unless you show us some config files, I doubt that anybody can help you...
On Wednesday 14 September 2005 16:46, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example 100
102
Hi, I m trying to install [EMAIL PROTECTED] after installing and logging in as root password i made network connections using netconfig command there i gave ip address as provided by my network provider it displays the ip address I m SORRY to ask that how can i access the net GUI if u can
Flobi wrote:
I've been messing with it for a couple weeks with MySQL. It seems
pretty good to me though I have had a couple crashes. I cane' say for
sure that the crashes were directly related to RealTime though. Also,
I'm still using CVS HEAD 2005-09-06 which was right before the beta
RTFM
prashant yadav wrote:
Hi, I m trying to install [EMAIL PROTECTED] after installing and logging in
as root password i made network connections using netconfig command
there i gave ip address as provided by my network provider it displays
the ip address I m SORRY to ask that how can i
Hi,
I am Newton from Liguetel in Brazil.
I have now a billing system based on SQLPostgress which is able to collect
real time CDRs and present in a web site all the accounts and CDRs related
to their calls.
This billing is also able to set accounts balance and for each call
balance goes down as
I use BINK to burn ISO Images and it works great.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, August 30, 2005 11:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] (no subject)
On Tue
having problems with installing [EMAIL PROTECTED] i downloaded the
asteriskathome-1.5.iso file from asteriskathome.sourceforge.net link burned it on a cd but it is not booting what seems to be the problem hoping for a quick reply
___
--Bandwidth
Sounds to me like you copied the file to a disk rather than burn an ISO
image. A common mistake folks make especially if they've never done an
iso before.
What tools are you using? I prefer k3b. It rocks
Mark
prashant yadav wrote:
having problems with installing [EMAIL PROTECTED] i
On Tue, Aug 30, 2005 at 05:27:37PM -0400, Mark Phillips wrote:
Sounds to me like you copied the file to a disk rather than burn an ISO
image. A common mistake folks make especially if they've never done an
iso before.
But then also wrote:
What tools are you using? I prefer k3b. It
Depends on the phon you are using. Park will do that, and you should use park.
On 8/28/05, bodra [EMAIL PROTECTED] wrote:
Hi all
i am developing a client for the asterisk that controls ur phone from an Xp
c# application
what functions in Asterisk that will allow you to put someone on
Hi all
i am developing a client for the asterisk that controls ur phone from an Xp c#
application
what functions in Asterisk that will allow you to put someone on hold but not
park calls and bring them back, without using flash hook cuz it will be a
button in that application
Powered by
Hi
I am getting this error after installing and configuration
of asterisk.
Aug 24 17:53:50 WARNING[9924]:
channel.c:472 ast_channel_walk_locked: Avoided initial deadlock for 'Zap/3-1', 10
retries!
I have upgraded asterisk to latest version but still
receiving the same error. Can
Hi:
I was running TDM12B. Both FXS and FXO were working
fine. Then all of the sudden FXS had problems. When I
pick-up the phone and dial any number, FXS doesn't
respond. I just keep hearing the normal signaling line
tone comming from the FXS. I changed the FXS module
and it had the same problem.
I am using the correct version of pwlib(1.5.2) and
openh323(1.12.2) for the stable asterisk build. Both packages configure and
compile with no problems. However
when compiling chan_h323 from the asterisksource/channels/h323 directory I get
this error.
Chan-h323.h:31: warning;
On Monday 15 Aug 2005 15:19, Tom Tobias wrote:
I am using the correct version of pwlib(1.5.2) and openh323(1.12.2) for the
stable asterisk build. Both packages configure and compile with no
problems. However when compiling chan_h323 from the
asterisksource/channels/h323 directory I get this
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
My linux box speaks pppoe to external DSL modem.
Nortel NTEX35 BAAB. It's up 24/7 and provides
web service...etc.
Has 6 nics, one of them is fiber.
Asterisk is on the same box.
Don't have any IP phones yet.
The asterisk default is to listen on all 6
enet interfaces? (this is what I'd want).
On Sat, Aug 13, 2005 at 08:10:03AM -0800, Cliff Savage wrote:
The digium board will be in the same box.
Does this mean:
Channel 4 to incoming phone line.
Channel 1 to DSL modem?
Or DSL modem to the incoming line...and then the pass thru
port on the DSL modem goes to Channel 4?
Will
TC ADSL should not bother PSTN as long as you use a proper filter. In our
TC case a proper filter was supplied by the phone company when we installed
TC the ADSL line. We Happily use Asterisk with an FXO card and an ADSL
TC connection from the same phone line.
Got a leviton DSL filter mounted
Im trying to get Asterisk to accept incoming
calls from budgetphone.nl.
When I dial my budgetphone nr on a PSTN KPN line it
immediately gives a busy tone.
I tried X-lite, which worked perfect, so my modem
(with nat) probably is not the problem.
I did a sip debug and got the following
Hello,
I am having some problems with faxing in asterisk. I have a TE100P
which is taking my PRI. This seems to be working fine. I also have a
TDM400P with 2 FXS. Again card seems to be working fine, I can dial
from phones attached to these to ports and everything seems to work
fine. I have
I am having some problems with faxing in asterisk. I have a TE100P
which is taking my PRI. This seems to be working fine. I also have a
TDM400P with 2 FXS. Again card seems to be working fine, I can dial
from phones attached to these to ports and everything seems to work
fine. I have 2
Hi, I
installed mpg123 v0.59r without error and defined as defaut folder
/var/lib/asterisk/mohmp3. When i set a call on hold everythinghs seem ok, but i cannot hear music. I'm using asterisk
1.0.8
*CLI -- Executing Dial("SIP/2339-4da6",
"SIP/2391|60|Thtr") in new stack -- Called
2391 --
can you help me to configure lcs2005 with asterisk...
I use SER to resolve the problem that there is for communication protocol...LCS
uses tcp, Asterik UDP.
Someone, knows how to do the configuration beetwen LCS and SER , SER and
Asterisk? the function of asterisk is SIP-PSTN Gateway for the
Hi
I have a PHP agi-bin scripted called callhander.php and its setup to
answer anything that comes into the PBX,
In the script I am trying to the get the system to play a file called home
which I know works, as I can get the Play function to work from the
extensions.conf file. However within
Rich,
What about a combination of your excellent/intelligent suggestion
something like this:
exten = 911,1,Dial(Zap/g17/${EXTEN})
exten = 911,2,SoftHangup(Zap/1-1)
exten = 911,3,Wait(1)
exten = 911,4,Goto(1)
... with the idea that if a line is not free, we forcible seize one.
Probably not
BJ,
BJ Weschke [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom
SIP termination vs. DS3
To: Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
Did I miss pricing
PROTECTED] wrote:
BJ,
BJ Weschke [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom
SIP termination vs. DS3
To: Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859
Hi friends !
Cvan anybody help me to configure asterisk with ser so that I can share the
load of the asterisk with ser server. I have tried it but my asterisk is not
showing registrations of the useragent, as given in the asterisk
wiki/asterisk+at+large. I don`t know what is the problem, but
hello,i'm a naw asterisk user i've configured my 2 sip phones and they can place calls .i,ve also fxo card and i've configured channel ;now it's possible to recieve analog calls with my sip phone but i want to make call with my sip phone to analog it's possible?
when i dial a number my sip phone
S.NASROLLAHI
hi
i am a new member
i want to learn what is TOS and LOG command in the access list and
what are they doing?
what is their advantage ?
when i should use them?
thank u
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Does anyone know what the [WARNING: . Changethread: Can't change device
'**Unknown**'] line means below..
I just set verbosity to level 5, and noticed that error everytime a
voicemail is left.. Everything seems to work ok, and I have no idea how long
that error has been there, but I'm just
On 4/27/05, Andre Normandin [EMAIL PROTECTED] wrote:
Does anyone know what the [WARNING: . Changethread: Can't change device
'**Unknown**'] line means below..
I just set verbosity to level 5, and noticed that error everytime a
voicemail is left.. Everything seems to work ok, and I have
Funny, they sell these old cards.. it seems like they are selling refurbs
as new.. ... anyways RMA is on its way, would be nice if they would send
one as a replacement first, so that we could continue our work and don't
have to delay it.
They can, its called cross-shipment, but
On Tue, 12 Apr 2005, Rich Adamson wrote:
Funny, they sell these old cards.. it seems like they are selling refurbs
as new.. ... anyways RMA is on its way, would be nice if they would send
one as a replacement first, so that we could continue our work and don't
have to delay it.
They
Hi,
I have just bought another TDM400P card from Digium directly, purchased
last Thursday, received it today:
Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1
1 WCTDM/0/0 FXOKS (In use)
2 WCTDM/0/1 FXOKS (In use)
3 WCTDM/0/2 FXSKS (In use)
4 WCTDM/0/3
The only firmware upgrade procedure is for you to call digium support.
Hi,
I have just bought another TDM400P card from Digium directly, purchased
last Thursday, received it today:
Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1
1 WCTDM/0/0 FXOKS (In
Funny, they sell these old cards.. it seems like they are selling refurbs
as new.. ... anyways RMA is on its way, would be nice if they would send
one as a replacement first, so that we could continue our work and don't
have to delay it.
On Tue, 12 Apr 2005, Rich Adamson wrote:
The only
Funny, they sell these old cards.. it seems like they are selling refurbs
as new.. ... anyways RMA is on its way, would be nice if they would send
one as a replacement first, so that we could continue our work and don't
have to delay it.
They can, its called cross-shipment, but they need a
We'd be honest and sell them as last years models.
Jon
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 12, 2005 10:25 AM
Subject: Re: [Asterisk-Users] (no subject
Good morning all..
I was following a discussion on this list about the
TDM400P revisions. It is my understanding that the current
revision that one should have is the Rev. H and not the
E/F. I have not yet been able to verify the rev stamped on
the board, but zaptel is reporting that I have
I was following a discussion on this list about the
TDM400P revisions. It is my understanding that the current
revision that one should have is the Rev. H and not the
E/F. I have not yet been able to verify the rev stamped on
the board, but zaptel is reporting that I have the Rev.
E/F.
Hi
Repeated e-mail as I forgot to make plain text - sorry. Newbie asterisk guy
here and forgive this slightly long mail, but I'm stuck on this for a week.
I'm having major problems getting a Fritz card to dial out in the UK (or indeed
answer, but I've been concentrating on dialing out). I'm
Hi,
I want to know if VAD is possible in Zaptel Analog lines
Reg,
parijat
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
_
Emotikony a pozadi programu MSN Messenger ozivi vasi konverzaci.
http://messenger.msn.cz/
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi!
If I want to use ISDN card for connecting phones to it, that card must be HFC-S,
because of NT mode.
How about if I am connecting ISDN card to the external ISDN phone line (to local
telephone companys s-bus) when card must be in TE mode, do I still have to have
HFC-s card that I could
I don't know what's means about register in sip.conf
such as:
register = user:secret:[EMAIL PROTECTED]:port/extension
even if I registe a sip proxy , but how use it ?
I think :
when incoming from sip proxy to asterisk :
user a -- sip proxy -- asterisk -- pstn
sip proxy : SER
in ser.cfg
Wow.. When I looked at this, I assumed someone had been kidnapped.
Maybe Thunderbird just made it look odd.. Anyway, you should look on
www.voip-info.org for plenty of info on how to use sip. If you want a
sip proxy to play with, set up an account on free world dialup.
VOIP-INFO.org gives
The problem was this line at the end of modules.conf
alias wcfxs wctdm
Ismael.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hello all,
I have an asterisk 1.0.3 stable instaled on a box.
All works fine with this machine, but the only problem i get is that
suddenly the machine hangs up all the establised calls and we have to
call again.
This problem occurs twice a day and i don not know how to debug it.
I read
FYI, I didn't read your message. With hundreds of messages/day, I use
the subject line to decide whether or not to read. Whenever I get a
message with (no subject) it is an instant delete.
Also, for those of you who think you're still on a 300baud modem and
have to conserve every keystroke,
I have a question
for using gastman. I have set up extensions for my IAX users as
IAX2/username, and I keep getting the following
Dunno how to tell if
IAX2/username/6 is IAX2/username
I was wondering if
there is some sort of wildcard character that can be used here? The number
changes
Since you are the only user that uses this list, I'm very happy to see
that you didn't include a subject in the message, that way it saved
some electrons. Or did it?
On Mon, 14 Feb 2005 11:49:07 -0600, Ron Frederick [EMAIL PROTECTED] wrote:
I have a question for using gastman. I have set up
I want to attach two SIP phones to an asterisk server. The SIP phone
I'm using is the software xlite for windows. The university LAN has a
firewall and xlite says that a blocked firewall was detected. How can
I get around this because they are not going to unblock UDP 5060 for
me.
Riz
Hi,
I try to use Manager API to originate a call from a channel to an
existing extension. Based on sip channel show command, the Manager
initiates a call to the channel only. It doesn't generate a call to
the extension. So the originate call API of Manager is failed. I think
I pretty much
Pat Delaney wrote:
Thanks for you comments. I have the one port card now. I plan on
purchasing the TDM400. My only question is whether or not the Dell
optiplex has pci 2.1 (I think)
Depends on the model, check Dells website. A quick googling show:
Specifications: *OptiPlex* GXi. *...*
AS a proof of concept experiment,
I want to try and integrate Asterisk with my Lucent Definity G3 switch. I dont
have an available T1 port on the G3 but I can round up 4 analog ports off the
G3. What I thought I could do is create a Hunt group on the G3. Lets say I
configure 5610 5613 as a
Pat Delaney wrote:
AS a proof of concept experiment, I want to try and integrate Asterisk
with my Lucent Definity G3 switch. I dont have an available T1 port
on the G3 but I can round up 4 analog ports off the G3. What I thought
I could do is create a Hunt group on the G3. Lets say I configure
Hello,
Asterisk provides its own Asterisk gatekeeper is there
other wise it supprots gnugk
please tell me
Thank u
Sailatha[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
HELP!
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw
register =
[EMAIL PROTECTED]:X:[EMAIL PROTECTED]/1234
[sip.broadvoice.com]
Hello
I setup Mediatrix 1124, I am able to make incoming call but unable to make
outgoing call. When ever I tried it just gave me a beep sound.
I appreciate any help on this.
Thanks
Deepak Malhotra
This message was sent
Hello,
I have X101P card.
But it seems to be dead. Always
app_dial.c:803
dial_exec: Unable to create channel of type 'Zap' (cause 0)
I've add the
line:exten = 999,1,Dial(Zap/1). But calling to 999 show the same
error.Zap show channel, lspci etc show everything is normal.
Could you tell
test
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Sorry to ask about this again, but Im trying
to get my head around a few things.
If a call is redirected from a PRI (quad e1) back out
into the PRI (same quad e1) what processes does asterisk undertake on the call?
Were getting terrible quality issues (and only with redirected calls).
So the only issue left I have is with this skinny not found when
0.0.0.0
is set in skinny.conf
in modules.conf
noload=chan_skinny.so
Oops
noload = chan_skinny.so
what's this skinny anyway?
Cisco's VoIP protocol, like SIP, or MGCP, but Cisco developed it
themselves, and it is the default
Hey All,
I would like to start moving all my .conf files into a database backend.
Before I go reinventing the wheel, I wanted to check on what is being worked
on. I don't think the Wiki has been updated on this subject (which I will be
happy to do one I am armed with the information)
Any tips,
There are some new checkins to the CVS that allow extensions to be done
via ODBC. IAX peers and voicemail are done like this already.
I am in the process of writing informal documentation for myself and my
developers. If you'd like we can collaborate off-list.
On Tue, 2004-11-30 at 15:39
What is the general consensus on the Polycom SIP Phones?
I am getting random gargled up sounds on mine and I really do think it is the Polycom
Regards,
Michael DiMartino
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Is there any problem with the compilation of channel h323 in asterisk 1.0.2?
I get the following error.
/asterisk-1.0.2/channels/h323# make
g++ -g -c -fno-rtti -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -DP_LINUX
Michael Welter [EMAIL PROTECTED],
If you would like personal answers from me you'd better stop using that
braindead blackholes.us that seems to blackhole the whole of Asia.
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hello,
I'm testing * with two XLite softphones, they are all running within a local
network. I was able to install * and register the phones to it. But when I
dial the other extension I get this messages:
Connected to Asterisk CVS-HEAD-10/25/04-23:24:03 currently running on
localhost (pid =
I've been working on the asterisk manager for a few days, today I
started on a little prototype click to talk on the web via php thing. I
have it working properly except for one thing. I do a SetVar:
CTTN=somenumber over the socket and when the call gets to the socket
its empty. Is this
Thank you for your reply.
I forgot to mention ... Asterisk dies with that error message ...
Everything goes ok with download/compile but when I want to run
Asterisk it dies.
Message: 7
Date: Sun, 10 Oct 2004 21:14:53 -0500
From: Brian West [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] newbie
Hello,
I´m trying to compile the Fritz CAPI
module for Debian, following the steps related in
http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install
But I always get the same error,
debian-asterisk:/home/ismaelg/fritz#
make
(cd src.drv; make CARD=fcpci)
Steve Maroney wrote:
Hey guys,
Im about to sign up for VoicePulse Connect. Of course, I plan on using my
asterisk server to register = with the service. I would rather sign up
with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is
behind another Linux box serving as my
hi all,
can anyone give solution for this.
wct1xxp - Digium Wildcard T100P T1/PRI Card 0
zttool gives
RED Digium Wildcard T100P T1/PRI Card 0
my zaptel.conf look like this
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone=us
defaultzone=us
the above 5 lines only
On Mon, 2004-09-13 at 05:39, Murali wrote:
hi all,
can anyone give solution for this.
wct1xxp - Digium Wildcard T100P T1/PRI Card 0
zttool gives
RED Digium Wildcard T100P T1/PRI Card 0
my zaptel.conf look like this
span=1,1,0,esf,b8zs
bchan=1-23
Hey guys,
Im about to sign up for VoicePulse Connect. Of course, I plan on using my
asterisk server to register = with the service. I would rather sign up
with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is
behind another Linux box serving as my nat/firewall. Does anybody
On Sun, 12 Sep 2004 13:35:14 -0500 (CDT), Steve Maroney
[EMAIL PROTECTED] wrote:
Hey guys,
Im about to sign up for VoicePulse Connect. Of course, I plan on using my
asterisk server to register = with the service. I would rather sign up
with VoicePulse via SIP instead of VoicePulse Connect.
First off, sorry about the missing subject.
VoicePulse Connect and VoicePulse seem to be two different companies.
It doesn't seem that VoicePulse offers IAX connectivity, Just SIP.
VoicePulse offers more packages than VoicePulse Connect.
Thank you,
Steve Maroney
On Sun, 12 Sep 2004, Marconi
On Sun, 12 Sep 2004 14:28:20 -0500 (CDT), Steve Maroney
[EMAIL PROTECTED] wrote:
First off, sorry about the missing subject.
VoicePulse Connect and VoicePulse seem to be two different companies.
It doesn't seem that VoicePulse offers IAX connectivity, Just SIP.
VoicePulse offers more
I need help,
I went through the Asterisk homepages and the links but i couldnt find
any configuration related to TDM 11B expect for the hardware
Now I bought an TDM11B (1 FXO Module 1 FXS Module) Dev Kit and manage
to install the cards with the help of the manuals
(i) modprobe zaptel
Try http://www.digium.com/index.php?menu=configuration#TDMX0B
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon
Sent: Monday, September 13, 2004 12:02 AM
To: asterisk
Subject: [Asterisk-Users] (no subject)
I need help,
I went through
David Cook wrote:
snip
3. Asterisk as a SIP server behind nat, clients on the outside
connecting to Asterisk
snip
Then it goes on to say:
* #3 Works with port forwarding and some header mangling magic
Can somebody explain a little more about the header mangling magic as
it is not discussed
The wiki page on Asterisk + Nat
(http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions) lists the
possible types of server/client relationships with one most probably
interesting to us being #3.
snip
3. Asterisk as a SIP server behind nat, clients on the outside
connecting to Asterisk
snip
hello,
if anyone is using asterisk as a voicemail system for SER I would be
grateful if i could see a working ser.cfg and extensions.conf of such a
setup. I am having some issues with rollover to voicemail when busy, and in
setting up a VM extension for users to retrieve their mail without having
Title: Message
Hi,
I have some queries
regarding Asterisk. They are as follows, any help would be greatly
appreciated.
1.We would
like to download modify Asterisk's sources distribute the binaries.
Is there any constraint on commercial use of Asterisk. Where can I get
more
Hello,
does anybody have any experience with CNAME
resource records in e164 zones.
Example:
e164.arpa zone
3.3.0.3.7.2.7.2.2.5.3.e164.arpa. IN CNAME
3.3.0.3.7.2.7.2.2.5.3.e164.lu.
e164.lu zone
3.3.0.3.7.2.7.2.2.5.3.e164.lu. IN NAPTR 100 10 u
E2U+SIP
!^\\+35227273033(.*)$!sip:[EMAIL
Hello again!
Just wondering if any one else has had a problem with stop and starting
asterisk?!? If I do it say 5/6times without restarting the computer then it
crashes. This doesn't seem normal to me, could this be because I'm running
fedora core 2? I know there's problems with using fedora to
Hi all,
I would like to study the asterisk source code(Program). I dont' know from which file
i've to start reading the code. can anyone helpme.
Regards
Shan.
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ShanKutti
Sent: Thursday, July 29, 2004 7:14 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] (no subject)
Hi all,
I would like to study the asterisk source code(Program). I dont' know from
which file i've to start reading the code. can anyone
Hi All,
I recently upgraded from a very old stable to HEAD. For some reason,
incoming callers don't hear ring tones when calling in. Everything else
is working fine. Where should I look for a fix?
ISDN -- X100P -- asterisk -- sipphones.
Thanks
Johan
Hi,
my setup:
Client: Win/linux client running x-lite or linphone
Server: debian running asterisk
on connect, incomming works well but outgoing to POTS has a lot of bad
sound (no, the mic is ok, using logitec usb headset). to ensure proper
work, tried normal p2p, worx well
the sound is nearly
Interesting though. :)
On Wed, 2004-07-07 at 01:32, eresmas wrote:
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Ok so here's one i have already asked but i don't know if anyone saw it
Has anyone managed to get the 'i' extension to work.
I have included within each context the following
exten = i,1,Goto(wrong-number,s,1)
then in
[wrong-number]
exten = s,1,GotoIf($[${EXTEN:0:2} = 43}]?10:2)
exten =
On Mon, 2004-06-28 at 09:55, Simon wrote:
Ok so here's one i have already asked but i don't know if anyone saw it
Has anyone managed to get the 'i' extension to work.
I have included within each context the following
exten = i,1,Goto(wrong-number,s,1)
then in
[wrong-number]
exten =
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: 28 June 2004 16:22
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] (no subject)
On Mon, 2004-06-28 at 09:55, Simon wrote:
Ok so here's one i have already asked but i don't know
Hello!
We are using the Digium 405PP card, and getting the following messages:
Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event:
6 on Primary D-channel of span 1
Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event:
8 on Primary D-channel of span 1
Jacob,
Please see here http://scottstuff.net/scott/archives/cat_asterisk.html
... Look for asterisk-lca-map
-jr
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jacob Hunter
Sent: Sunday, June 13, 2004 1:08
AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
201 - 300 of 396 matches
Mail list logo
