similar to: cannot dial any extension except xlite

I've tried to setup multiple extension dialing - ie dial 1 number and it rings at a number of sources. For the most part its worked.... Now if someone dials 107 it rings Sip phones at 102 and 107, then goes to voicemail after 40 seconds. exten => 107,1,Dial(SIP/102&SIP/107,40|r) exten => 107,2,Voicemail(u102@pstn) exten => 107,3,Hangup exten => 107,102,Voicemail(b102@pstn)

I have just installed the latest version of R on a openSUSE 12.1 system running on an ORacle VM VirtualBox and have encountered a problem with installing ChemometricsWithR. Here is the output: > library("compiler") > install.packages("ChemometricsWithR") Installing package(s) into ?/home/computation/R/x86_64-unknown-linux-gnu- library/2.15? (as ?lib? is unspecified)

Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect

Hello, I try to exectute a command on my NT server from my unix server with the following command : smbclient \\\\monnet01\\C$ -U admin_nan -c "help.exe" C$ is my share admin_nan is the user Smbclient doesn't ask a password so i get the following error message : Anonymous login successful Domain=[FR-MON] OS=[Windows NT 4.0] Server=[NT LAN Manager 4.0] tree connect failed: ERRDOS

Hello, I'm experiencing various output bugs using --out-format with %f in rsync 2.6.9 I test it with rsync 2.6.9-2 and 2.6.9-3 in debian etch, running both in deamon and client mode on the same host. This bug seems to be known by the developer as said in http://samba.anu.edu.au/ftp/unpacked/rsync/NEWS : NEWS for rsync 3.0.0 (UNRELEASED) Protocol: 30 (changed) Changes since 2.6.9: BUG

Dear list, sorry for bothering you with a probably simple question... I searched the archive but couldn't find the right answer. I have computed upper and lower bounds of a region and would like to plot them in a boxplot. My problem is now to get the two boxplots in the same graphic so that the boxplot of the upper bounds is directly on top of the boxplot of the lower bounds. The reason I

I have gotten this sort of problem some weeks ago and have solved it with the mailing list archive. But now I have this problem again with the class library. What should I put in library/class/R/zzz.R? Everything I put in that file leads to the same error message. I'm running 0.61.3 on Linux. > knn1(train, test, cl) Error in .C("VR_knn1", as.integer(ntr), as.integer(nte),

I've been using R for the last few weeks, but not with much success. If I try sourcing test.R from the 'class' package, I get this error: Error in .C("VR_knn1", as.integer(ntr), as.integer(nte), as.integer(p), : C/Fortran function not in load table I checked the class.so file, and sure enough VR_knn1 is there. Why can't R find it? By the way, I'm running RedHat Linux,

I've been using R for the last few weeks, but not with much success. If I try sourcing test.R from the 'class' package, I get this error: Error in .C("VR_knn1", as.integer(ntr), as.integer(nte), as.integer(p), : C/Fortran function not in load table I checked the class.so file, and sure enough VR_knn1 is there. Why can't R find it? By the way, I'm running RedHat Linux,

Hello, I have samba 2.0.7 installed on AIX4.3.3. I have 2 bugs : - the first is that I can't transfer large file (more than 4GB) : I've seen there is a bug and patch - the second one is a secure pb referenced by a colleague : a root access on the destination machine Are these 2 problems corrected on version 2.2.8a ? Thanks for your help. Sophie VERDAVAINNE NIS SchlumbergerSema * :

Hello all I asked similar question some time ago but didn't get answer... Maybe this should asked in asterisk-dev or bugs.digium.com? For example, I have 3 sip phones defined in sip.conf - 101, 103, 109 and this simple dialplan: [local-ext] exten => 101,1,Dial(SIP/101,,t) exten => 107,1,Dial(SIP/107,,t) exten => 109,1,Dial(SIP/109,,t) exten =>

That way we can ditch our gem_prime_res_obj implementation. Since ttm absolutely needs the right reservation object all the boilerplate is already there and we just have to wire it up correctly. Note that gem/prime doesn't care when we do this, as long as we do it before the bo is registered and someone can call the handle2fd ioctl on it. Aside: ttm_buffer_object.ttm_resv could probably be

Hello, I'm experiencing various output bugs using --out-format with %f in rsync 2.6.9 I test it with rsync 2.6.9-2 and 2.6.9-3 in debian etch, running both in deamon and client mode on the same host. This bug seems to be known by the developer as said in http://samba.anu.edu.au/ftp/unpacked/rsync/NEWS : NEWS for rsync 3.0.0 (UNRELEASED) Protocol: 30 (changed) Changes since 2.6.9: BUG

Hello, I am using the mda package and in particular the fda routine to classify/predict in terms of color to a set of 20 samples for which i don?t know the color. I preformed a flexible discriminant analysis (FDA) using a set of 147 samples for which i know all the information. My script and data follow in attachment. A total of 23 predictors were considered. 20 of the predictors are

Hi, I installed asterisk server in my linux box. I configured a user 1000 using xlite and registered with asterisk server in the same linux box. I configured one more user 1001 in other box and this user also got registered with asterisk. But i am facing two issues here. 1. When a call is made from 1001 to 1000 i could see an incoming call blinking but no audio flow is observed. 2. When i made a

Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp

I am using the latest xlite phone to connect to the latest version of asterisk (20040727). When I try to make a call the xlite phone tells me "Call not approved". I used the configuration options that were listed on the wiki. The context in the sip.conf file is "from-sip". I have a matching context listed in the extensions.conf file. The phone is able to register

-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 -- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 RFC3389: 5 bytes, level 0... Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Killed Whenever I make a call between extension 101 and 1009 which are both Xten Xlite SIP clients, I get that error and

Hello Asteriskians! I have an Asterisk box with a simple HFC card in it and a bunch of people using the Xlite software to connect. The HFC card is connected to an internal extension on our legacy PBX. So far so good. The Xlite clients can call each other, and the internal extensions on the PBX and the Xlites can call each other, no problem. The problem is when using an Xlite to dial an external

Unfortunately I seem to have another problem! I am using sipgate for the incoming line - and it appears that you cannot get DTMF to work in that configuration. Unless anyone knows anything different of course!! > > I just want one of my incoming numbers to go to an IVR service that > > will allow me to select what I want. > > > > For example > > > >