similar to: Sipura SPA-941 not available after Asterisk & Freepbx upgrade

when check java process lock statistics, plockstat failed, please see below: # prstat -mLp 21162 PID USERNAME USR SYS TRP TFL DFL LCK SLP LAT VCX ICX SCL SIG PROCESS/LWPID 21162 7677 0.9 0.1 0.0 0.0 0.0 99 0.0 0.3 83 89 215 0 java/81 21162 7677 0.3 0.1 0.0 0.0 0.0 0.0 99 0.2 106 33 305 0 java/35 21162 7677 0.1 0.0 0.0 0.0 0.0 100 0.0 0.1 79 6 85 0 java/59

Is there a way to recover Metropolis-step acceptance rates AFTER completing posterior draws? The immediate application is in the probit.bayes and logit.bayes models used by Zelig... which I believe is merely calling MCMCpack. So one strategy, to which I am fixing to resort, is to call, say, MCMClogit with verbose set to mcmc (or mcmc divided by an integer) and then look at my screen.

Greetings, After the modesetting commit, I have been unable to compile xf86-video-nouveau. The git site for drm given on the wiki page does not get the proper xf86drmMode.h, etc. I noted that libdrm git has over 100 branches and I really don't want to try all of them. Fedora 9 has a patch of the libdrm source rpm that includes an earlier version of xf86drmMode.h but it's now out of

This is a new branch of an old thread, made necessary because the email system here purges sent messages after a period of time so I can't reply to the last message in the thread. The operative portion of that last message (retrieved from the archives and dated July 3, 2023) follows: /*****/ So I set up a fresh key to use for this test, and gave it similar parameters. I wasn't aware of

Hi, there. I have asterisknow beta 5 with the following data: Ip 192.168.0.60 mask 255.255.255.0 gw 192.168.0.1 the router (a linksys) has port forwarded the port udp 5060 and from 16384 to 16482 udp-tcp from the internet to the asterisk machine. the only protocol allowed is g729. Which work fine for the ip phones I already have setup in the LAN. My problem is trying to register to a voip

I have noticed that Asterisk (version 1.2.13) is not hanging up a call when the wifi handset moves out of range. My setup is Nokia E61 connected to wifi access point (private IP range) and then to server on internet (public IP). I have been testing using the talking clock application, and walking out of range does not hang up the call. The call will continue for hours even though the handset

Has anyone played with this phone? i cant seem to get it to work properly, i manged to get it registered and can make calls from it, but i havent been able to make it receive calls. Weird thing its that if you make a call from it and while you are on that call you dial its number does calls go thru in second line, but as soon as you terminate both calls it wont recieve any calls again. Heres

Hello, I'm wondering why asterisk is not transferring the callerid to the sip device. Scenario as follows: sangoma <---> zaptel <---> asterisk <---> sip <---> SIP-Device zaptel is reporting the callerid, but in the sip packages the sip-address shows unknown as user part, as this sip debug package shows: Executing Dial("Zap/62-1",

Hi I got the email-id from the web site. Kindly let me know whether we can use ssh to communicate b/w unix & windows(the initiation has to be from unix only), if yes then how & where can i find the releated informations If its not possible usnig the ssh then whats the other way to pull or push file from Unix usnig a encrypted way where the password is not hacked on the network. Kindly let

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

Asterisk 1.4.29 BLF-SUBSCRIBE go to internal IP (ngrep output): U 2010/03/02 11:34:06.013515 212.78.xxx.xxx:2048 -> 62.134.xxx.xxx:5060 SUBSCRIBE sip:12 at 62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP 212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport..From: <sip:K922002626 at 62.134.xxx.xxx>;tag=vyx8c0trgx..To: <sip:12 at 62.134.xxx.xxx>;tag=as13e7cb7c..Call-ID:

Hey all, Has anyone had any issues with phones having multiple lines that are in different contexts? We've got a couple phones that we're testing intercom functionality for, and I'm noticing that for some strange reason, no matter what line we use, the phones tend to be completely in one context or another, not segregated like I would expect. Our contexts look like this: context

[Since samba-bugs has been shutdown, I'm posting this here...] I'm running RH 7.0 with both 2.2.16 and 2.4.2 kernels I've got the following samba packages installed: samba-client-2.0.7-21ssl samba-common-2.0.7-21ssl I have a Windows NT 4 server (SP6a) which stores many of my files since I use both Linux and Windows to access them (Linux on my desktop, Windows on my laptop) I

Hi, I'm looking for a full list of xml provisioning variables of the SPA-2100/3000. Currently the Sipura website has example XMLs only for the SPA-841 [1] and SPA-941 [2]. I'm mostly interested in the CallerID type selector variables and whatever variables control the PSTN<->VoIP settings. Sipura Configuration website form field names are numeral only. :( [1]

I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is

I've been struggling with a Trixbox running Asterisk 1.6 for one of our customers as of late. The service provider in question is using BroadWorks and requires a single trunk registration for the trunk group. We have 4 users(lines/numbers) in the TG. The sip trunk is setup as follows: type=peer host=192.168.1.1 fromuser=<tgid> fromdomain=<sip domain> dtmfmode=rfc2833

I have a Cisco 1760V with FXO ports hooked to POTS lines talking SIP to asterisk 1.8.15.0. imagining in extensions.conf: exten => 1,1,Dial(SIP/121) exten => 2,1,Dial(SIP/121&SIP/122) When a caller dials extension 2 /and/ I have trustrpid=yes generaterpid=yes sendrpid=yes in sip.conf and I use the pickup exten, the caller is disconnected. see:

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> I have an Asterisk 1.4.18 with a mix of cordless phones connected using Linksys SPA2102 ATAs and Cisco 7940G

Hi. I am thinking about building an asterisk system for a small business and want to be able to page through the phones. It seems like to do this asterisk needs auto-answer support in the phone. I know the SPA-841's support this, as do Cisco phones, but I have been unable to determine if the new Cisco/Linksys SPA-941's do. Does anyone here have experience with trying to use auto-answer

I'm trying to disable call waiting for Linksys SPA-941, but unfortunately as far as I have seen, there are no parameters on the web interface regarding this feature. I just want callers to hear the busy tone when the called party is at the phone. Probably I can accomplish this by using the "disable call waiting" in asterisk as well, but I have not been able to find any