similar to: Starting * with G729 licences

Hi, Can anybody explain me in configuring Asterisk to record a conference? Regards... Girish _________________________________________________________________ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now!

Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change the default SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this

Hi all: Somebody knows the mysql table structure for VoiceMail2 application? Thanks in advance, Gus

Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus -------------- next part -------------- An

Hi all: Very disappointed, finally I left the attended call transfer with ATA 186 using SIP. With image 2.16-1, ATA sens '486 - Busy Here' when trying to transfer the call.. I consulted with Cisco guys and accepts that some problems with this service exist. Soon as I can I will try using MGCP. My doubt now is if somebody proved the DynExtenDB application. I read some commentaries but

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

How do I use the Authenticate application in my IVR menu, where do I put the password? here is my menu. I need to ask for a password before I let users log into my conference room. [conf1] exten => s,1,Ringing exten => s,2,Wait,2 exten => s,3,Answer exten => s,4,Authenticate(1234) exten => s,5,Hangup exten => a,1,Meetme,1251 I also can not figure out what "Unknown RTP

Hi all: I've got some core dumps with chan_skinny. The client is ATA186 with v2.16.1.ms ata18x (Build 030814a). The * version is CVS-10/05/03-16:03:26. When I make a call, the phone connected with ATA rings only 1 time and * dies. Maybe I have some errores in ATA config. If someone has proven configs for ATA, please send me the details. Thanks in advance, Gus The logs: *CLI> Version

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

Hello, I have that problem with codec G729. Please can somebody help me! WARNING[16384]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 == Detected 4 licensed G.729 transcoders WARNING[16384]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to

Hi all: I've got some core dumps when I use chan_h323. I dial an extension using h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes * hangs, sometimes not. The client used for test es SjPhone (http://www.sjlabs.com/). This is the data for one core dump: (gdb) bt #0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790 #1 0x41f8879c in create_connection

I haven't been able to find the answer online, and am not currently able to conduct an experiment to find the answer... I understand that in a SIP call where G729 has been negotiated as the preferred codec, a G.729 licence is not consumed until there is a need to perform transcoding, e.g. play a non-g729 sound, or do voicemail, or enter a Meetme, etc. What happens when a SIP call in progress

Just updated from CVS 12-23-03 to tarbal 0.7.1. Identical settings in zconfig.h for echo cancellation (MARK2, aggressive OFF). The echo got worse, much worse. It takes longer to train and overspeak now disables cancellation, which it did not before. In fact, I now have echo on VoIP to VoIP on our local network, which I never had before. Was something changed with the echo supression which

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

I send to the list the perl script to make the inverse of sys2ansi.pl and convert any ansi file to syslinux format. I recommend jave editor to deal with ascii arts and ascii fonts www.jave.de and for painting colours over the ascii file, the ansieditor from http://www.win.ua.ac.be/~wschrep/ansied/ with the patch attached for import and export to syslinux directly and for colour paint mode to

Hi all: I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message: *CLI> WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs! The "show codecs" command shows: *CLI> show codecs 1 (1 << 0) G.723.1 2 (1 << 1) GSM 4 (1 << 2) G.711 u-law 8 (1 << 3) G.711 A-law 16 (1 <<

That wasn't the part that confused me. What confused me was what you expected to be encoded into the instruction. Your math indicated that you multiple n by 3 and added 1 to it to get your 13 bytes. So that means you intend to use 4 bytes to store an address in 32 bits which implied to me that you intended to have a fixed address encoded. But what if the address in in a register as would often

Whoops - sorry for the confusion. n would be set in stone beforehand. I basically meant to indicate that we'd either be looking at a 32 bit or 64 bit system, ie 4 byte or 8 byte addresses. On Tue, Mar 20, 2018, 1:07 PM Craig Topper <craig.topper at gmail.com> wrote: > Hi Gus, > > When you say "n byte destination" you mean you want to encode an n byte > address as

Hi Gus, When you say "n byte destination" you mean you want to encode an n byte address as a constant within the instruction? That would mean you couldn't encode an address that comes from a register. ~Craig On Tue, Mar 20, 2018 at 9:48 AM, Gus Smith via llvm-dev < llvm-dev at lists.llvm.org> wrote: > Hi all. > > tl;dr: I would like to add a long x86 instruction

There is no immediate support for this in the test-suite cmake buildsystem. There is something in the Makefile system indeed, but it lacks documentation and I'm not aware of any examples of how to use it. I would recommend to use `CMAKE_C_FLAGS=--save-temps=obj cmake ...` in the test-suite. The --save-temps=obj flag will make clang place a bitcode file next to each .o file. - Matthias >