similar to: Early B3

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

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

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

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

Hello everybody, I am trying to understand the Advance routing and Traffic control. In the examples section of the how to an example script is given. Section 15.8.2 (the ultimate setup for Your internet connection) The example says download is 800 and upload is 220 ###### uplink # install root CBQ tc qdisc add dev ppp0 root handle 1: cbq avpkt 1000 bandwidth 10mbit Maybe I am

Hi, Is it possible to have the Asterisk CLI to scroll certain amount of lines and wait for, say spacear to be pressed to continue. If so how ? I am using the CVS stable version Thanks -- Togan Muftuoglu

I can not register via sip to iptel or sipgate and do not see sip into ethereal. I do not unterstand why thats Wudu .. but i am new to asterisk and sip. I am behind a susefirewall2 but asterisk even do not register if it is down. The asterisk is running onto the machine witch is connected to the internet. No answer seems coming back from iptel (sip debug in asterisk). Ports are open (5060,

Hi, I've installing a AVM Fritz Card in my ASterisk Box I've configured everything and its running perfectly. The problem is that everybody 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

Hi, Just to pass the news [1],shorewall firewall package will be part of the next openSUSE release, meaning users would not need to set up an extra repository (of course they can always choose to follow the development repo). Thanks to all answering my questions Togan [1]<http://lizards.opensuse.org/2011/08/05/factory-progress-2011-08-05/>

Hi, Just to pass the news [1],shorewall firewall package will be part of the next openSUSE release, meaning users would not need to set up an extra repository (of course they can always choose to follow the development repo). Thanks to all answering my questions Togan [1]<http://lizards.opensuse.org/2011/08/05/factory-progress-2011-08-05/>

Hi, Attached is patch for correcting zones in Samples6 for iv6 Thanks Togan ------------------------------------------------------------------------------ Achieve unprecedented app performance and reliability What every C/C++ and Fortran developer should know. Learn how Intel has extended the reach of its next-generation tools to help boost performance applications - inlcuding clusters.

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, 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

Hi, I've just installed Asterisk 1.2.0-beta1 for the first time and downloaded chan_capi and compiled it in and run it. (For comparison purposes, I've tried this with both chan_capi-0.4.0PRE1 and chan_capi-cm-0.5.4 and chan_capi-cm latest CVS). Whilst most things are fine, it seems that if I specify the 'b' parameter in the dial string before the number, sometimes the early

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

Dear sorry for my next post on the same object but i need to know if is possible to hack libPRI to have the function that KAPEJOD made in CAPI for the "early B3" function. I'm the only in the list have this type of problem with Free Message? :-) Thanks in advance Dimitri

Hi, is there any mean to get ringing tones for the E100P, according the signals from the telco? When I use the r-option in extensions.conf, I have too early ringing tones, i.e. when busy, I have ringing tones for a while and afterwords the busy tone. If I don't use the r-option I have no tone at all. Is there anything similar the b-option for chan_capi? Roger.

Hi, I recently upgraded to Asterisk 1.4.1 from Asterisk 1.4.0 and suddenly my early B3 seems to have stopped working. When I call a number that is out of service from my SIP phones, instead of hearing the announcement from the telco, I immediately receive a SIP Busy message and the call is disconnected. hear no audio. This was working fine in Asterisk 1.4.0 with exactly the same

Hi, I tested the capi_chan with latest cvs of * and I have problems with Early-B3. The following dialstring works for me (without Early B3): exten => _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30) But if I add the 'b' for using Early-B3 exten => _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30) nothing changes (no dialtone). If in this example the called party discards the call, there is no

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