This was posted to me by Vegastream tech support in regards to your
earlier question (I emailed them your question last week), sorry I'm
just getting familiar with both boxes so I'm not able to help you at
this stage I have just signed a deal for distribution of the vegastream
here in Australia
--On Sunday, March 21, 2004 8:11 pm + Dee Lowndes [EMAIL PROTECTED]
wrote:
If I find the voltage drop out can I configure the x100p to do it based on
the new voltage drop. If so where and how?
To a certain extent yes. Im fact, in the absence of measurements you could
just try a couple of
I'm jsut starting to setup an Asterisk system (my first) for a new office
and I'm having some problems getting DTMF tones working correctly with
Dial-Com Lite (software SIP phone). The only DTMF option that won't generate
console errors is rfc2833 but the keys are doubling up (when I press 1,
11
Hi everybody,
Recently I've sent a mail to Astersik Doc Mailing list but it seems that
it is not arrived as well.
Can you check this:http://lapindigo.free.fr/asterisk/astdiagram.htm to
see if my diagram is like it will be to represent a higher point of view
of Asterisk.
Thank you in advance.
I'm using asterisk and it works ok the only thing is thata ecvery 2/3
days get the cpu load up to 99% and the only way I can shutdown the
service is to use a killall -9 asterisk.
any suggestions?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hello,
I have an X100P connected to a phone line that is used for ADSL. The DSL
modem is connected directly to the wall jack. A regular phone was
connected on this line through an ADSL line filter.
The audio part works with * just like it does on the phone, but it
appears callerid is not
Matteo Rancilio wrote:
I'm using asterisk and it works ok the only thing is thata ecvery 2/3
days get the cpu load up to 99% and the only way I can shutdown the
service is to use a killall -9 asterisk.
any suggestions?
Nope, but you have to provide us with more information about your
I use libssl-dev out of the debian testing tree.
---
Zac Amsler
Computer Consulting Group
WNOC.COM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas Schroeter
Sent: Sunday, March 21, 2004 3:10 PM
To: [EMAIL PROTECTED]
Olle E. Johansson ha scritto:
Nope, but you have to provide us with more information about your
configuration to be able to help you. What O/S version?
Any hardware cards, what configuration of Asterisk?
/O
___
You're right :)
I'm using Asterisk 7.2 on
Is it possible to change the var CALLERIDNUM?
I need to put the 0 in front of the incoming number to be able to make a
redial on a missing call.
We need the 0 to rich the external line
I tried with
exten = s,1,Answer
exten = s,2,Setvar(CALLERIDNUM=0${CALLERIDNUM})
On console I see the new
Hi,
I would like to use Asterisk as a SIP B2BUA, for CDR generation. I prefer to
avoid doing RTP proxy in Asterisk for SIP UAs for increasing performance.
Could i configure sip channel for that? Some kind of dont_touch_sdp=1?
Thank you very much.
G.
-Mensaje original-
De: [EMAIL
Hi,
I am a new user and I would like to find a documentation about
asterisk, I have a computer linux with hardware digium, (xfp X100P , ip
phone, etc..) and I would like to test it. I have install now in my
computer(linux), the software asterisk, zaptel and libpri, so I donacute;t
know what can
--On Monday, March 22, 2004 12:51 pm +0100 Matteo Rancilio
[EMAIL PROTECTED] wrote:
You're right :)
I'm using Asterisk 7.2 on a SuSE 8.2 installation.
Hardware:
Dual Intel PIII
1Gb ram
AVM Fritz! ISDN card
SIP
CISCO Phones
Codec g711 (switching today to g729)
... and what applications? AGI,
Matteo Rancilio ha scritto:
Is it possible to change the var CALLERIDNUM?
I need to put the 0 in front of the incoming number to be able to make
a redial on a missing call.
We need the 0 to rich the external line
I tried with
exten = s,1,Answer
exten = s,2,Setvar(CALLERIDNUM=0${CALLERIDNUM})
On
Iain Stevenson ha scritto:
--On Monday, March 22, 2004 12:51 pm +0100 Matteo Rancilio
[EMAIL PROTECTED] wrote:
You're right :)
I'm using Asterisk 7.2 on a SuSE 8.2 installation.
Hardware:
Dual Intel PIII
1Gb ram
AVM Fritz! ISDN card
SIP
CISCO Phones
Codec g711 (switching today to g729)
...
Make sure you have done a cvs update. We had a cpu problem where we were
hitting 99.9% on every call. An update sorted it out.
Thanks
Tan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Iain
Stevenson
Sent: 22 March 2004 12:14
To: [EMAIL PROTECTED]
http://bugs.digium.com/bug_view_page.php?bug_id=899
A patch that improves the DTMF support for ISDN4Linux and adds functionality for
CallerID handling with EuroISDN networks.
This patch needs testing and comments on the bug tracker. If you're using ISDN4Linux
(not CAPI) and have spare time to
[EMAIL PROTECTED] ha scritto:
Make sure you have done a cvs update. We had a cpu problem where we were
hitting 99.9% on every call. An update sorted it out.
Thanks
Tan
Thanx Tan,
I'm using the tarball. Is it still affected?
___
Asterisk-Users
Hello,
Is there some project about integration Asterisk with Radius?
If yes, where I can find it?
--
Best regards,
~*-,._.,-*~'`^`'~*-,._.,-*~'`^`'~*-,.
Mike Tkachuk, ph:380-3433-67067
YES ISP, fx:380-3433-67067
Valova 17,[EMAIL PROTECTED]
Kolomyia,
randulo wrote:
I have an X100P connected to a phone line that is used for ADSL. The DSL
modem is connected directly to the wall jack. A regular phone was
connected on this line through an ADSL line filter.
DEFINITIVE ANSWER:
One kind soul responded off list. At any rate, it was the phone company
Senad Jordanovic wrote:
And yes, there's a config in iax.conf so you can turn it off if you
for some reason want to bother B with staying in the middle of the
call.
Yap. Great stuff :)
Just so everyone knows the config is: notransfer=yes
It would be good to know what
I am watching memory usingTop and when Asterisk is running, memory usage increases 8 k every 10 seconds or so. I stop Asterisk and this memory increase stops. This doesn't sound normal. Anybody else experiencing this?
Sorry, my previous post was almost unreadable.
I am watching memory using Top and when Asterisk is
running, memory usage increases 8 k every 10 seconds
or so. I stop Asterisk and this memory increase
stops. This doesn't sound normal. Anybody else
experiencing this?
Hi all,
If you have had trouble with multiple concurrent channels running
app_rxfax or ap_txfax, where was a silly bug. Updated versions are
available at ftp://ftp.opencall.org/pub/spandsp
The latest spandsp-0.0.1f seems to working for quite a lot of people. I
guess there will still be plenty
Thanks a lot, I'll look into it from here.
AJ
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi,
INSTALLED ASTERISK PROPERLY AND WORKS FINE. AFTER I
MADE A WRONG DECISION TO INSTALL iaxComm in the
Client, it's created problem. When i type asterisk -r
command, Now i got display as
[EMAIL PROTECTED] asterisk]# asterisk -r
Asterisk CVS-03/18/04-18:01:45, Copyright (C)
1999-2004 Digium.
Title: Message
I posted this
problem another time, but with another problem tied in...so let's try this
out.
I've got an X100P
and a T100P on the same box (the x100p was initially for testing, but since it's
working fine we are still using it). However, the X100P is tied into a
different
where can we find configuration for the LipZ4 Sip Soft Phone for * and
how do you configure * for the LipZ4 Sip Soft Phone?
same questions for xten lite.
thanks, eliot
On Mon, 2004-03-22 at 09:34, [EMAIL PROTECTED] wrote:
Thanks a lot, I'll look into it from here.
AJ
On Mon, 22 Mar 2004 06:31:00 -0800 (PST), suresh kumar [EMAIL PROTECTED]
wrote:
Hi,
PLEASE STOP YELLING
INSTALLED ASTERISK PROPERLY AND WORKS FINE. AFTER I
MADE A WRONG DECISION TO INSTALL iaxComm in the
Client, it's created problem. When i type asterisk -r
command, Now i got display as
[EMAIL
Title: 7960 Configuration question
First, thank you all I have been able to get my system(s) up and running just by searching the list and I thank you all for your previously answered questions..
However there's something I couldn't find..
Is it possible with the Cisco 7960 to define
The Asterisk Demo prompts come through loud and clear on any phone
that I use to call in on. When someone leaves me voicemail, it also
comes through loud and clear.
When I use the Record application and then use the recorded file in
a Playback or Background application, it is very soft (clear,
On Mon, 2004-03-22 at 07:53, Jeffrey Thomas wrote:
I am watching memory using Top and when Asterisk is running, memory
usage increases 8 k every 10 seconds or so. I stop Asterisk and this
memory increase stops. This doesn't sound normal. Anybody else
experiencing this?
Please do not post
Here is the app_radius project URL, we are finishing our beta tests
and the project applications will be released on 02 april 2004
http://appradius.minitelecom.org/
Lubo
Mike Tkachuk wrote:
Hello,
Is there some project about integration Asterisk with Radius?
If yes, where I can find it?
Hello,
Does anybody have already use TEDAS VoIP DECT PABX or some Kirk hardware
to integrate DECT to VoIP?
Thanks!
Ignace
___
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To UNSUBSCRIBE or
Try
exten = s,1,Answer
exten = s,2,SetCallerID(0${CALLERIDNUM})
-Alfred
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matteo
Rancilio
Sent: Monday, March 22, 2004 4:12 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] setvar CALLERIDNUM
Matteo
On Mon, 2004-03-22 at 09:41, Chris Tooley wrote:
Can this be used as a RAS product to develop a PPP connection?
Only for ISDN connections. There is no modem software for zap hardware
other than the fax app.
--
Steven Critchfield [EMAIL PROTECTED]
I have latest asterisk running on redhat 9;
I use mediatrix gateways running SIP protocol.
I have installed hisax compatible passive adapter on my asterisk box
(HFC-S PCI Active chip).
Problem is following; when I dial through my ISDN adapter and run echo
test I got excellent response (clear
On Mar 22, 2004, at 10:41 AM, Chris Tooley wrote:
Can this be used as a RAS product to develop a PPP connection?
Chris
Should work -- more info at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20zapras
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20ZapRAS
Note -- I have
one more thing
I have just configured so that I enter asterisk through ttyI0 and then
exit back to PSTN (or in my case ISDN) thru ttyI1 (second B channel on
the same adapter) and zero problems, sound is perfect, no jittering,
breaks or any problem whatsoever
so something happens in between
I use mediatrix gateways running SIP protocol.
I have installed hisax compatible passive adapter on my asterisk box
(HFC-S PCI Active chip).
Problem is following; when I dial through my ISDN adapter and run echo
test I got excellent response (clear sound, no breaks), when I connect
my
Does anybody have already use TEDAS VoIP DECT PABX or some Kirk hardware
to integrate DECT to VoIP?
I've tried integrate Kirk, but with no special success.
There was any codec problem, so it was only
'one way voice' integrated.
I'v tried only with skinny and with standard Kirk configuration.
Hi,
I just equipped my asterisk with a second fritzcard, in order to watch
two different ISDN connections and to bridge between them.
Unfortunately, the second fritzcard doesn't get recognized by asterisk.
This is what my config files look like:
/etc/capi.conf:
#SuSEconfig.isdn generated
For info,
I receive the mailing list on a brand new account that is not used for
anything else.
Just received, a virus (*apparently*) From: [EMAIL PROTECTED]
I suppose there may be 8,000 people getting it but just in case.
___
Asterisk-Users mailing
I have a problem trying to redial a number from Missed Calls menu in
Directories, I get this error with both 7960 and 7912 series:
chan_sip.c:5282 handle_request: Failed to authenticate user USERNAME sip:[EMAIL PROTECTED];
Redialing from Received Calls and Placed Calls works fine.
Anyone with
I've looked at the * docs and wiki further, and sketched out this map of
how the phone/network at our call center (4 seats) would work.
- From the CO, 4 phone lines which go into ...
- a channel bank with 4-8 FXO interfaces and a T1 interface which leads
into ...
- the * server, with a T100P
Title: Message
Please
post the portion of your dialplan that you are explaining. More than
likely you don't have an "r" in your dial command. That lets the
calling party hear a ring.
e.g.
Dial(SIP/1234|20|Tr)
-sb
-Original Message-From:
[EMAIL PROTECTED]
Hey All,
I'm developing a reception style console (like many others) to answer incoming calls
to a main line number, request who they want to speak to and then have the
receptionist call the desired party and announce the calling party before putting them
through.
This should be fairly
On Mon, 2004-03-22 at 09:11, Hadar Pedhazur wrote:
The Asterisk Demo prompts come through loud and clear on any phone
that I use to call in on. When someone leaves me voicemail, it also
comes through loud and clear.
When I use the Record application and then use the recorded file in
a
Hi,
I´m a member from this list and I do not have a virus because I check my email
from my server linux.
cheers.
vozip
Mensaje citado por randulo [EMAIL PROTECTED]:
For info,
I receive the mailing list on a brand new account that is not used for
anything else.
Just received, a
On 22/03/04 17:58 +0100, randulo wrote:
For info,
I receive the mailing list on a brand new account that is not used for
anything else.
Just received, a virus (*apparently*) From: [EMAIL PROTECTED]
I suppose there may be 8,000 people getting it but just in case.
No, not necessarily.
Well, a SIP client inside a netword with a firewall without portforwarding to this
, and a asterisk server in another network, this have a internet public ip.
The client can connect to the server?, this is possible?
The port forwarding is neccesary?, only will rule a client with portforwarding?
Hi,
I tried to install the following hack.
http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO
But the 2nd AVM Fritz PCI card is still not showing up.
My environment is:
debian 2.4.24
(asterisk 0.72)
Just a quick explanation what I did:
- edited the filed as described above
- make clean
- make
Hey Steve,
On Mon, Mar 22, 2004 at 22:13:37 +0800, Steve Underwood wrote:
Hi all,
If you have had trouble with multiple concurrent channels running
app_rxfax or ap_txfax, where was a silly bug. Updated versions are
available at ftp://ftp.opencall.org/pub/spandsp
The latest
You could simply redirect both of them to a meetme room with the 'q' flag
set for no messages. I'm using that method for an application right now.
MATT---
-Original Message-
From: Low, Adam [mailto:[EMAIL PROTECTED]
Sent: Monday, March 22, 2004 12:15 PM
To: '[EMAIL PROTECTED]'
Subject:
Hallo Jakob Strebel
On Mon, 22 Mar 2004 18:27:44 +0100 you wrote:
Hi,
I tried to install the following hack.
http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO
But the 2nd AVM Fritz PCI card is still not showing up.
My environment is:
debian 2.4.24
(asterisk 0.72)
Just a quick
[EMAIL PROTECTED] wrote:
Hi,
I´m a member from this list and I do not have a virus because I check my email
from my server linux.
cheers.
vozip
Mensaje citado por randulo [EMAIL PROTECTED]:
For info,
I receive the mailing list on a brand new account that is not used for
anything else.
On Mon, 2004-03-22 at 11:17, [EMAIL PROTECTED] wrote:
Mensaje citado por randulo [EMAIL PROTECTED]:
Just received, a virus (*apparently*) From: [EMAIL PROTECTED]
Hi,
I´m a member from this list and I do not have a virus because I check my email
from my server linux.
Notice the
Title: Message
Thanks
for the response. Here are two contexts from my extensions.conf. The
number being dialed is in the "bob" context.
[bob]exten = s,1,Goto(uti-mainst,2450,1)
include =defaultinclude =outboundinclude
=uti-mainst
[uti-mainst]
exten
= 2450,1,Dial(SIP/bob,45,Ttr)
include
Hi Azher-
I experience many of the same problems, especially the frame
retransmissions, which cause channel hangups from time to time. These
channel hangups clear themselves after a few minutes normally but do
interfere with calls.
I'm not the expert on Linux internals, so I'm not sure what can
Title: Message
Nothing is jumping out.
Why
don't you try simplifying your dialplan a little without all the gotos and
includes, and see if you can get it to ring.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Mark
Messmore, Technical
On Sat, 2004-03-20 at 16:36, Michael Devenijn wrote:
Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out
why the call doesn't go trough ...
extensions.conf extract (from the contact [tlsgw]) :
exten = 57228047,Dial(SIP/cs001,40,tr)
The above line does not
On Mon, Mar 22, 2004 at 10:16:24AM -0800, Scott Laird wrote:
Can someone with a 7905 jump in? The wiki says that the display on the
7905 is higher resolution then the 7960, but doesn't give any details.
How many lines of text does it show, and does it use a 1-line or 2-line
format for
Title: Message
I did
that just after I sent off the note to you...still nothing.
Mark
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bisker,
Scott (7805)Sent: Monday, March 22, 2004 1:34 PMTo:
[EMAIL PROTECTED]Subject: RE:
Dear Scott,
Thnx for the details. I will talk with Mark on these issues today.
Regards
Azher
---
http://www.consulttech.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Stingel
Sent: Monday, March 22, 2004 11:29
Thomas,
thank you for the reply.
Looks like you did not all the steps!
Don't edit the source, you have to edit the binary file!
Do you mean with the binary file ../lib/fcpci-lib.o ?
I have two in this directory
one the original fcpci-lib.o
and the f2pci-lib.o in which I did the replacements
On Mar 22, 2004, at 10:48 AM, Walker Haddock wrote:
On Mon, Mar 22, 2004 at 10:16:24AM -0800, Scott Laird wrote:
Can someone with a 7905 jump in? The wiki says that the display on
the
7905 is higher resolution then the 7960, but doesn't give any details.
How many lines of text does it show,
hi..??
do you know web site where i can download document about install and configure
software asterisk and zaptel...??
please.!
Cheers.
vozip
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Progress
It seems I can't hear the Say Time, due to RTP Double NAT
I'm guess this is both problem 1 and 2 really issue.
My config:
IP Phone - Router (Nat) - Internet - Linux (NAT) - * Server
ANyone know of work arounds the double NAT? or other methods
to route RTP with snom 200s, to work
Thanks Steve. Right after I sent my note, it occurred to me that I
could create a dummy voicemail account, and point my current Record
extension to that, and use Voicemail to record the higher volume
version.
I just did one quick test, unscientific at best, and I think the above
works
Hi Michael,
Thanks a lot for your help.
I can see the four files when i untar the iaxComm.
How can i get the *CLI prompt?
Is it possible to uninstall the asterisk package?
If so, how can i uninstall so that i can configure
asterisk with iaxComm.
Thanks Regards,
Suresh
--- Michael Van
At 11:39 AM 3/22/2004, you wrote:
Progress
It seems I can't hear the Say Time, due to RTP Double NAT
I'm guess this is both problem 1 and 2 really issue.
My config:
IP Phone - Router (Nat) - Internet - Linux (NAT) - * Server
ANyone know of work arounds the double NAT? or other methods
to route
Just upgraded to stable CVS-03/20/04-11:54:56
C7960 - * C7960 (all on the same wire), call from on phone
to the other does not receive any ringback signal. Total silence
while the phone is actually ringing, then hear voicemail anouncements
after the 15 second timeout.
Was working fine before
Title: RE: [Asterisk-Users] SoftFAX/spandsp
Hi,
With the latest release available (spandsp-0.0.1f) we still cannot receive
faxes from Dialogic. There are no problems with several fax machines we've tried and
it works fine with J2, so it shouldn't be slipping...
-- Executing
Hello all,
Through the previous two weeks I have been working on my first
asterisk installation. So far all my doubts and problems were aswered by
the list history or on-line documentation. I have a two SIP softphones +
external analog line working fine. However, I just came across a
If you look at
http://www.cisco.com/image/jpeg/en/us/guest/products/ps379/c1122/cdccont_0900aecd800add13.jpg
you'll see the screens on the 7905/7912 (bottom left two) are
substantially smaller than the 7940/7960 (top right three). So for a
similar pixel size (if that is the case) the
Hi everyone,
I have the problem not exactly like this one but here is the whole
story...
I have installed everything about 3-4 months ago and it worked fine for
some time. After a while I noticed that calls drop from time to time,
and after few weeks of troubleshooting I asked Digium support to
Hi List,
I'm trying to get the DIALED NUMBER on a inbound
800 call.
I need to know what number the calling party called
so that I can route the call properly.
I really don't want to burn a DID per WATTS line
so that I can route on the DID number.
Any pointers??
john brown
Hi Tomica-
Please let us all know what the resolution is.
Digium has been cooperative in the past on this problem (PRI driver issues,
if that's what is causing your problems), but recently they've had their
hands quite full with new features and a long bug list.
I have actually demonstrated the
On Mon, 2004-03-22 at 14:35, John Brown (CV) wrote:
I really don't want to burn a DID per WATTS line
so that I can route on the DID number.
Look at asterisk/doc/README.variables. Assuming you are using ZAP
interfaces look at ${DNID}.
--
Eric Wieling * BTEL Consulting * 504-899-1387
On Sunday 21 March 2004 07:10, Steve Underwood wrote:
I have received more excellent problem report information, and I
have resolved a number of issues affecting my soft FAX machine when
working with various models of real FAX machine. The code now seems
to be working with a much greater range
I've had my asterisk running for a couple of weeks and just noticed that it
takes about 98% of the CPU time (Linux RH9). Is this what you would expect?
Is it just that the program is polling for things to do, calling sleep(0)
or something simlar so as to relinquish the machine but otherwise
On Mon, 22 Mar 2004 08:00:58 -0800 (PST), suresh kumar [EMAIL PROTECTED]
wrote:
Hi Michael,
Thanks a lot for your help.
I can see the four files when i untar the iaxComm.
How can i get the *CLI prompt?
Just as you have been: asterisk -r
Is it possible to uninstall the asterisk package?
If so,
On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote:
I've had my asterisk running for a couple of weeks and just noticed that it
takes about 98% of the CPU time (Linux RH9). Is this what you would expect?
Is it just that the program is polling for things to do, calling sleep(0)
or something simlar
Hi all,
I may need to connect to a system with EM connectivity.
Am I right in assuming a T1 card and Channel Bank will give me this
connectivity?
Regards
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
What is it about asterisk that makes this happen? My other apps that wait
on a select take hardly any CPU time at all.
I didn't find anything like ldassume using google. Can you tell me more
about that?
Thanks,
Bill.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Barry Fawthrop wrote:
Progress
It seems I can't hear the Say Time, due to RTP Double NAT
I'm guess this is both problem 1 and 2 really issue.
My config:
IP Phone - Router (Nat) - Internet - Linux (NAT) - * Server
ANyone know of work arounds the double NAT? or other methods
to route RTP with
The X100P hangup problem is indeed pervasive. My current
testbed has the X100P connecting to an FXS breakout of a
dual ISDN channel box. Indeed, remote hangup is NOT
detected. When I switched it to a POTS line, all the sudden
it seemed to work OK. This is a serious limitation in some
scenarios
David J Carter wrote:
I may need to connect to a system with EM connectivity.
Am I right in assuming a T1 card and Channel Bank will give me this
connectivity?
Depends on the channel bank -- make sure it supports EM.
If you don't need to support a lot of EM trunks you can take a look at
On Mon, 2004-03-22 at 15:35, David J Carter wrote:
Hi all,
I may need to connect to a system with EM connectivity.
Am I right in assuming a T1 card and Channel Bank will give me this
connectivity?
A T1 card will. If the channel bank is populated with the right cards,
it may be able to
I think Steve is referring to the following line:
export LD_ASSUME_KERNEL=2.4.1
If you put this in your command line before starting asterisk, you will get
around the RH9 problem of leaving zombies when AGI processes quit. Other
than that, I don't think it influences CPU load.
Note that the
Bill,
I think your looking for setting the environment variable
LD_ASSUME_KERNEL=2.4.1.
If I remember correctly this effectively disables the new NTPL (new
threading model) in RH9.
Hope this helps.
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On Mon, 2004-03-22 at 15:26, [EMAIL PROTECTED] wrote:
The X100P hangup problem is indeed pervasive. My current
testbed has the X100P connecting to an FXS breakout of a
dual ISDN channel box. Indeed, remote hangup is NOT
detected. When I switched it to a POTS line, all the sudden
it seemed
export LD_ASSUME_KERNEL=2.4.1
- Original Message -
From: Bill Hamlin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 22, 2004 4:22 PM
Subject: RE: [Asterisk-Users] question about CPU usage
What is it about asterisk that makes this happen? My other apps that wait
on a
On Mon, 2004-03-22 at 15:22, Bill Hamlin wrote:
I didn't find anything like ldassume using google. Can you tell me more
about that?
It's in the RedHat 9 RELEASE NOTES.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the
On Mon, 22 Mar 2004 [EMAIL PROTECTED] wrote:
do you know web site where i can download document about install and configure
software asterisk and zaptel...??
the final lines of the file README included in the asterisk-0.7.1 tarball:
- begin -
* MORE INFORMATION
See the doc directory
On Monday 22 March 2004 15:35, David J Carter wrote:
I may need to connect to a system with EM connectivity.
Am I right in assuming a T1 card and Channel Bank will give me this
connectivity?
You typically do not need a channel bank when using EM. However,
you will probably need a T1 crossover
You must have port mapping in the Linux NAT that allows the SIP-level
packets to get to the * Server, so you need to add a port mapping for the
RTP packets. I may be wrong but I think * sends RTP on the same port that
it receives RTP on, so once the phone sends some RTP to * then the RTP
coming
Quick question -- How do I continue a macro after hangup (I need to run
a script)
I'm using RxFax(Spandsp) and it exits -1 (I even changed the code to
return 0, but no luck)
My macro ends with -- Hungup 'Zap/3-1'
Here is the Macro (the System(echo)'s are for debugging)
(Only the first System
Thomas,
I restarted the hack and found that I did not edit all the files in
src.drv. My mistake sorry.
[EMAIL PROTECTED]:/usr/src/fritz2/src.drv$ grep _fcpci_ *
defs.h:#elif defined (__fcpci__)
driver.c:#elif defined (__fcpci__)
driver.c:#if defined (__fcpci__)
driver.c:#if defined (__fcpci__)
Nope same problem. I just started it and did a couple of ps aux's and got
this output:
[EMAIL PROTECTED] root]# ps aux|grep ast
root 20140 91.6 1.3 115880 6676 ? R15:43 1:10
asterisk -vgcd
root 20221 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast
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