similar to: Audio problems ins conference zap->sip

Anyone know if it is possible to control how aggressively the "Aggressive" mode behaves. Meaning, is it possible to dial back the aggressive mode to have a happy medium between Regular and the Aggressive defaults. I have a situation where Normal echo cancellation is not quite enough, however when I turn on aggressive mode We are attacking it to hard and I am unhappy with the walkie

Hi all I am beginning in asterisk and am making tests with an ata-186. For the time being the tests are going well, however have a doubt. I am thinking about using a canal e1 with plate digium. Assuming that the company of telecommunications supplies e1 with 30 canals and numeration to me 4000-0001 4000-0029. she is possible to configure asterisk in way that somebody of is dials 4000-0025, to

Hey sorry to hijack this thread, but I just remembered a request I wanted to make to the speex devs. I tried using the activity detector, but I just couldn't get it working well. I ended up using my own, where I think it just considered voice on if it passed a certain threshold (I know, pretty primitive). I also tried one that checked for a signal, like if the strongest frequency

Hi Using 2.2.1a and getting 401 SWAT Bad Authorization error. If I use the -a option everything is OK. The documentation refers to the -a option which leaves everything wide open but I can't find any documentaion regarding any other form of access control for SWAT. What actually controls the access? We installed from source with no selected config options. Hope someone can help, I have

Greetings everyone, I've been having some strange issues with my Asterisk box and some snom phones. In some cases, when I talk, the sound in the other end is cut off, I stop earing the background noise - looks like a walkie-talkie. I've tried this between phones in the same network and in all but one this happens. The one where it doesn't happen is the one connected directly to the

> Anyway, my request is, can you build in a pre and post buffer into the > VAD? In mine, if I detect voice any time between now and say a quarter > second later, I start sending, and then I wait a half second or whatever > after I stop detecting. You pretty much have to have this, or people > start getting anxious talking over an internet stream. They have to > enunciate

homals-0.9.0 is on CRAN -- by Jan de Leeuw and Patrick Mair This package implements the methods discussed in Gifi, Nonlinear Multivariate Analysis, Wiley, 1990. In the Gifi terminology it covers homals, princals, canals, morals, criminals, and overals. The R implementation fills several gaps in Gifi, adding multiple ordinal, numerical, and polynomial data transformations. Differences with

homals-0.9.0 is on CRAN -- by Jan de Leeuw and Patrick Mair This package implements the methods discussed in Gifi, Nonlinear Multivariate Analysis, Wiley, 1990. In the Gifi terminology it covers homals, princals, canals, morals, criminals, and overals. The R implementation fills several gaps in Gifi, adding multiple ordinal, numerical, and polynomial data transformations. Differences with

> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > > the JITTERBUFFER function? > > You only need to use the JITTERBUFFER function. > > The jbenable option will enable a jitter buffer on every channel > created for that peer (or, if global, for every peer in the system). > Depending on the version of Asterisk, it will also place the

(Please accept our apologies if you receive this message multiple times) **** CALL FOR PARTICIPATION **** *************************************************************** *** ** EARLY REGISTRATION DEADLINE: May 25, 2012 (CET) ** *** *************************************************************** The 21st International ACM Symposium on

(Please accept our apologies if you receive this message multiple times) **** CALL FOR PARTICIPATION **** *************************************************************** *** ** EARLY REGISTRATION DEADLINE: May 25, 2012 (CET) ** *** *************************************************************** The 21st International ACM Symposium on

Hi Guys I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest of the ports in their own groups so that I can have different signaling on each? [channels] language=en switchtype=euroisdn pridialplan=unknown resetinterval=600 echocancel=yes echotraining=yes

ISSUE: In CommandLine.h, gcc 2.96 thinks that the apply() template function, when called as: apply("Some text string", x) should be expanded to applicator<const char[n]>("Some text string", x) instead of applicator<char[n]>("Some text string", x). ACTION: Duplicate the template specialization for applicator<char[n]> as applicator<const

Well, to do an upgrade on a traditional system you have the same issues, perhaps even worse as everything is physically wired to one system. To develop for production you must have a dev environment, a beta test and a scheduled release right? Todd Jonathan Moore <moorejon@usd465.com> wrote: __________ >These are good issues, but I am even thinking of something simpler and more

Hi, I have a SIP client which is registered to asterisk. Asterisk is registered to a SIP trunk and also handles the media. Now since my client has some issues in its RTP Tx, which seems to have some amount of jitter (mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms and max delta is 85 ms), to over come that I have enabled jitter buffer in the SIP channel by setting sip.conf

4 Port PRI sangoma a104 From: jg [mailto:webaccounts173 at jgoettgens.de] Sent: Wednesday, March 18, 2015 2:09 PM To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 4 Port PRI I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest

Hi all again, I've got a problem with sporadic one way audio calls, which means sometimes I can't hear the calling party (call is established, but audio is missing). Today I received ~90 calls, one of them got this problem. I've got two networks involved, without NAT: - 192.168.1.X, in there one nic of my server and all the phones - a private net to my provider, in there a nic of my

hello every one, i have an asterisk system and want to act as gateway and send calls to cisco gatekeeper. this is my h323.conf file: [general] port=1720 binaddr=192.168.0.YY context=from-trunk faststart=yes h245tunneling=yes gatekeeper=192.168.0.XX //cisco address progress_setup=8 progress_alert=8 dtmfmode=rfc2833 jbenable=yes jbforce=no jbmaxsize=200 jbresyncthreshold=1000 jbimpl=fixed jblog=no

Hi, Asterisk 1.4 Working (jitter buffers created as expected): ZAP -> SIP SIP -> ZAP Not working (no jitter buffers created): SIP -> chan_local (with /nj) -> ZAP SIP -> chan_local (with /j) -> ZAP SIP -> chan_local (with no flags) -> ZAP I have this in zapata.conf: jbenable=yes jbforce=no jbimpl=fixed jbmaxsize=300 Is there something I haven't tried that will make

Hi all, I'm fighting with a really strange problem that is really busting me. I have an asterisk 1.4.22 ( from a trixbox 2.6.2 ) and mISDN 1.1.7 3 extension on hardphone and 3 extension in softphone ( zoiper ) What happens is that sometimes the people on the other side of communication hear my voice as metallic and chopped. This happen either on incoming call than on outgoing call. If I