similar to: CAPI and Extensions.conf Security problem

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 05
1
Early B3
Does anybody can explain me what is early B3? Thanks! Ignace
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040506/937ae19c/attachment.htm
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