similar to: spa 3102 configuration

Hi to everybody, I have an spa 3102 where i connected an analog phone (in the fxs port) and the pstn line (in the fxo port). This is my problem: the incoming call doesn't arrive to asterisk. In the spa web page i configured this dialplane: (<:line01@192.168.1.220:5060>) where line01 is the context in sip.conf, 192.168.1.220 is the asterisk ip and 5060 is the asterisk sip port.

Hello to everybody, I have a problem with the disposition filed that asterisk write in mysql table. What I notice is that for every outbound calls (for example to a mobile phone) I see in disposition field the string "ANSWERED" when I reject the call and also when I really answer the call, while in the variable DIALSTAUS I have the correct status of the call (BUSY, CHANUNAVAIL,

I would to know if it's possible with R put an apex on a character ( i.e 10-2 or aij ) Thanks in advance Damiano -------------- next part -------------- An HTML attachment was scrubbed... URL: https://stat.ethz.ch/pipermail/r-help/attachments/20011105/75518704/attachment.html

I would like to know how I can set with a R-command the values from decimal to exponential (e.g. from 0.001 to 10^-3) on the axis y. Thank you Damiano -------------- next part -------------- An HTML attachment was scrubbed... URL: https://stat.ethz.ch/pipermail/r-help/attachments/20011105/472776fb/attachment.html

Hy, I have dowloaded the R-version for Windows. I would want to plot different pictures on the same device, but I am not able to do it (i.e I would like make 2 qqplot on the same graphic). Can you help me? Thanks in advance Davide reply to: d.tarfanelli at libero.it -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read

Where and how do you put custom logger statements in cucumber? I understood (more or less) how to do this in rspec in the spec_helper file but I do not know where to start with cucumber. I want to add a simple identifying text line in the log file to assist in picking through the output. Something akin to: Running Scenario: X Feature I should have this logged .... Feature This should

Hello, I am using Samba 2.2.8a with 2.6.6 CIFS Linux clients and I've got problems with symbolic links : I can't make them point outside the share. I understand that it is a security feature if the server resolves the link. But, in my case, I'd like Samba to serve link files *as is*, so that the Unix clients resolve the link. Symlinks are an important filesystem feature and their

Dear all I have ported the pppoatm.c plugin for ppp version 2.3.x to the 2.4.1 ppp. I am however having problems to decide what is the right place where to send it. I did look at http://www.samba.org/ppp and this seems to be the home page for ppp Then I did look for a mailing list the only one I have found is majordomo@vger.kernel.org but I really don't know if that is official for ppp

Hello everybody, Which one is a better choice 1. Gateway device with FXO <-> SIP ( example Addpac http://www.addpac.com/addpac_eng2/addpac_product_view_detail.php?class_id=19&item_id=59 ) 2. Digium (Wildcard TDM400P) 3. Sangoma (A200 Analog FXO/FXS) All i need is to put asterisk in place with 4-8 incomming lines (ordinary POTS ). With IVR, Voice mail and International Call via SIP.

I would want to know if it's possible to plot two histograms on the same graphic in order to compare the bins one to one, like a Excel graphic. If it is possible can you help me to do it, because I don't know the exact R-commands. Thanks for your help Damiano > > >Martin Maechler <maechler at stat.math.ethz.ch> http://stat.ethz.ch/~maechler/ >Seminar fuer Statistik,

Hi, Is it possible to make a filter with tc which exclude an IP like iptables ?? ex: iptable -t mangle -A PREROUTING -i eth0 -d ! 192.168.1.222 -j MARK ... I try the ! with tc but it doesn''t work. Thanks, doude. _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

Dear Developer: i am working a project including iSCSI booting. i have a iscsi server with windows xp system image and fedora system image, and a booting server providing dhcp and tftp services.I made some menus using pxelinux and things are working fine with linux, but failed with windows . i use the following strings in my dhcpd.conf: ....... filename = "pxelinux.0";

Team: My Debian Sarge "testing" box got upgraded from Samba 2.2.6, I believe, to Samba 3 Beta 2 in the latest dist-upgrade supplied spate of updates (90+, and I missed Sambe being in there . . . ). Not a big deal except that now I have lost access to the logon profiles directories at logon time. It's a basic Samba setup. Uses Unix passwords and Unix accounts, etc. Nothing fancy

Dear all, I want to setup the incoming calls, that don't use authentication in sip.conf file. My configurations as follows, [2000] type=peer host=dynamic insecure=port,invite ; (both) context=Testing But when I call '2000', I noticed the following message in Asterisk console, "NOTICE[5702]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user

I am trying call transfer with asterisk. blind transfer (#) is working perfectly, but attended transfer doesn't fonction (*2). I don't know what is the problem. Anyone could help? _________________________________________________________________ Lancez des recherches en toute s?curit? depuis n'importe quelle page Web. T?l?chargez GRATUITEMENT Windows Live Toolbar aujourd'hui !

Hello I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite on XP as an SIP client: http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png The problem is that by default, Asterisk doesn't rewrite the CID name + number in incoming calls, so that XLite displays whatever name I used in the 3102 and the extension the 3102 uses to register with Asterisk. How can I tell

Hi, I have a Linksys SPA 3102 using as ATA. The call routing is : Phone -> PSTN -> SPA 3102 -> SIP Proxy -> Asterisk The problem I am having is that when the phone hangs up, SPA 3102 can't detect it and relay the CANCEL message. Is this problem with my SPA 3102 config or it just works like that by default? Thanks in advance for your help. Regards, Mark -------------- next

So I was able to get some things up and running. However, I created classes/base.pp : class base { file { "/etc/ldap.conf": ensure => exists, owner => root, group => root, mode => 440, source => "puppet://192.168.1.222/dist/apps/base/ldap.conf", } file {

Has any advancement been made to get 3102 operational in either a SIP or H323 asterisk environment. A post back in time mentioned a downloader service. >From the posts and articles I have read, the NCP is acting like a bootp and tftp server which uploads the configuration to the phone?? Am I close? if so, where does one get the SIP image for he 3102 and 2102 phones? I had 8 donated, but they

Dear Expert, I am stuck when trying to register SPA-3102 on AsteriskNow .. could any body please advise .. where can I find the article for doing this? .. I googled but got nothing.. Regards bie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071203/6bbbecd8/attachment.htm