similar to: Were i make mistake

HI I have 3 Linksys SIP901 IP phones I also have a pc I'm not using it amd athlon 1800+ 512mb ram and 40 gb hdd I'm looking to connect this phones together and to make calls between them Not from outside of my lan I don't know how to configure asterisknow beta Can somebody help I'm doing this in my house to connect rooms With respect Ardit Saliu

Hi all ,? I have managed to install and configure the? 1. asterisk-1.8-current 2. dahdi-linux-complete-current I did not faced any issues during the installation. After that I installed X-Lite soft phone in two different PCs and tested the setup. every thing was success. I was able make calls from each?extensions. But when I observe the log files , i could see some messages ......

Hi all, I tested with few Asterisk versions from 1.4.18 to 1.4.21, same result. Here is the problem: I have a peer -which is peer AND user- setted up like this [MyPeer] ; type=peer host=xxx.xxx.xxx.139 deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142 permit=yyy.yyy.yyy.yyy/255.255.255.255 context=from-MyPeer dtfmode=auto disallow=all allow=ulaw,alaw

Dear All I have this setup [asterisk]----[mediant2000]-------E1 Trunk----------[Avaya] When i call from avaya to asterisk i got long ringing tone then hangup but when i call from asterisk to avaya i got 4 ringback and then hangup with this error *CLI> Jun 26 01:26:08 NOTICE[5555]: chan_local.c:523 local_alloc: No such extension/context 1022 at mysip

Hello, I am having trouble passing DTMF digits from a Polycom 330 SIP phone to my FXS/FXO lines. I am running Asterisk 1.4.21.1 In sip.conf I configured dtmfmode=inband. RTP traffic (voice) goes perfectly from SIP to FXS, but in the SIP phone I only hear a continuous noise. However, when I press any digit in the pone (FXS) I hear the DTMF tone fine in the SIP phone (the noise goes away for as

I have tried everything and still can't place / receive calls from the fwd network. At one point today I was able to call my test machine on the fwd network, I'd answer the call on the test machine (which stated Call Connected), but then the computer I was calling from, through the Asterisk server would give me a 403 Error. I am using sjphone software. I am able to call various

I've just setup a new asterisk box (cvs HEAD) with a digium tdm411 and a couple computers with eyebeam. I have one small. I cannot call the eyebeam clients from the phone connected the fxs port. I can call the phone from the eyebeem clients. And, I get both the fxs phone and eyebeam clients to ring when a call comes in through the fxo port. I have been trying to get this straightened out

hello list, I am trying to setup a PXE boot server. Below are the details of the server: OS : Fedora Core 5 32-bit DHCP : dhcp 4.1 TFTP : tftp-hpa-5.0 Number of NIC : 2 eth0 : IP : 192.168.100.17, Subnet : 255.255.255.0, Gateway : 192.168.100.1 eth1 : IP : 192.168.1.1, Subnet : 255.255.255.0, Gateway : 192.168.1.1 Client: OS : None, fresh machine Number of NIC : 2 Intel Gigabit eth0 of server

Using a default xen3.0.1 setup for dom0(rhel4 distro) and domUs (gentoo, rhel4, centos4.2 distros, ttylinux) with the domUs all having dhcp turned on, is there an easy way to tell from within the dom0 what ip addresses were assigned to the domUs after bootup? Toby Ford USi _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com

Hi all, After successful joined a new DC (my third), no new DNS entry was created. The end of samba_dnsupdate --fail-immediately --verbose Calling nsupdate for A bp.net 192.168.1.240 (add) Outgoing update query: ;; ->>HEADER<<- opcode: UPDATE, status: NOERROR, id: 0 ;; flags:; ZONE: 0, PREREQ: 0, UPDATE: 0, ADDITIONAL: 0 ;; UPDATE SECTION: bp.net. 900 IN A 192.168.1.240

Hello all, I have a Salicru UPS (Salicru SPS One 700VA) working great in Debian Wheezy, but I want to move it to a OpenWRT router. I configured same as in Debian but I can not get it working: # upsd Network UPS Tools upsd 2.6.5 fopen /var/run/upsd.pid: No such file or directory listening on 192.168.1.240 port 3493 /etc/nut/ is world readable Can't connect to UPS [salicru]

Hai, ? I have automated the install of my member server. Followed the wiki : https://wiki.samba.org/index.php/Samba/Domain_Member? ? Everything works nicely, but...?.. read on..? ;-) ? ok, so wiki says: https://wiki.samba.org/index.php/Setup_and_configure_file_shares? ? and now im at the point : SeDiskOperatorPrivilege and .. for the DC's installed this worked without problems... ? but

Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI> rtp debug RTP Debugging Enabled -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered

Hello, I was using garchFit {fGarch} to fit some GARCH processes. I noticed that the result contains "Log Likelihood" value (right above "Description"), but when I use .. at fit$llh to retrieve Log Likelihood value, the sign switched. I am confused about which value I should choose to report... Any help here? Thanks a lot! Ted -- View this message in context:

Hi all, I make a mistake. The partition with syslinux and the minlogon component was put complete to ram by initrd. But the partition, that then boots is not the image in ram but the first partition on the new harddisk. I cant see this first, because there are no possibilitys with the minlogon component. But I wondered, why there was letter C: for the partition that has been booted and than letter

dear r-help. i want top report a mistake in the documentation help(splineDesign). the last sentence of "value" is " Each B-spline is defined by a set of 'ord' successive knots so the total number of B-splines is 'length(knots)-ord'. " it is not correct! one b-spline is defines by a set of 'ord+1' successive knots. take an easy example: degree 0 is a

Huhu, I followed the HOWTO, but I guess I did something wrong URL to appdb-article: http://appdb.winehq.org/objectManager.php?sClass=version&iId=3126 When I start the app with cmd at the last step of the first howto point I get this error: bas at Ubuntu:~$ ~/wine-war3/wine "C:\Programme\Warcraft III\Frozen Throne.exe" /home/bas/wine-war3/wine: could not locate Wine source tree I

Hi, I've just managed to completely nuke most of my Mailing Lists folders in Outlook - will someone using Outlook Express (or similar) please export all the Vorbis-Dev messages from the 6th of June 2001 onwards(starting with "[vorbis-dev] problem with vorbis autoconf stuff" by Bill Nottingham) and e-mail me them (send them in a ZIP file if they're over a few hundred Kbytes)?

This was sent to me regarding the wiki. ---------- Forwarded message ---------- From: "Martin Kon??ek" <mkonicek12 at gmail.com> Date: Mar 7, 2013 4:44 AM Subject: mistake on Securing SSH To: <timothy.ty.lee at gmail.com> Cc: Hi TImothy, I saw wiki http://wiki.centos.org/HowTos/Network/SecuringSSH and it is pretty good, but there is a mistake. *Instead of having* iptables

Hi, according to the description of Pickup() on page http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup I can use this command to pickup a call at a certain extensions. When I try this with e.g. exten => *8200,1,Pickup(200) Asterisk tells me that the highest value for the Pickup command is 63. Wenn I enter the number of a callgroup instead of an extension, I can pickup the call.