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...onfirmation or other comments about getting the following apps to work > in WINE -- so I can finally make the jump to Linux as my desktop OS and > not just my server OS! > > Those programs are: > - Flash 5 > - Dreamweaver Ultradev > - Rational Rose Enterprise Ed. 2000 > - Kinetix 3D Studio Max R5 > > Thanks in advance, > -Rob Flash 5 works in Linux. I've gotten Flash 4 to work but haven't personally tried Flash 5. Linux specific versions are available from Macromedia.com Not sure about the middle 2 but as far as 3DSMax goes there is Houdini for Linu...

Hi, I want to be able to bridge two sip channels using direct RTP between my endpoints (Audio IP : not local) but without using reinvites. So I set up my asterisk sip endpoints as follows: [test1] type=friend host=dynamic username=test1 dtmfmode=info context=test_rtp allow=all canreinvite=no directrtpsetup=yes [test2] type=friend host=dynamic username=test2 dtmfmode=info context=test_rtp

...ium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tzafrir Cohen Sent: 05 December 2008 13:27 To: asterisk-users at lists.digium.com Subject: [asterisk-users] top posting again [was: Re: CDR Design] Top posting strikes again: On Fri, Dec 05, 2008 at 01:39:59PM +0200, regs at kinetix.gr wrote: > Quote : "Like I said earlier - the CDR's aren't > reliable enough for a billing platform (as you've > rightly pointed out) but are OK for very basic call > logging (something the customer can look at)." Who wrote that? [snip the rest of the reply]...

Hi, A glance at sourceforge.net/projects/openh323 Help Forum made me wonder if this location is the one to use (I got trouble in the past when google pointed to an obsolete site) : some quite old messages remain unanswered. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tzafrir Cohen Sent: 05 December 2008 13:27 To: asterisk-users at lists.digium.com Subject: [asterisk-users] top posting again [was: Re: CDR Design] Top posting strikes again: On Fri, Dec 05, 2008 at 01:39:59PM +0200, regs at kinetix.gr wrote: > Quote : "Like I said earlier - the CDR's aren't > reliable enough for a billing platform (as you've > rightly pointed out) but are OK for very basic call > logging (something the customer can look at)." Who wrote that? [snip the rest of the reply]...

hi sorry about the urgent but it is urgent i have problems configuring a connection between asterisk and avaya using H323. the module i am usign is ooh323 what do you need to help me? and any tip or hint? thanks!!! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An

Hello! Most are probably bored seeing another letter about this, but I've put in a fair amount work on a spec for rewriting the CDR system in Asterisk, and I have some questions: First, please look at what I've written so far: svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs and look at the file "CDRfix2.rfc.txt" in the RFCs dir. The spec SIGNIFICANTLY alters the way

Hello, On an Asterisk 1.4.33.1 in a simple scenario: [test] exten => _X.,1,Dial(SIP/12345 at peer01,,,) exten => i,1,Hangup(${HANGUPCAUSE}) exten => t,1,Hangup(${HANGUPCAUSE}) exten => h,1,Hangup(${HANGUPCAUSE}) I have noticed that no matter what value we set in the Hangup(<cause code>) commands, if the call is not answered by peer01 for any reason, the actual cause code

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

Hello, Does anyone know whether chan_sip in Asterisk supports DTMF in clock rates other than 8000? I looked for telephone-event/16000 in the changelog and in Jira but no luck. Any help would be appreciated. -- Best regards, Vlasis Chatzistavrou.

Hello, I have used the Dial option 'r' before in older Asterisk versions and it behaved as expected, that is, Asterisk would always give ringback audio before the call was answered no matter what. It has been a while that I have used version 1.4.33.1 without any the 'r' option. Now that I had to use it for a while, I noticed that 'r' would not give ANY audio until the