search for: ignace

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

Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Ignace

...geTone 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 what exactly fill in the configuration web server). Each information will be usefull Thank you in advance Ignace

...ice 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 header into the chan_mgcp.c from "MGCP 1.0" to 'MGCP 1.0 NCS 1.0" but always the same result) Thanks in advance Ignace CARIA

...dy is allow to call through it. Explaination: All users registered in Asterisk can make a call towards the ISDN network But, everybody from the Internet, knowing the extension of CAPI in the dialplan, can call through my Asterisk to any phone number Heeeeeellllllppppp mmmeeee please ! Thanks Ignace

Does anybody can explain me what is early B3? Thanks! Ignace

Hey, I''m looking for a way to get the content of my div in my Rails view. I would like the literal HTML content, so only HTML and no Rails I thought inner_html would work, but I get a RJS exception (TypeError: $("content").innerHtml is not a function) Does anyone know how I can get the content? Thanks! In my controller I use this code: [code] def save_test render :update

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

I've looke throught the archives and can't find much to help my situation. I have a Slackware 10 server running Samba 3.0.4 on a Win2k/2K3 Active Directory Domain. My win2k clients can connect to the samba shares but the WinXP clients cannot. I've tried the enableplaintextpassword hreg hack but that doesn't solve the issue. Here is my [global] section of my smb.conf Can anyone

Hello, Does anybody have already use TEDAS VoIP DECT PABX or some Kirk hardware to integrate DECT to VoIP? Thanks! Ignace

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the

Hello, Does anybody know if Asterisk can support QSIG protocols to be interconnected with a Traditionnal PABX? (Using a HFC chipset based ISDN card to emulate NT Interface) Thank you in advance ;-) Ignace

...r' Version'(null)' Accoding to RFC: 510 = The transaction could not be executed because a protocol error was detected. BUT I have read that somebody ( Roy Sigurd Karlsbakk) have "nice" response form his D-Link Device.... I need more information....or a miracle .... :-s Ignace

...bindaddr = 213.177.xxx.yyyy ; My Pubic Address. context = incoming_SIP disallow=all allow=alaw allow=gsm allow=ulaw #include sip_additional.conf Extensions.conf exten => 4700,1,Dial(SIP/4700@213.177.xxx.yyy|30|m) Adress to Contact... Please help Regards, Ignace

...ble, it launchs the Voicemail application. Right... But I cannot heard anything (the little pretty voice of the woman in Comedian mail) in my Budgtone. Is it a codec problem? SDP negociation? RTP flow? If you encountered the same problem, please reply (with an answer of course :-)) Thanks Ignace

I have been fooling around with DynExtenDB and run into two glitches. 1) The code is looking for (chan->dnis) and in my case I find (null). I forced (chan-dnis) to be the same as (chan->exten). So far so good. Now I can connect and talk. This lead me to the second glitch. 2) As soon as the call ends by hanging up, the code issues a (ast_spawn_extension). This causes asterisk to drop

Hi, We have been working with samba 3.0.4 and MIT 1.3.6 on SUSE SLES 8 (SP3) for some time. We use security=ads and our samba is joined to the W2K3 domain. We use this with winbind to use the authentication from w2k3 instead of local users (telnet, squid, samba and via smbauth also apache). When preparing a new server with suse SLES 9 and heimdal 0.6.1 RC3 (suses package) we started running

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

Hello all, I am not able to compile zaptel... I am using kernel 2.4.21-202 and it's source is linked to /usr/src/linux. I already have to test installations running at SuSE 9.0 (but here I have an SMP system with to processors). Could you give me a hint? /usr/src/linux/include/linux/highmem.h:93: error: `clear_user_page' undeclared (first use in this function)