similar to: SIP 486 "Busy Here"

*Hi all, I got problems installing RSTAR, MSVAR, and MSVECM packages. * > install.packages("RSTAR")Installing package(s) into ‘C:/Program Files/R/R-2.15.1/library’ (as ‘lib’ is unspecified)Warning in install.packages : package ‘RSTAR’ is not available (for R version 2.15.1) > install.packages("MSVAR") Installing package(s) into ‘C:/Program Files/R/R-2.15.1/library’

Hi all, I have problem to estimate a SETAR model. I always get an error message. Here is the code: ## there are 4175 observation in the series (a). > a[1:10,1] [1] 1.496498 1.496602 1.496636 1.496515 1.496515 1.496463 1.496429 1.496549 1.496480 [10] 1.496498 > library("tsDyn") > selectSETAR(a, m=2)Using maximum autoregressive order for low regime: mL = 2 Using maximum

Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? ? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006

Hi, I know the problem now. Previously i use as.timeSeries function, but the error message of setar function still came out. Anyway, many thanks to Pascal for the solution. Best Regards, Ario On Mon, Jul 23, 2012 at 4:28 PM, Pascal Oettli <kridox@ymail.com> wrote: > Hello, > > It works for me (with a warning message), by adding this line before the > setar procedure: > >

Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA&SIP/deviceB&SIP/deviceC) As expected, by dialing 2000, all three devices will ring. And that's fine. However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an extension to dial in order to modify the dialplan. Here is what I

Hello As far as ive understood, you can just write Exten => s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and "${AVAILSTATUS}" would return "1" Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Brian Capouch Sendt: 27. juni 2006 09:10 Til:

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6

I've read about Fax to Email, but is there such a beast as email to fax? If not, what do people use to take care of outbound faxing?

I'm trying to integrate my Asterisk box with my call manager 8 server. I can call from the call manager to a phone on asterisk, but I can't call from a phone on asterisk to call manager. Any help would be greatly appreciated. sip.conf [2000] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes [2001] type=friend secret= dtmfmode=rfc2833

I'm looking to install Asterisk on an EM64T Dell 1850. PERC raid 1, 1GB ram, single 3Ghz Xeon. Any red flags or anything I should know? Should I bother installing a 64 bit OS? (gentoo-amd64)? Does asterisk work in 64 bit mode? Should I turn hyper threading off? Etc?

My server and client are running Ubuntu Lucid, libvirt-bin 0.7.5-5ubuntu27, qemu-kvm-0.12.3+noroms-0ubuntu9 and I'm using virt-viewer-0.0.3-6ubuntu7.xul19 or virt-manager-0.8.2-2ubuntu8 to connect. I configured SASL2 to use GSSAPI for libvirt following the instructions in the libvirt docs, created a keytab with libvirt/my.fully.qualified.domain at MY-REALM.COM (has a dash fwiw) and pointed

Hi all, i am playing around with several optipoint4x0 and run into trouble trying to get hint functionality to work. I notice that there is no status notifications. But afaik this should be implemented via the SUBSCRIBE/NOTIFY mechanism. I can see INVITE, TRYING, RINGING, ACK, BYE but no SUBSBCRIBE in my sip debug traces. I have problem to understand how hint priority works. I follow the

Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate

At approximately 3:15pm I shut down the office MySQL server to change out some hardware. Shortly after I received a call from one of two customers whose asterisk servers output CDR data to that server. They could not place or receive calls. Shortly after that I received a call from the other customer. I'm below providing output from the message log (At debug level). I don't see much

Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call

Hi All Has anyone managed to get the hint priority with Swissvoice IP10S phones working? I have 2 phones: a Snom 360, setup as the reception phone on extension 11, and a Swissvoice IP10S on extension 12. When calling each other (tested both ways) I can only ever see the Snom 360 in the Active State from 'show hints'. The Swissvoice stubbornly remains in the Idle State when on a call!

Ok, shaping on Linux is new to me.. so bear with me if i am just stupid. curtain:/etc/shorewall# grep TC shorewall.conf | grep -v ^# TCP_FLAGS_LOG_LEVEL=info TC_ENABLED=Yes CLEAR_TC=Yes TCP_FLAGS_DISPOSITION=DROP curtain:/etc/shorewall# So it should be enabled, right? ---- tcrules ---- 1 eth0 0.0.0.0/0 all 2 eth1 0.0.0.0/0 all 2 eth2 0.0.0.0/0

Hello, I'm trying to set up a conference room. When I dial it's extension, I get an audible error saying "Not a valid conference room, please try again" followed by a disconnect. I've got debug sip peer 1001 (my X-Lite client) and I see this in the logs: (I'm pretty sure it has something to do with ztdummy, but I dunno... I have the port installed, but I

Hi folks, I've been a trixbox user for a few years now but I'm thinking about jumping ship. Trixbox is great, but it's missing two features out of the box which are really important to me: outbound faxing (hylafax) and imap voicemail. I see no indication that they will be included anytime in the near future, so I have a choice to make - I've looked around at Elastix,

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24