Dear Asterisk Group.
I have two Asterisk servers serving two data/help desk
centers, both centers have a near identical setup.
However, when connected to one of my data centers, I call a
user, I can see on the CLI that the phone is ringing, but I hear no ringing on
my SIP soft phone?
Hi
Out of interest, (this may be not possible) but I think it
would be an excellent idea to modify firmware to handle the
IAX2 protocol. Especially since its a linux based phone.
Thoughts?
Regards
Clive
On Mon, 19 Jul 2004 21:54:59 +
Joshua Colp [EMAIL PROTECTED] wrote:
Hello
Clive,
Freshtel who provide the Firefly IAX softphone have some IAX hardware
based phones coming out in the next few months.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, 21 July 2004 4:01 PM
To: [EMAIL
Is there an application I could use to test this? I.E. like the echo
test, but doesn't send anything back...
app_record.so ?
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To UNSUBSCRIBE or
The source code found heere http://www.holgerschurig.de/destar.html
is in an unsupported TAR format.
It isn't. It's tarred and bzip2'd. If your tar can't do this, then you can
resort to this:
bzip2 -d *.tar.bz2 | tar xv
But tar nowaday has the j option for bzip2 and the z option to gzip.
If you really need a iax2 capable softphone, you may check this:
http://www.holgerschurig.de/files/linux/qtiax-0.1.tar.bz2
Yeah, but you should continue to develop it and send in patches.
Currently I focus on DeStar, so qtiax (a Qt3 based IAX phone) doesn't get
much handholding. Currently, it
And, and I forgot: Patches welcome.
A nice tool to make patches is patcher (much easier to use Quilt, I'd
say). See http://www.holgerschurig.de/patcher.html
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Hi,
Is anyone aware of an integrated management tool for asterisk? Specifically,
I'm looking for something that can:
1) Generate CDR reports
2) Manage a 'switchboard'
3) Add/remove/edit extensions
So far I've seen applications that do one of the three, but I haven't come
across something that
That did it. I have the wcfxo running and channeled. Now I just have
to beat my dial pan. I can dial internally to all my SIPs but outbound
and inbound off the X100P are still not running. Do I just do this...
Define [incoming] in extensions
[incoming]
exten = 1234567,1,Dial(SIP/2000) ;
muralikrishnan lakshmanan wrote:
Hello friends,
I got one page from net http://www.voip-info.org/wiki-CLASS;
In that page I saw lot of *xx codes for asterisk feautres.
I don't know how to use these codes.
If anyone used these codes can you teach me.
This is just a list of
I'm having many troubles with x101p (orginal from Digium, wcfxo kernel module)
on analog line simulated by a Cisco Router
I'm experiency random hung up while zaptel doesn't recognize call progress
(Italy signalling)
Signalling is simple ignored as if someone hangs up on the other end of the
HI ALL;
I have an ATA phone registered with
GUNGK.Iwant to send a call to another ATA with has an extention in my *
box.
my network looks like the following:
(h323 registration)
ATA1(h323
ep)gungkasteriskATA2(h323
ext)
But when I try to send a call from ATA1 to ATA2, it
fails. I
Holger Schurig wrote:
When listening to GSM-compressed voice prompts from either G.729 or
iLBC codec, the sound quality is distinctly sub-optimal due to the use
of multiple transcoding.
Would
sox sound.gsm sound.au
help a little bit?
This should help with CPU usage, but not with actual
I have one TDM04b(4FXO) that BTW came with a broken module and I'm sending
the module to RMA.
The other channels work well with one phone but with some specific
brand/models don't work.
For example:
Sharp CJV-743W
http://www.sharp.co.jp/products/cj/index.html#cjv743w
Using the cordless phones
The extension of an incoming call through the X100P is s. So,
[incoming]
exten = s,1,Answer
exten = s,2,Dial(SIP/200)
exten = s,3,Hangup
[outgoing]
exten = _9.,1,Dial(ZAP/g1/${EXTEN,1})
You need to dial 9 from your SIP phone to get an outside line and then the
number you wish to dial.
g1
Welcome to the Asterisk users community!
Asterisk.org is a fast moving project. New code is added every day.
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Our community is also growing
Hi
Just receiveda couple ofSI-7800 wifi
phones.
nice looking phone, got it to work after a bit of a
headache, which I thought I would share.
sip.conf
[1007]type=friendusername=1007secret=blahhost=dynamiccontext=from-sipdisallow=allallow=ulaw
The phone has a problem selecting codec's so I
Hi, i've a proble using voicemail. when i make a call and start voicemail
asterisk tell me mail address is missing even if i used it as written
mailbox = name,pwd,[EMAIL PROTECTED]
I saw that modifying in app_voicemail.c line 836 in this manner: if (vmu
ast_strlen_zero(vmu-email)), so replacing
hi, i've a question. is it possible to buy digium x100p card from italy
in some store (also online) without ordering it from USA?
on more, did anyone buy a modem with intel chipset 537 or md3200 and where
(in italy)?
Thanks
__
You can buy them from Telappliant in the UK. They take credit cards so within the EU
there are no customs issues.
http://www.telappliant.co.uk
OEM cards are around... http://www.goods2world.com/product_info.php?products_id=55 for
about £15 each. They seem to be identical to the Digium cards.
Title: Message
Yes, I
have seen this as well but I haven;t quite understood why. I am keeping an
eye on it and wil ltry and get some traces...
Rgds
Tim
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shad
MortazaviSent: 21 July 2004
hi!
How do I play background music for the caller and callee while they are in
the coversation.
A caller comes to Asterisk box and then do dtmf input for the second callers
number
then the box dials the second caller
Hence they are bridged.
I need both of them to listen to some music while they
hi!
What's the libr2 status for Asterisk ? I've got R2 E1 delivered to my * box.
I have TE410P digium quad card with newest CVS.
How much % is completed with libr2 ?
what's completed ?
What's missing ?
Thanks,
bit123.
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Asterisk-Users mailing list
Title: Asterisk RC1 and bristuff
Hello,
Is the bristuff from junghanns.net are implemented in the asterisk RC1 release?
If no, is there a new patch from Junghanns in order the quadBRI card works?
Thanks by advance.
GIBERT Frédéric
Mobile: +33 6 72 08 35 16
Fax : +33 1 30 71 39 33
Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ
labs
softphone, i have the most recent Asterisk version, but when connecting to the PSTN i
have
choppy voice problems, not internally just when connecting with my Mediatrix gateway
and
ATA, my SJLabs
Is there an application I could use to test this? I.E. like the echo
test, but doesn't send anything back...
app_record.so ?
If you want to test towards the telco's central office, find out what
their quiet terminiation number is. Just about every central office
has a piece of equipment
Hi,
I'm using asterisk as softphone for a certain application.
It uses chan_capi for PSDN connection and chan_oss and
the manager as user interface.
When calling someone, who is busy, I can hear at the
speaker the busy indication, but the manager command
Status still tells Ringing (chan_oss) or
I had the same problem with a Mediatrix, it turned out to be a defective
unit. No matter what we did the audio was very choppy, when I replaced the
unit my problems went away.
Are you running it as SIP or MGCP?
Norm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hello all,
I need to write a queue_log parser that is going to implement more or less
the functionalities described here
http://lists.digium.com/pipermail/asterisk-users/2003-July/014965.html
of course not everything from scratch, but this is where I'd like it to go.
I am looking for
-
I have both cards and they look the same to me.
The only thing I would pass on is that the card has a fixed impedance of
600 ohms and thus you will probably have echo issues .
Chris
On Wed, 2004-07-21 at 11:39, Robinson Tim-W10277 wrote:
You can buy them from Telappliant in the UK. They take
On Tue, 20 Jul 2004 23:50:05 +0200, Andy Powell
[EMAIL PROTECTED] wrote:
Hi,
I've just (earlier today) updated from CVS so that I can apply the dtmf caller id
patches. Unfortunately this has had an undesired effect.
I'm using * with an IX66 and no issues, with CVS head I suggest you
have a
I've recently enabled monitoring (recording) of incoming calls that
arrive in the queue (all calls come in through the queue) using the
config options in queues.conf.
However, it seems that as soon as the call is placed on
hold/transferred, the monitoring stops. I would like to know if it is
Title: Caller based routing
Hello,
Can someone explain me how to do caller based routing.
Here is my example.
I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture.
My problem is when I want to place
I'm having an
interesting problem with a Cisco 7960 phone, and two Asterisk servers. I'm not
sure if this problem is specific to the 7960, or even to Asterisk for that
matter.
Here's the scenario.
I have an * server at one location with a public IP address (i.e. not behing
NAT). I have a
Hi,
I have some issues with communication between to * servers. They are
connected over DSL (3Mbps). One is behind NAT and the other on routable
network. Almost every time caller will hear the other end like fast forward
while the other end will have perfect quality. It doesn't matter if we use
Hello,
I am receiving the following repeated Errors and Warnings with
Galaxyvoice. I have placed the sip context below, perhaps someone can
offer suggestions how I could troubleshoot this. Thanks
Kevin
Jul 20 12:35:48 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum
retries exceeded
Hi
Just create a new context, and use ex girlfreind logic.
cheers
Clive
On Wed, 21 Jul 2004 14:58:17 +0200
GIBERT Frédéric [EMAIL PROTECTED] wrote:
Hello,
Can someone explain me how to do caller based routing.
Here is my example.
I have an asterisk between a PBX and the PSTN. The
bit123 wrote:
hi!
What's the libr2 status for Asterisk ? I've got R2 E1 delivered to my * box.
I have TE410P digium quad card with newest CVS.
How much % is completed with libr2 ?
what's completed ?
What's missing ?
Thanks,
bit123.
libr2 gives you about 5% of a very bad R2 implementation. I
In your dialplan for your voip routing you'd put a gotoif that jumped to
your PSTN context if it matched your criteria (e.g. EXTEN = faxextension)
Steve
-Original Message-
From: GIBERT Frédéric
To: [EMAIL PROTECTED]
Sent: 21/07/04 13:58
Subject: [Asterisk-Users] Caller based routing
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
i've installed asterisk by last cvs and i note
res_parking.c
is not anymore there; chan_capi-0.3.4b INSTALL file require:
in /etc/asterisk/modules.conf insert the line:
load = res_parking.so
load = chan_capi.so
running asterisk i get:
James H. Cloos Jr. wrote:
The demand exists; is anyone up for spulying that demand?
Interesting conversation... a partner and I are setting up _exactly_
this sort of business right now, but not in the areas the OP wanted.
I see a great deal of market for VOIP trunk service exactly as mentioned
Andrew Kohlsmith wrote:
On Tuesday 20 July 2004 18:18, George Pajari wrote:
In spite of what my learned colleague implies above, there is more to
Canada than Ontario (Bell's territory).
Please retract your statement that I implied anything of the sort; I never
even mentioned the province I was
What markets are you targeting? Do you have any pricing yet?
/carmi
On 21 Jul, 2004, at 9:51, Kevin P. Fleming wrote:
James H. Cloos Jr. wrote:
The demand exists; is anyone up for spulying that demand?
Interesting conversation... a partner and I are setting up _exactly_
this sort of business
On Wednesday 21 July 2004 09:51, Stephen R. Besch wrote:
Some people just have bristly whiskers!
I'm not looking for a kiss from the man... :-)
-A.
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[EMAIL PROTECTED]
Carmi Weinzweig wrote:
What markets are you targeting? Do you have any pricing yet?
Initially we will be a small player, serving only the Phoenix
metropolitan area (Phoenix, Scottsdale, Tempe, Mesa, Glendale, Peoria,
etc.) We are using services from a CLEC with presence in a large number
of
Doesn't it go the other way 'round?
Smaller companies = more lines/employee;
Larger Companies = fewer lines/employee
?
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 21, 2004 8:52 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] DID
Jay Milk wrote:
Doesn't it go the other way 'round?
Smaller companies = more lines/employee;
Larger Companies = fewer lines/employee
Well, the crossover point is pretty low; we are seeing small companies
(6-8 employees) with four lines but only one or two are in use 95% of
the time. They have
On Tue, 2004-07-20 at 16:56, Steve wrote:
I think the above is related to the Grandstream going bad. A few times when I
power it up it does not boot all the way. Now it did not even accept key
presses in VM, though it did accept the VM button...
I've talked to Grandstream engineers and they
Hi,
I'm setting up multiple asterisk servers and trying to do the classic
DIAL(IAX2/asterisk1/${EXTEN}IAX2/asterisk2/${EXTEN}IAX2/asterisk3/${EXTEN},15)
After googling a bit, I fell on a discussion about putting this in a
variable so that adding additionnal servers would be easy. I can't seem
I'm using a Cologne chip card in my Asterisk box with zapHFC drivers
(bristuff-0.0.2). The system works well, but this way I'm not able to run
newer version of Asterisk.
Do you think it's better to use i4l support and newer version of Asterisk or
keep the bristuff with older asterisk ??
Have
Nobody? Yes? No? Maybe?
- Original Message -
From: Chris Shaw [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 20, 2004 5:54 PM
Subject: [Asterisk-Users] Latest CVS (7/20/2004) stops answering SIP calls
after 5 min
CVS 7/16/04 (the latest one I have b4 today) seems to have
Hello,
I have a one-way audio problem. If any one can give me a clue on how to
solve it, I'd highly appreciate.
My configuration is:
Both Asterisk server and a SIP phone run within a LAN. Asterisk:
CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp
14262. The Linux box
On Wed, 2004-07-21 at 16:55, Massimo De Nadal wrote:
I'm using a Cologne chip card in my Asterisk box with zapHFC drivers
(bristuff-0.0.2). The system works well, but this way I'm not able to run
newer version of Asterisk.
Do you think it's better to use i4l support and newer version of
I am on many mailing lists and lots of them have similar problems with people posting
messages they
could better answer themselves.
Since many of these messages are from people posting for the first time,
I think to some degree this is a failing of the mailing list structure itself.
I've
Hi,
I know that this issue has been discused guite a lot, but I haven't managed
to get a definite answer. Is those two values supposed to be floats (e.g.
3.5) or integers with the percent symbol (e.g. 20%)?
Thanks,
Yiannis.
___
Asterisk-Users
going to i4l means... incoming sound sometimes gets interpreted as DTMF
- and when your caller humms a '#' - transfer kicks in... Outgoing DTMF
mhhh almost unuseful but surely funny ;-)
There is an Update patch for bristuff... look carefully in the download
directory.
do you mean
Hi all
Two S100I-IAXY configured * the CVS-HEAD and following the IAXY´s
Configuration Guide v. 1.0 by Digium.
The first one S100I-IAXY have IP 10.0.0.5. (my home)
The second S100I-IAXY have IP 200.253.232.23. (my office)
I only obtain to establish a linking enters the two S100I-IAXY when I
Title: Mensaje
I have the same problem.I'm usinr
asterisk-1.0-RC1. Anyone could help us?
regards,
srsergio
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Dan
FernandezEnviado el: viernes, 16 de julio de 2004
20:42Para: [EMAIL PROTECTED]Asunto:
James H. Thompson wrote:
(B
(BI've wondered if a mechanism like this would help:
(BFor the first N messages you post to the mailing list,
(Byour post does not automatically get posted.
(BInstead you get a message similar to Olle's below,
(Bending with something like:
(B
(B "If you
Title: Nachricht
Hi Frédéric,
If
no, is there a new patch from Junghanns in order the
quadBRI card works
yes,there is a new one from
Junghanns.I use itsince last weekend without
a
problem.
http://www.junghanns.net/asterisk/downloads/bri-stuff-0.1.0-RC1.tar.gz
Karl
What would be the bare minimum hardware and software requirements to
run asterisk in it's full glory with agi support to handle 1 fxo, 1
fxs, and sip off to a provider such as voicepulse.
Eric
--
They that would give up essential liberty for temporary safety deserve
neither liberty nor safety.
Gonzalo Gasca wrote:
Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ
labs softphone, i have the most recent Asterisk version, but when connecting to the
PSTN i have choppy voice problems, not internally just when connecting with my
Mediatrix gateway and ATA,
Hia
i try to use the fonction Getvar of asterisk to get a variable myDNIS
that i have define. i use it as follow
Action: Getvar
Channel: SIP...
Variable: myDNIS
but asterisk don't know it .i have the response as follow
Response: Error
Message: Invalid/unknown command
does everybody meet
Hello,
We just had some new T1s turned up today to replace others that our contract
has run out on and we are now getting more echo on the new T1 lines than we
had on the old ones.
The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they
replaced)
The problem is that we are
sorry, I misread your post.
check from asterisk console:
show manager commands
if the function getvar is registered.
here with rc1 works without probs.
Matteo.
Il mer, 2004-07-21 alle 19:13, Brancaleoni Matteo ha scritto:
dialplan apps are not manager apps
matteo.
Il mer, 2004-07-21
On Wed, 21 Jul 2004, Michael Wang wrote:
How do I change configuration of Asterisk so that phone B can use
aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address?
sounds like * is using reinvite to get itself out of the loop and let the
phones send RTP directly between
Hi everyone,
I have a TDM400 configured with 4 FXS ports, each connected to a
caller-id analog trunk port on a Nortel system. Outgoing calls work
great. But on incoming calls it appears that loop current is getting
dropped momentarily about 10 seconds after the call is answered. Since
the
Alejandro
Why can't you use IAX? I'd love to test your
termination.
Saludos
Daniel
- Original Message -
From:
Alejandro Sosa
To: [EMAIL PROTECTED]
Sent: Tuesday, July 20, 2004 2:54
PM
Subject: [Asterisk-Users] FREE (305) and
(786) termination. Anyone
Hi
I have a TDM400 configured with 4 FXS ports, each connected to a
caller-id analog trunk port on a Nortel system. Outgoing calls work
great. But on incoming calls it appears that loop current is getting
dropped momentarily about 10 seconds after the call is answered. Since
the Nortel
I don't use T1's, only regular lines, but echotraining works with any
zaptel interface as far as I know.
I would try echotraining=yes and echotraining=800 (if your using a
relatively new CVS version).
I personally haven't noticed any pause when using echotraining, I
think its less then 1 second,
Hi, Does anyone know if there is a E1 pci card that can work with asterisk and support modified R2? Is this functionality of the card or the libpri driver ? Regards.Marcelo RodriguezIxNetworks
Hi I am kindda new to this mailing list. I have buit asterisk alrealdy once,
but this time I am having a hard time to build it. Does anyone have
anysuggestion why am I getting so many errors.
Thanks
Felippe Kilian Martins
_
MSN
Hi
Il mer, 2004-07-21 alle 19:37, Marcelo Rodriguez ha scritto:
Hi,
Does anyone know if there is a E1 pci card that can work with
asterisk and support modified R2? Is this functionality of the card or
the libpri driver ?
the protocol (isdn,r2,whatever) is in userspace.
isdnco is in
Hi,
Il mer, 2004-07-21 alle 20:19, Felippe Martins ha scritto:
Hi I am kindda new to this mailing list. I have buit asterisk alrealdy once,
but this time I am having a hard time to build it. Does anyone have
anysuggestion why am I getting so many errors.
unfortunately, this list doesn't
Putting on Tin Foil Hat to pickup brain waves
Let's see here, from the information I'm receiving from my Brain Wave
Reader, it would seem that you aren't emitting enough activity for me to
determine much of anything.
I would suggest posting some of the errors you're getting.
On Wed, 2004-07-21
Hello,
Sorry, it's near-end echo
Also, I am running Slackware 10.0 with Asterisk CVS from 2004-07-06 on a P4
with a TE405P quad T1 card.
Thanks,
MATT---
-Original Message-
From: Mike Benoit [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 21, 2004 2:06 PM
To: [EMAIL PROTECTED]
Title: Asterisk Server gives 403 forbidden
Hi
I am a new Asterisk user, I am trying to make a call between 2 Windows messenger clients.
At present I am trying to get one client to register with the Asterisk Server. I get a 403 forbidden
Could anyone tell me what I am doing wrong?
A
On Wed, 2004-07-21 at 11:48, Yiannis Costopoulos wrote:
Hi,
I know that this issue has been discused guite a lot, but I haven't managed
to get a definite answer. Is those two values supposed to be floats (e.g.
3.5) or integers with the percent symbol (e.g. 20%)?
It's on the Wiki:
Title: SIP Hard Disconnect Detection
Hello. I have a question regarding Asterisk internal API.
I am developing a new asterisk module application using asterisk internal c API. I am having problem detecting hard hangups when the SIP clients disconnect (suppose power goes off in the phones). I
Title: Message
I installed the
QuadBri card in my * server.
I'minstalling*on a RedHat 9
server
I run the install.sh
file. So far no problems.
If I try to start
/sbin/ztcfg -v -c /etc/zaptel.conf
I will see the
following error:
Zaptel
Configuration==
SPAN 1: CCS/
Hi.
I'm from Brazil, and I have some problems due the instalation of zaptel.
Using RH9, kernel 2.4.20-8.
I don't understand the error and i need help.
While the compilation of zaptel 1.0, this return many errors and warnings. The
errors is listed below:
# make
gcc -I/usr/src/linux-2.4/include
I'm having a problem with some customers sitting behind hopefully SIP aware routers
doing NAT. These routers translate port 5060 to something different (ie. 10001) in
order to be able to connect more than one SIP client on a single NATted LAN.
Unfortunately, after a while the router seems to
I currently have a Toshiba Strata DK424 with a Stratagy
voicemail system (4 ports). I am looking to go from having a receptionist
answering the phone to an automated attendant. It appears that Asterisk
can be the solution, but I have some questions. Do I just replace the
Stratagy with the
ok thanks
I checked it and effectively i don't
have function getvar in the list. How can i do to get it ? is there something to
install ??
i try a cvs update but no
changes.
thanks in advance
sorry, I misread your post.check from
asterisk console:show manager commandsif the function
On Wed, 21 Jul 2004, Preeti Gopalan wrote:
[EMAIL PROTECTED]
type=user ; either friend (peer+user), peer or
user
context=default
[EMAIL PROTECTED]; usually matches the
section title
host=172.16.4.79 ; we have a static but private IP
Hello List,
I'm from Germany and I want to use a Asterisk
System.
I have a few Accounts at my SIP-Provider www.sipgate.de and now I want to use my
ISDN-Phone on the Sip-System.
My idea was i set up a Asterisk-System and i will
put in an ISDN Card where I can plug a ISDN Phone, I will have
Hello everyone,
I playing around with ENUM and have configured * to query a
few sources for testing purposes (fierymoon, e164.arpa, e164.org). Id
like to know if there is a way to query these servers manually (ie outside of
asterisk via nslookup or equivalent) to find out if particular
I have a P3-800 with two IDE drives in a software RAID1 configuration.
Each drive is on a separate IDE channel. Now anytime there is HD
activity, I hear beeps and cutting out on a call using the X100P
card.
I ran the zttest program, and discovered HD activity would drop the
accuracy down to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I'm experiencing frequent kernel panics when using the X100P card under
the 2.6.6 Fedora kernel. I've attached the kernel output to this message
- - it looks like the IRQ stack is overflowing and trashing some memory,
causing a series of oopses
Hi guys,
Anyone running go2call setup ? can anyone send me the configuratio sip.conf
lines ?
I am planning on using asterisk with a linejack and phonejack. I am not sure
if this will work. These cards use g729 and g723.1.
I also have some x100p and tdm cards from digium but without the codecs. I
I think your kernel module isn't loaded for your card.
Once those get loaded, the stuff in /dev gets created.
Look in /lib/modules/kernel number/misc for the kernel modules
Do a 'demod -a' first and then you can do a blanket modprobe like this:
modprobe \*
It'll pretty much load all your
It would help if you included a brief description of the errors you're
getting.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Felippe Martins
Sent: 21 July 2004 19:20
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Building Asterisk
Hi I am kindda new
I fixed this error on mine by creating a symbolic link in /usr/src with:
ln -s linux-2.4.21-15.0.3.EL linux-2.4
of course using your particular flavor of redhat kernel instead of
linux-2.4.21-15.0.3.EL.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi
(B
(BIf you have installed Asterisk on your Mac using our
(Binstall package downloaded before Tue July 20th 2004 9am
(BGMT, then your installation may be incomplete, as
(Bpreviously discussed on the list.
(B
(BI have just uploaded a patch which will install any
(Bmissing files.
(B
Hi Michael-
You might try reading up a little in the user-maintained web site, called
the Wiki, and then post more specific questions:
http://www.voip-info.org/tiki-index.php?page=Asterisk
Hope this gets you started - sounds like asterisk will work well for you.
It's much less expensive than
On Wed, 2004-07-21 at 13:50, Ricardo Maia Martins dos Santos wrote:
Hi.
Just because it is Urgent to you doesn't make it urgent to anyone else.
Our help is voluntary. If you want urgent care, call a consultant. You
may encounter more hostility next time you invoke urgent without a
check in hand.
Is your computer turned on? If not, turn it on and try building
Asterisk again... Otherwise, it could be any number of things.
-Original Message-
From: Felippe Martins [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 21, 2004 1:20 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 21 July 2004 02:45 pm, Preeti Gopalan wrote:
Hi
I am a new Asterisk user, I am trying to make a call between 2 Windows
messenger clients.
At present I am trying to get one client to register with the Asterisk
Server. I get a 403
Hello-
First, it sounds like asterisk can do what you want to do. You have a
number of requirements, though. I think its too much to expect people on
here to design your application for you for free. Perhaps you might hire a
consultant for a few hours to help you out (see the asterisk Wiki for
On Tue, 2004-07-20 at 13:22, Chris A. Icide wrote:
On 10:41 AM 7/20/2004, Carmi Weinzweig wrote:
I want many phone numbers so that each phone in my facility has its own
phone number, but I really do not need that many simultaneous calls and
it would be cost prohibitive to pay several
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