similar to: three way call using sip - SOLVED -

Hi, i cant make a three way call using grandstream phones (BT-100) and asterisk using sip, is this supported or i need a zap interface? thanks

Thank you Miguel for your comments -- that''s exactly awdwall was intended for... happy to share that point with you too. Best regards, - Andrea > > ----- Original Message ----- > From: <mmiranda@americatel.com.sv> > To: <shorewall-users@lists.shorewall.net> > Sent: Thursday, March 10, 2005 12:34 AM > Subject: RE: [Shorewall-users] WebGUI Scripts

I sent this in first on 30 July. Now that UseR! is over I'm trying again (slightly extended version from last time). With R 2.5.1 or R 2.6.0 (2007-08-04 r42421) "L-BFGS-B" behaves differently from all of the other optim() methods, which return the value of the function when they are given a trivial function (i.e., one with no variable arguments) to optimize. This is not a bug in

Hello guys, I'm using llvm 3.3 on windows with mingw (triple: "i386-unknown-mingw32") to compile IR code that my program generates. I do it in two phases: first I emit assembly code and then assemble it. TargetMachine targetMachine = target.createTargetMachine(triple); targetMachine.setAsmVerbosityDefault(true); targetMachine.setFunctionSections(true);

Hi people, what is the current status of send/receive fax on asterisk extensions, i dont want to receive the fax and send an email or viceversa, i want to connect a standard fax machine to a Linksys' ATA (FXS RJ11 port) . Webdoc?, pointers? Thanks

Hi, im running shorewall 2.0.16 with centos 3 (iptables v1.2.8), everything is working fine for several days, i have configured a masq lan and all the outgoing traffic is ok, but now i want to redirect (port forward) the external web traffic to an internal machine, somethig like this INTERNET ---------> SHOREWALL -------------------> INTERNAL_MACHINE [public

Hi People, what files is processed first?, balcklist or rules, i want to globally filter imesh, but at the same time allow managers to connect, i.e. , imesh work on port 1214, i have this: /etc/shorewall/blacklist #ADDRESS/SUBNET PROTOCOL PORT 192.168.0.0/16 tcp 1214 192.168.0.0/16 udp 1214

I'm trying to get Asterisk to pass thru calls using the G729 codec. I've got a 7960 phone and my gateway is an AS5400. I got the following messages when debugging SIP (7778881000 is the 7960): WARNING[1872]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/7778881000-2874(4) to SIP/as5400-35c1(256) WARNING[1872]: app_dial.c:1002 dial_exec: Had to drop call

Hi, R developers. when running mle inside a loop I found a nasty behavior. From time to time, my model had a degenerate minimum and the loop just crashed. I tracked it down to "vcov <- if (length(coef)) solve(oout$hessian)" line, being the hessian singular. Note that the minimum reached was good, it just did not make sense to calculate the covariance matrix as the inverse of a

With the exception of "L-BFGS-B", all of the other optim() methods return the value of the function when they are given a trivial function (i.e., one with no variable arguments) to optimize. I don't think this is a "bug" in L-BFGS-B (more like a response to an undefined condition), but it leads to a bug in stats4::mle -- a spurious error saying that a better fit has been

So I am trying to use the mle command (from stats4 package) to estimate a number of parameters using data but it keeps throwing up this error message: Error in solve.default(oout$hessian) : Lapack routine dgesv: system is exactly singular: U[1,1] = 0 This error sometimes indicates that the list of starting values is too far from optimum but this is unlikely since I picked values close to where

Good Day list, Need assistance determining the best place to read up on whether Asterisk can help me out. I have a situation where I need to do the following <PRI from Telco> ------- <Analog Channel Bank>------------<Proprietary Box> | | | | | | <PRI Port 1 of Digium Quad T1> <PRI Port 2 of Digium Quad T1> | | | | | |

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Development Lab in Richardson, Texas testing our setup. We have a three server system

(inline) > On 7 Nov 2017, at 06:02 , Hugo Andr? <karlhugoandre at gmail.com> wrote: > > So I am trying to use the mle command (from stats4 package) to estimate a > number of parameters using data but it keeps throwing up this error message: > > Error in solve.default(oout$hessian) : > Lapack routine dgesv: system is exactly singular: U[1,1] = 0 > > This error

Hi list, I am a happy user of shorewall, i have followed the instructions in the shorewall''s web site relative to the squid transparent proxy configuration,all works ok, but i have been instructed to let adicional specified ports (aplications) to be snated (allowed to run) together to the web browsing service,i mean if i snat the network (i have a static ip from my isp), all

Hi people, is there any good document on setup asterisk using only its voip functions , i mean without e1/t1 hardware, i have a cisco 5400 gateway and a cisco 7000 working as gatekeeper, can asterisk register with this GK?, im thinking in using cisco voip phones Any pointer will be apreciated, and sorry for my bad english Miguel

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We

R 2.3.0 Linux, SuSE 10.0 Hi I have two problems with mle - probably I am using it the wrong way so please let me know. I want to fit different distributions to an observed count of seeds and in the next step use AIC or BIC to identify the best distribution. But when I run the script below (which is part of my original script), I get one error message for the first call of mle: Error in

i want to use Cisco AS5400 media Gateway as my PSTN Gateway instead of using the E1 PCI cards in asterisk box ,is this practically possible? can i use SIP in the connection between Asterisk and Cisco AS 5400 Gateway? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE!

Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie --------------