGI or DeadAGI ?
Paradise Dove wrote:
hi,
i'm using asterisk 1.4.21.2, and i use channel variables in my agi scripts
Thanks, It works now!
but i get this warning as well: Running DeadAGI on a live channel
will cause problems, please use AGI
is it serious? what problems will occur!??
On Fri, Aug 8, 2008 at 11:30 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
Try DeadAGI and it should work..
Paradise Dove
i have two extensions which have call forwarding enabled when they are
busy to forward the caller to each other.
11 ==on busy== 12
12 ==on busy== 11
when both extensions are Busy a large number of stale calls will be
made in the system!
how can i prevent this mess in my system?
hi
is there any way to find out that an fxo module is connected to telco
line or not?
paradise dove
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On 6/15/07, Steve Underwood [EMAIL PROTECTED] wrote:
Paradise Dove wrote:
can anybody help me to choose the most reliable fax solution for * .
after googling the net i found that there are at least two solutions
for this, app_rxfax+spandsp and iaxmodem+hylafax.
- what's
can anybody help me to choose the most reliable fax solution for * .
after googling the net i found that there are at least two solutions
for this, app_rxfax+spandsp and iaxmodem+hylafax.
- what's the differences between these two?
- which one's better? why?
thanks
does astribank from xorcom do the same for me?
asterisk-astribank-Modem/Fax
On 6/13/07, Doug Lytle [EMAIL PROTECTED] wrote:
Jeremy Mann wrote:
So you're doing PRI-Channel bank?
Yes, for inbound:
PRI-Asterisk-Chanel Bank-Modem/Fax/Cheapy Phone
For outbound:
Modem/Fax/Cheapy Phone-Chanel
so how to avoid CPC??
On 6/14/07, C F [EMAIL PROTECTED] wrote:
Its called CPC
On 6/12/07, Kyle Vorster [EMAIL PROTECTED] wrote:
Hello,
Sorry if this is a real dumb question but when sending a fax and the end
user does not enable fax on their side and then just hangs up does not
force
:
- the card with / without vpm module has the same dtmf detection problem.
- relaxdtmf=yes/no didn't solve the problem
- toneduration=300 / 350 / 400 didn't help also.
- vpmdtmfsupport=1 / 0 didn't solve again.
what else could be the possible cause for this problem?
please help!
- paradise dove
On 5/8/07, Kevin Collins [EMAIL PROTECTED] wrote:
I modified chan_sip.c to turn on a dsp to do fax detect based on inband dtmf
being selected. And when reading rtp if 'f' character shows up vector to
fax extension
can i have your patched chan_sip.c ?
Kevin Collins
-Original
hi,
is there anyway to Answer() the caller channel after the called number
pickedup the phone.
when an outside caller calls * system just continue ringing and not pick up
the line and just dial an extension and then answer the caller channel after
the called extension picked up the phone.
is this
doesn't bridge the call and just ring
for the caller and noise for called!!
is it a bug or it's normal?
The noise may indicate other problems.
Yuan Liu
Paradise Dove wrote:
hi,
is there anyway to Answer() the caller channel after the called number
pickedup the phone.
when an outside caller
On 2/22/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Paradise Dove wrote:
On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 09:39:22 +0100
I think, this can be solved using phone autoanswer feature, look at
wiki...
exten
On 2/23/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Feb 22, 2007 at 09:40:54PM +0330, Paradise Dove wrote:
On 2/22/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Paradise Dove wrote:
On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Date
Hi,
I'm running FC3 with kernel 2.6.11.
All the binary files and zaptel kernel modules is not available to system at
boot time.
They are extracted in a ram disk at system startup and then zaptel modules
are loaded manually and so on.
I have no problem with this boot routine and i've been tested
On 2/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Feb 17, 2007 at 07:52:25PM +0330, Paradise Dove wrote:
Hi,
I'm running FC3 with kernel 2.6.11.
All the binary files and zaptel kernel modules is not available to
system at
boot time.
They are extracted in a ram disk at system
wctdm2400p: reg is a04c0004
Resetting the modules...
On 2/18/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Feb 17, 2007 at 11:52:32PM +0330, Paradise Dove wrote:
On 2/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Feb 17, 2007 at 07:52:25PM +0330, Paradise Dove wrote:
Hi,
I'm
does TDM2400 work on 3.3v pci slot?
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:
On 2/11/07, Paradise Dove [EMAIL PROTECTED] wrote:
does TDM2400 work on 3.3v pci slot?
Yes, all of Digium's analog cards are dual voltage and can work with
either 3.3V or 5V slots. You just need to make sure you have an extra
molex connector if you're going to be using FXS modules on the card
i have a full fxo TDM24 and i have problem with installing it.
when i run modprobe wctdm24xxp dmesg shows the following messages.
and it waits for ever and nothing will happen.
i'm sure that:
- the power is plugged into tdm24 board
- udev is configured and is working with other tdm cards.
-
hi,
rx_fax fails to get fax on a bit noisy lines
but real fax devices can do that on the same line
with no problem!
what's the problem?
thanks
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hi,
i've just upgraded to latest trunk. everything compiles fine but when
starting this message appears and fails to start.
WARNING[3990] loader.c: module chan_zap.so error
/usr/lib/asterisk/modules/chan_zap.so: undefined symbol:
ast_pickup_call
thanks,
paradise dove
pre25 is working fine for me.
On 2/19/06, Jesse Guardiani [EMAIL PROTECTED] wrote:
Hello,
Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or
1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax,
and it builds, but I'm not having any luck
this is a time issue.
change your date to older value. everything works again.
paradise dove
On 1/25/06, stevanus [EMAIL PROTECTED] wrote:
Hi guys,
I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at
the third days I activated setting jitterbuffer=yes and suddenly
.
Anyone shed any light on this? I'm hacking our CDRs currently to work
around the difference in year, but I've obviously also had to disable
ntp and I hate to think what setting the date by hand will have done to
our CDR collation between machines...
Paradise Dove wrote:
this is a new bug which
this is a new bug which is submitted: http://bugs.digium.com/view.php?id=6349
change your system date to an older value. everything will work again.
paradise dove
On 1/25/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
Hi,
I set up a small system over the last couple of days and all
] app_rxfax.c: FLOW Fast carrier down
Jan 18 11:54:52 DEBUG[5157] app_rxfax.c: FLOW Fast carrier up
Jan 18 11:54:52 DEBUG[5157] app_rxfax.c: FLOW Fast carrier down
Jan 18 11:54:53 DEBUG[5157] app_rxfax.c: Got hangup
thanks,
paradise dove
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i have the same problem and also have submitted it as bug
http://bugs.digium.com/view.php?id=5281.
the Patch-5281-v2.txt in the mentioned bug will solve your problem.
Paradise Dove
On 1/3/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Joseph Rothstein wrote:
I am setting up 10 SNOM 320s
i have the same problem. but when i remove all hints from my dialplan
in extensions.conf.
on more crash will occur.
Paradise Dove
On 1/2/06, Andrew Gough [EMAIL PROTECTED] wrote:
I don't think this is the same problem I am experiencing. As you can see
below the two BT's are almost identical
hi,
is it possible to use GROUP_COUNT function in AGIs.
i could not make it work. :-(
thanks
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the Asterisk crash.
I have 1.2.0
On 12/3/05, Paradise Dove [EMAIL PROTECTED] wrote:
hi,
This is the new update_call_counter() which works fine for me:
/*! \brief update_call_counter: Handle call_limit for SIP users
* Note: This is going to be replaced by app_groupcount
* Thought
*/
// paradise dove
p = find_peer(name, NULL, 1);
if (p) {
inuse = p-inUse;
call_limit = p-call_limit;
} else if (!u) {
/* Try to find user */
u = find_user(name, 1);
if (u) {
inuse = u-inUse;
call_limit = u-call_limit;
} else
user to peer you will see that hints
works fine.
another way to find this bug is to run the command sip show inuse on
CLI when some sip extensions are in a call. you will see that just the
user counter of sip friends are updated.
Paradise Dove
On 11/29/05, Alvaro Parres [EMAIL PROTECTED] wrote
btw, i've patched this part of code and now its working fine for me.
i'm going to upload it.
Paradise Dove
On 11/30/05, Kevin Hanson [EMAIL PROTECTED] wrote:
Paradise Dove wrote:
Yes with version 1.2. I have tried already with call-limit and the same.
i agree with you, it seems
hi,
how can i hangup such calls without restarting asterisk?
the Zap channel on this case is busy for more than 7 hours
some logs are followed.
thanks,
Paradise Dove
-
Nov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25788
as i said before, i've ran soft hangup on both sip and zap channels
on this call several times but no success.
by exploring the code in chan_sip.c it shows that * also attempts to
run softhangup on this call.
is this probably be a bug?
thanks,
paradise dove
On 11/25/05, tracinet [EMAIL PROTECTED
hi,
is there anyway to just enable faxdetection in voicemail?
thanks,
paradise dove
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* can't detect the callerid.
thanks,
paradise dove
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me too!
i had hints working for months before upgrading to CVS HEAD.
i've also submitted a bugs: http://bugs.digium.com/view.php?id=5281
my question is that is there anybody who is using CVS HEAD and hints
works for him?
btw, thanks,
Paradise Dove
On 10/7/05, Stefan Tichy [EMAIL PROTECTED
Hi,
i was just wondering that is there anybody who has
any success with hints on CVS HEAD?
a sample configuration of sip.conf and extensions.conf
is pleased.
Paradise Dove
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i get this error on dmesg:
zaptel: Unknown symbol __stack_smash_handler
zaptel: Unknown symbol __guard
paradise dove
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has anybody succeeded in connecting an E1 CB to asterisk using R2
Digital signalling and Unicall?
any help will be appreciated,
Paradise Dove
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Is there any success in connecting Valiant E1 CB with Unicall to asterisk?
any help will be appreciated,
Paradise Dove
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i've tried it on both snom190 and eyeBeam none of them work.
nothing is changed in configs.
is there any success in making snom LEDs work on CVS HEAD?
thanks,
paradise dove
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does * support call return?
i want when the operator transfers a call if the transferee is busy or
doesn't answer the call the call return back to operator again...
this feature may be called:
call return on busy
call return on no answer
Paradise Dove
] channel.c: Avoided initial
deadlock for '0xb7c861b8', 10 retries!
warning:Jul 22 14:54:37 WARNING[26237] channel.c: Avoided initial
deadlock for '0xb7c861b8', 10 retries!
... tones of these messages...
I'm using latest CVS HEAD.
thanks,
Paradise Dove
=4,0,0,esf,b8zs
fxsks=73-96
loadzone=us
defaultzone=us
zapata.conf:
[channels]
context=incoming
callerid=asreceived
busydetect=yes
busycount=7
faxdetect=no
signalling=fxs_ks
overlapdial=no
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
channel = 1-96
thanks.
Paradise Dove
: Disconnecting call
'SIP/2399-27f7' for lack of RTP activity in 8108 seconds
thanks,
Paradise Dove
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hi all,
is there any way to force all sip peers to re-authenticate themselves?
thanks,
Paradise Dove
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I have the same problem.
seems that tdm400b is not working on CVS HEAD
On 6/18/05, Steve Totaro [EMAIL PROTECTED] wrote:
did you udate first?
- Original Message -
From: David Romero
To: Asterisk-Users@lists.digium.com
Sent: Friday, June 17, 2005 9:36 AM
Subject:
i have the same problem.
it seems to be a bug.
On 6/5/05, Master Abi [EMAIL PROTECTED] wrote:
Hi
I am trying to develop a night divert. Caller dials in after hours on
Zap and it gets divert to a mobile number via a second Zap. The call
bridges but will not hangup the channels when the
Is there any way to detect * deadlocks automatically?
i.e with a running program in background.
Paradise Dove
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i'm using latest CVS Head.
any ideas?
Thanks,
Paradise Dove
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i'm using latest CVS Head.
any ideas?
Thanks,
Paradise Dove
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what about CAS 3 Bit?
does * support it?
thanks,
Paradise Dove
On 4/8/05, Steve Underwood [EMAIL PROTECTED] wrote:
David Hajek wrote:
Hi,
is it possible to use Asterisk with T110P and CAS (channel associated
signalling)?
There are hundreds of CAS protocols. Quite a few currently
also add snom-190 and snom-360 to your list
PolyCom 500 and 600 have the same feature too.
On 4/15/05, Brian Leyton [EMAIL PROTECTED] wrote:
Or Flash Operator Panel. http://www.asternic.org
Brian Leyton
IT Manager
Commercial Petroleum Equipment
-Original Message-
From:
i have the same plroblem.
no link on xten site!
On Thu, 31 Mar 2005 14:49:11 -0300, Carlos Gabriel Drach
[EMAIL PROTECTED] wrote:
Kris Edwards wrote:
This is the best linux sip phone I've used so far. Audio quality has
been perfect and it seems really stable, so hopefully it will be out of
consider this scenario:
A Calls B
B transfers A to C
C (is busy or does not answer) so A backs to B
On Tue, 1 Mar 2005 23:07:17 +0330, Paradise Dove [EMAIL PROTECTED] wrote:
consider this scenario:
A Calls B
B transfers A to C
C (is busy or does not answer) so B backs to A
On Tue, 1 Mar
upgrade to latest CVS Stable.
it's solved there!
On Wed, 2 Mar 2005 22:02:58 -0600, Eric Rees [EMAIL PROTECTED] wrote:
I had asterisk 1.0.5 running fine. I upgraded to 1.0.6 and now the
music on hold does not work.
More Detail:
While I was running asterisk 1.0.5, when someone called
hi,
is there anyway to implement callback on busy and callback on no answer
on asterisk? has anybody done this before?
thanks,
Paradise Dove
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consider this scenario:
A Calls B
B transfers A to C
C (is busy or does not answer) so B backs to A
On Tue, 1 Mar 2005 14:25:33 -0500, C F [EMAIL PROTECTED] wrote:
use retrydial.
in the cli type show application retrydial
have fun.
On Tue, 1 Mar 2005 22:17:35 +0330, Paradise Dove [EMAIL
does MD 110 support SIP?
On Thu, 24 Feb 2005 15:08:49 +, Niksa Baldun [EMAIL PROTECTED] wrote:
Your span definition should be fine (except there should be commas
instead of dots, but that is probably just a typo). You need to play
with various parameters on the MD-110 side, those in RODAI
what about senao SI-7800H?
this is the link:
http://www.senao.com.tw/english/product/product_wireless01_outdoor_1.asp?pgtl=Wirelesstp1id=02tp2id=06proid=000131
On Mon, 21 Feb 2005 23:42:30 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
Kurt Fankhauser wrote:
Sounds like I'm going to
what is the password for Administrator in the softphone?
On Tue, 8 Feb 2005 08:01:07 +0100, Christian Stredicke
[EMAIL PROTECTED] wrote:
Go to the web page, in Preferences there are two pull down menus for
Audio Input and Autio Output.
CS
-Original Message-
From: [EMAIL
but when i remove uhci_hcd module i will fall in a big trouble, look:
the problem will solve when i load uhci_hcd again!! i've a TE405P card
installed and modules loaded.
Feb 6 08:11:16 WARNING[2907]: Failed to create new channel thread
Feb 6 08:11:16 WARNING[2907]: Failed to start PBX :(
Feb
submit a bug in bug tracker at http://bugs.digium.com
On Wed, 2 Feb 2005 21:55:16 +0100, Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
I've spotted weird crash of Asterisk cvs Stable. I have defined queue in
queues.conf :
[prodaja]
music = default
announce = queue-markq
strategy =
what is the meaning of (cause 0).
i know that in * code it indicates an undefined cause but that's not enough.
i have many of this message in my logs.
what would be the posiible causes for this message?
i have also the same message with SIP channels...
thanks,
Paradise Dove
but still the main question mark remains:
what are the possible causes which make this warning appear
thanks!
On Tue, 1 Feb 2005 18:10:03 +0330, Paradise Dove [EMAIL PROTECTED] wrote:
but still the main question mark remains:
what are the possible causes which make this warning
so you mean that it depends on the type of motherboard and the chipset
which is using. am i right?
if yes, which mainboards and chipsets is recommended for a large scale * box?
On Mon, 31 Jan 2005 12:19:50 -0800, William Boehlke
[EMAIL PROTECTED] wrote:
On Intel it is our experience that the
hi all,
just want to know, if there is any workaround to add SRTP support to *.
as i know there is an open source library (libsrtp
http://srtp.sourceforge.net/srtp.html) which makes it more possible to
be done.
any idea?
thanks,
Paradise Dove
hi,
just got an strange crash, and don't know what could cause this type of crashs
- hardware failure
- memory
- cpu
?
i have 1xTE405P installed with 4xTA750. using fresh kernel 2.6.9 (no patch).
* version is latest CVS HEAD.
thanks
Program terminated with signal 11, Segmentation fault.
this is what i've typed to get the crash info:
gdb /usr/sbin/asterisk --core=/core.3673
is it wrong?
On Sun, 30 Jan 2005 03:11:24 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Sun, 2005-01-30 at 12:31 +0330, Paradise Dove wrote:
hi,
just got an strange crash, and don't know what
the same result!
On Sun, 30 Jan 2005 03:24:38 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Sun, 2005-01-30 at 12:46 +0330, Paradise Dove wrote:
this is what i've typed to get the crash info:
gdb /usr/sbin/asterisk --core=/core.3673
Not sure if that is wrong, but I also see
i have the same problem...
i've also added a feature request to bug tracker
(http://bugs.digium.com/bug_view_page.php?bug_id=0002612) regarding
this issue.
On Sun, 30 Jan 2005 13:40:06 -0600, Jon Gabrielson
[EMAIL PROTECTED] wrote:
Can't asterisk look for a dialtone? Even a $5 modem
can
that any followups could be done there. maybe setting bounty
on this issue speedup the process!
thanks,
Paradise Dove
Jon.
On Sunday 30 January 2005 04:13 pm, Steven Critchfield wrote:
On Sun, 2005-01-30 at 13:40 -0600, Jon Gabrielson wrote:
Can't asterisk look for a dialtone
it would help to know all the possible causes for this warning,
something like:
- kernel
- hardware latency (MB, cpu, ...)
- buggy sip device
- lack of resource
- ...
just let us know if anybody knows.
thanks,
Paradise Dove
On Mon, 24 Jan 2005 11:28:34 -0600, [EMAIL PROTECTED]
[EMAIL
polycom is better for the same quality and lower price.
On Sun, 16 Jan 2005 17:27:20 -0800 (PST), Robert Augustyn
[EMAIL PROTECTED] wrote:
Any preferences?
And why?
Thanks in advance.
robert
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i have no problem with 2.6.
On Sat, 15 Jan 2005 13:12:18 +, Jeremy SALMON
[EMAIL PROTECTED] wrote:
Hi,
Just a question,
For you, what is the more reliable kernel for an asterisk prod server...
Thanks
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type these 3 command inorder to get CVS HEAD.
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
cvs login
cvs checkout zaptel libpri asterisk
On Wed, 12 Jan 2005 16:20:06 +, John Middleton
[EMAIL PROTECTED] wrote:
When you say CVS HEAD is the the same as stable? where do you get it
just to make sure:
when i have zaptel devices on my box and i also use meetme and iax2,
do i need to have USB device enabled and it's modules loaded?
On Wed, 12 Jan 2005 12:24:55 +, Bob Goddard [EMAIL PROTECTED] wrote:
On Tuesday 11 January 2005 23:01, Warren Burstein wrote:
Michael
I don't think it's possible. Asterisk would have to emulate analog modem,
does anybody know if there ia any works on emulating analog modems
(not specially to work with asterisk).
something like Steve's spandsp for fax.
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it's clear that your processor is overloaded.
recommend you to use rawplayer instead of mpg123 for moh
by converting your mp3 files to raw using sox (with mp3 support)
take a look at cvs head.
On Mon, 10 Jan 2005 06:45:54 -0800 (PST), Jason Goecke
[EMAIL PROTECTED] wrote:
Hello,
Ever since
the only way is to set callprogress=yes but it's very experimental
and makes many wrong alarms.
by the way this feature is really missing in *.
On Sat, 08 Jan 2005 17:42:42 +0200, Gilad Ben-Yossef
[EMAIL PROTECTED] wrote:
Samudra E. Haque wrote:
hello, using Asterisk, is there any clever way
i dont know how to use * native moh feature which is added recently to CVS HEAD
each time i hold a call i will get this warning on cli:
WARNING[24235]: res_musiconhold.c:837 local_ast_moh_start: No class: default
Paradise Dove
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does TE405P support 3Bit CAS?
what are the configuration tips?
thanx,
Paradise Dove
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I get seg. fault with my * box. at the crash time i had about 35
Bridged Channel.
i have:
- dual xeon box (3.2Ghz)
- 2Gb of memory
- E7501 chipset motherboard.
- U320 scsi disks
- intel Gb ethernet device.
- i only use sip for clients (no fxs in box)
- TE405P for fxo (with 4 atran TA750).
- ulaw
I'm using FC2. but with a fresh 2.6.9 kernel downloaded from kernel.org.
I've recently upgraded my Glibc to glibc-2.3.3-27.1.
I'm also using ECC Reg Memory.
and this is my Xeon CPU info: (HyperThreading is ON)
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model
I got another crash... the core dumped file shows that the
crash has been occurred at the same point as the previous crash.
Program terminated with signal 11, Segmentation fault.
#0 0xb7fbbce4 in ?? ()
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I have seen lots of this kind of problems before. We
had lots of stability problems with GNUgk on Debian
Woody.
is there any relation between * and GNUgk?
thanks
Paradise Dove
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it (upgrading a bios) seem to fix the problem?
thanks,
Paradise Dove
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I'm using an A101u and it seems to work fine connected to a
Carrier Access Access Bank I (24 FXS).
How did you get it working with asterisk?
- Paradise Dove
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what does this warning really mean?
does it have any side effect on my * box? 'cose I've recently had
random seg. faults on my box.
I'm using latest CVS -r v1-0
Dec 1 12:08:42 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries!
Dec 1 12:08:43 WARNING[6189]: Avoided deadlock for
Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked:
Avoided deadlock for 'SIP/2502-6303', 10 retries!
Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked:
Avoided deadlock for 'SIP/2502-6303', 10 retries!
Dec 1 12:08:44 WARNING[6189]: channel.c:495
/2502-6303' for
lack of RTP activity in 4795 seconds
Dec 1 12:44:47 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for
lack of RTP activity in 4795 seconds
Paradise Dove
Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked:
Avoided deadlock for 'SIP/2502-6303', 10 retries!
Dec
[6189]: channel.c:495 ast_channel_walk_locked: Avoided
deadlock for 'SIP/2502-6303', 10 retries!
I have these two lines in my sip.conf
rtptimeout=300
rtpholdtimeout=480
it seems that these options don't work as expected.
Paradise Dove
On Wed, 1 Dec 2004 07:11:53 -0500, mattf [EMAIL PROTECTED
all i have is random echo
I have already 4 TA750 with full FXO
echocancel=yes and echo training=800
- what should i do?
- could it be solved with tweaking echo params on *?
- is there any additional devices that can be added between Channel
Bank and * to get rid off echo forever?
if its
appreciated
Paradise Dove
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how can i get a CCO account?
or is there any other place for cisco downloadable stuff without user/pass?
or a free to all CCO account!!!??
On Fri, 12 Nov 2004 10:50:18 -0600, Eric Wieling [EMAIL PROTECTED] wrote:
You CANNOT download Cisco firmware without a CCO account AND support
contract.
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