similar to: Possible bug in app_meetme.c

In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I

We've started to use Asterisk for conferencing and have been getting some complaints. Our configuration is that some people call in from home, but we have a physical conference room with a Polycom. When somebody was giving a presentation in the physical conference room, we were told that the remote people kept hearing him cut in and our. To me, this sounds like the talking optimization was

Hi there I am using IAX2 softphones dialing into meetme conferences. I also have jitterbuffer=yes, with typical jitterbuffer settings. The problem I am having is that as soon as there is a delay from a participant, then the delay continues until the participant hangs up and dials in again. When dialing in again the delay seems to go. It seems to me as though as soon as the server registers

I'm having a wierd problem. Somewhere around 1-2% of the time, the first DTMF digit dialed gets dropped. This is occurring during a SpeechBackground application call. If the caller reenters the digits when given a second chance, all is OK. Any suggestions how to debug this intermittent problem?

Hi all ( dev & user list ), I'm starting to implement the missing features in Meetme application : 's' -- send user to admin/user menu if '*' is received Line 438 -------- app_meetme.c ----------------------------------------------------------------------------- else if ((f->frametype == AST_FRAME_DTMF) && (f->subclass == '*') &&

Hello Asteriskos, Screenshot: http://www.areski.net/asterisk-meetme/about.php The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has

It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that?

I'm confused exactly what's supported with LDAP and Asterisk. What I want to do is to have SIP peer information read directly (in realtime) from LDAP. Can this be done? If so, with what Asterisk versions?

> CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan.

Hello, Situation: I've got two asterisk 1.2.4 servers, connected to each other over the internet with IAX2 with about 20msec delay. One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are

Interesting. Are you using the included cbend.php script to terminate conferences? I occasionally get questions about using WMM with Confbridge, and to date I have not had an answer . If you can provide details, even vague ones, about how you did it, I can update the WMM package. Dan -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at

Hello all ! How can I make conference authorization based on pin number ? I have: exten => 1,1,Meetme,1234|ps|2222 where 2222 is a pin number and this doesn't works Where do I have to add information about pin number ?? Greetings Andrzej Radke

I'm getting dozens of these at a very high rate: [Jan 20 09:15:27] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received from '"" <sip:121 at gnat.com>;tag=as5f1a9480' [Jan 20 09:15:28] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received from '"" <sip:130 at

It's kind of low for me. How does one control that volume?

Which speed recognition products will also recognize DTMF? In other words, I want to say "Please speak or dial the conference number". Does Vestec allow that? LumenVox? Any other way?

I recently tried to update my Aastra 57i to version 3.2 and ran into a problem. It won't properly register and says "contact mismatch". I added "sip contact matching: 2" to aastra.cfg, but that didn't help. When I look at the SIP trace, but I see is the Aastra sending a REGISTER and Asterisk replying with the 401. The phone then sends the REGISTER again, this time

How does one do this? We have a particular SIP phone that needs a large jitterbuffer, but all I can see is how to put it on the *read* side of the channel.

Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and

I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to*

When I run ./configure, it aborts with: checking for uuid_generate_random in -luuid... no checking for uuid_generate_random in -le2fs-uuid... no checking for uuid_generate_random... no configure: error: *** uuid support not found (this typically means the uuid development package is missing) But it *is* installed: [root at asterisk asterisk-12.2.0]# yum list installed | grep uuid uuid.i386