search for: mhess

...: betabar [1] 0 0 0 0 0 A [,1] [,2] [,3] [,4] [,5] [1,] 0.01 0.00 0.00 0.00 0.00 [2,] 0.00 0.01 0.00 0.00 0.00 [3,] 0.00 0.00 0.01 0.00 0.00 [4,] 0.00 0.00 0.00 0.01 0.00 [5,] 0.00 0.00 0.00 0.00 0.01 MCMC parms: R= 10 keep= 1 nu (df for st candidates) = 6 Mistake in chol(mhess) : the leading Minor of the order 1 is not positive defined By debugging: debug(rmnlIndepMetrop) rmnlIndepMetrop(Data=Data,Prior=Prior,Mcmc=Mcmc) beta [1] 1.46867390 -0.52889086 0.23710427 0.00188178 0.06021696 mhess=mnlHess(y,X,beta) > mhess X0 X0.1 X0.2 X0.3 0 0...

...his case.. just incoming from the peer.. I also saw in one list post that supposedly the accountcode entry in the sip peer should not be relied upon.. but no detail or explanation was given as to why that was.. -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 279 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050721/c27ccb4d/mhess.vcf

...have it hangs up.. I get "you have" and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 279 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041201/ed8c6939/mhess.vcf

Maybe I've missed it but I'm wondering if there has been any movement towards getting t.38 support into asterisk.. has there been any news? Where is t.38 support at? will it even happen? -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 279 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050316/b4e27b85/mhess.vcf

Haven't seen email since the 29th.. just testing. -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 288 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/96555713/mhess.vcf

...9;m rather in a hurry my thinking is somebody here has both run into and found a way to get this compiled and running. Using stable asterisk and the most recent app_conference from it's cvs on sourceforge.. -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 279 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050114/619247ec/mhess.vcf

...or even does receive the dtmf inband if it's using sip info anyway right? A few pointers would be appreciated as to smoothing asterisk out some so that other calls being setup do not affect current calls. -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 279 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050713/7744fd9c/mhess.vcf

...the list actively use g726 for anything and what have those experiences been? The g726 codec for me at least seems to be the poor man's solution for lack of g729 capability. (apply clue stick as necessary) -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 288 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/af706810/mhess.vcf

Does anyone have any success using AudioCodes FXO terminating calls ? Ehsan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051002/5cfef736/attachment.htm

Hi, I looked at the docs and probably missed it: is there a way to set a dialplan on the GXP-2000? (to avoid having to press "Send") Thanks, -- "Computers are useless. They can only give answers." - Pablo Picasso

I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial("IAX2/firefly@89280250/3",

I was doing a daily make update for asterisk. On the 19th the new version compiled fine. I installed it. All of my ata 186's can call out to pstn etc. All inbound calls, the phones ring but when you pickup, just silence both local and remote with no complaints in the cli. I backed down to the r 1.0 1.0.3 on the 20th CVS-v1-0-12/20/04-18:24:52 it worked ok. Yesterday I did a cvs update on the

...session id or not should not matter as the fact seems to be that asterisk allows this situation to happen when (from what I've been reading) it should not. Does anyone have any comments or thoughts on this? -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 288 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051018/a87739b8/mhess.vcf

I am currently experiencing intermittent silences with my asterisk system. The symptoms are as follows: * Both for incoming and outgoing calls, I (and other users) occasionally experience a brief period of silence. * The silence lasts anywhere from 3 to 10 seconds. * It is not due to silence suppression, because the silences generally occur in the middle of sentences. * Silences occur at

...) {A=matrix(rep(.01,nvar*nvar),ncol=nvar)} else {A=Prior$A} } R=Mcmc$R keep=Mcmc$keep s=Mcmc$s # Check beta0 argument # if (missing(beta0)) {beta0=c(rep(0,nvar))} # betadraw=matrix(double(floor(R/keep)*nvar),ncol=nvar) # # compute required quantities for indep candidates # oldbeta=beta0 mhess=diag(nvar) candcov=chol2inv(chol(mhess)) root=s*chol(candcov) priorcov=chol2inv(chol(A)) rootp=chol(priorcov) rootpi=backsolve(rootp,diag(nvar)) # # start main iteration loop # oldlpost=llmnl(y,X,beta0)+lmvn(beta0,betabar,rootpi) naccept=0 for (rep in 1:R) { betac=oldbeta+t(root)%*%rnor...

...6.185.98.152' Destroying call '1494993609221200001530@66.185.98.152' Destroying call '1494993807221200001530@66.185.98.152' Destroying call '1494994003221200001530@66.185.98.152' *CLI> -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 288 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/ce5ce526/mhess.vcf

Has anyone setup SIP services with ss7 and lis trunks? If so .. what was used hardware and software.. we're trying to do a SIP -> pstn setup and not having much luck as QWEST keeps pushing dates off (aka trying to screw us over) for our pri lines due to the recent court and fcc activity in regards to unbundled switching and I'm looking for solutions/ideas involving SS7..