Displaying 20 results from an estimated 1000 matches similar to: "CAPI and Extensions.conf Security problem"
2004 May 07
4
SIP Wokflow diagram
Hi everybody,
I would like to create SIP call flow Diagram under Windows. Is anybody
know a program to perform it? I have already Ethereal and I would like
an explicit diagram just to show where something have problems...
Thanks
Ignace
2004 May 17
4
Asterisk Proxy Type
Perhaps stupid question but, is Asterisk a statefull or stateless proxy?
Ignace
2004 May 03
1
Asterisk & MGCP / NCS
Hi everybody,
I have a MTA from Terayon that I try to make run with Asterisk using
MGCP channel.
The device is running with MGCP 1.0 NCS 1.0
Each time Asterisk try to send a Request (Request Notify, Audit
Endpoint....) the device returns error 510 "Protocol Error"
Does anybody have already meet this problem and provide me support to
make run it ?! (I have already try to change
2004 Apr 26
2
Registering a Grandstream Budgetone with Asterisk from Home
Hello guys,
I ask you to share your experience with your BudgeTone 100....
I have my asterisk @ work and I've bought a GrandStream BudgeTone (SIP
phone) and I usually use X-Lite
I have plugged my BudgeTone into my home network because I want to be
called even at home.
I succeed to register my X-Lite with Asterisk from home but I can't do
that with my BudgeTone. (I don't know
2004 May 06
3
mpg123 versions ?
We find that mpg123 0.59r works best. mpg123 0.59s-mh4 = the devil.
What versions does everyone use without problems.
0.59r is PERFECT
bkw
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2004 Apr 29
2
conference & sip
Good day all
I've installed asterisk with sip on my LAN,no special cards,if done
sip.conf and extensions.conf and all work 100,I'm using x-lite as a
client.
I'm trying to do conferencing.What I did was to has out the meetme.conf
looks like
[rooms]
conf => 9876
conf => 2345,9938
and extension.conf
exten => 9876,1,MeetMe,9876
When I go onto x-lite and type 9876 it gives me
2004 Apr 15
3
* Announcement * Astricon 2004 - call for speakers!
We're proud to announce Astricon 2004 - the first Asterisk user's
and developer's conference!
* Where? Atlanta, USA
* When? September 22-24, 2004
The conference is arranged in partnership with Digium.inc and the keynote speaker is
Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers
already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara
2004 Apr 28
4
Mysql Confusion..
Ok I know this may have been covered and I did have a look back in the
archives but didn't find anthing so I am asking it now..
Many moons ago the MySQL CDR functions and MySQL Voicemail functions had
to be removed from the main asterisk code because of licensing issues..
Now there is new MySQL stuff like MySQL FRIENDS for SIP and IAX
definitions..
So how is it that these options
2004 Jul 30
5
Non standard usage of X100P card.
I have two X100P card in my box. I want to connect regular phone (not the
phone line!) to one of thse cards. Does anybody think about the same?
I don't really want an expensive solution buying additional card with FXS
port, I prefer to make something by myself. It'll be great if somebody can
point me to technical materials or show electric scheme of such converter. I
believe it should
2004 May 28
1
Immortal SIP & NAT problem
Hi guies,
I know I know this subject have been The most written subject about VoIP
Right... but I just want to make clear, just one time !
If Asterisk is on a Public IP Address and a softphone behind the nat,
sip.conf must contains for this phone: nat=yes ....
Now if I want to configure my sipphone (X-Lite) placing behing the NAT,
it must have in "Domain/Realm" the external IP
2004 Dec 23
2
Re: Asterisk and Capi
Dear list,
I tried to install the CAPI support RPM from the SUSE 9.2 disks. YaST
tells me it is happy with the process. The Asterisk release I am using is
the one that comes packaged in RPM format, also included in the distribution.
Still starting asterisk with the usual asterisk -vvvc I see that
something goes wrong.
[app_capiCD.so]Dec 23 19:21:45 WARNING[1076850816]: loader.c:242
2004 Jan 16
7
CAPI not installed, after changed from i4l to CAPI
I had unexpected hangups from my asterix box using the i4l driver. (SIP
<-> SIP calls worked execellent, but SIP<->ISDN didn't.)
Then I changed the i4l driver in modem.conf with the chan_capi from
jungham. (http://www.junghanns.net/asterisk)
I followed his instructions in the INSTALL file, and first encountered
some errors compiling it. It help by deinstalling several
2004 Apr 29
1
Need an explanation about different protocols
Hello,
Is there someody who can explain me the meaning of these sentence.
"Sip is philosophically horizontal and H.323/MGCP are vertical"
Thank U
(if you have some links to share about this protocols, share it :) )
Ignace
2004 Jul 26
6
Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
Hi,
I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box
(customized kernel version 2.4.24). I want calls from my SIP soft-phones
to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap
HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc.
I've read everything I've found at www.voip-info.org, then I've downloaded
the
2004 Apr 29
1
CAPI ptp does not work
Hallo all,
I am trying to get * with chan_capi and a ptp-ISDN with 4 lines on a AVM C4
card to work.
But weather inbound nor outbound is working :(
My capi.conf:
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
mode=immediate
isdnmode=ptp
msn=8993
incomingmsn=*
mode=immediate
controller=1,2,3,4
softdtmf=1
;accountcode=
context=demo
2004 Jan 08
5
AbsoluteTimeout Users Messages
Hi, All
Is there a provision for "AbsoluteTimeout" application to notify
called and calling party of the reason why the call suddenly ended?
This way, the parties will be much better informed, hence they
will/should not think that
their VOIP/telco provider(s) are providing bad service.
Ta
SJ
2003 Nov 07
2
Callgroups and Pickupgroups in Console/dsp
Hi all.
I've made a patch for chan_oss.c to enable
callgroups and pickupgroups in it (since wasn't enabled).
I needed it for a special use of the console (pickup
calls arriving to the console from another phone)
btw, If someone is interested, I can submit a patch
to the bugtracker. I won't do it until
that's usefult for someone... since is a very special
features that probably no
2005 Jun 14
2
ERROR[5539]: chan_zap.c:9750 setup_zap: Unable to load config zapata.conf
Hi,
I have this error, I have a digium card TE110P Tiger3xx
When I'm load the dirvers by this command modprobe wcte11xp I got this
error
"Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
1 error(s) detected
FATAL: Error running install command for wct1xxp
"
when I'm Lunch the command ztcfg -vv
I got this error.
2004 Jan 04
8
Grandstream Handytone 286 RTP Problems
I am trying to get the handytone 286 to make a very simple call to * and
having problems. It registers with * just fine, but when I place a call
(to echo test, for example), the RTP stream seems to have problems
opening. Here is there error I get in *:
WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for