similar to: SV: sending fax

[outgoing-fax] exten => _0XXXXXXXXX,1,SetVar(NumberCalled=${EXTEN}) exten => _0XXXXXXXXX,2,Wait(10) exten => fax,1,SetCallerid(${FAX_CALLERID}) exten => fax,2,Dial(Zap/g1/${NumberCalled},60) exten => fax,3,Hangup exten => t,1,Busy exten => i,1,Busy -----Oorspronkelijk bericht----- Van: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]

I also want to try that asterisk guide. But i'm not sure if i understood it correctly. What exactly do i need to do? Do i need to compile Asterisk with the spanDSP plugin or just configure extensions.conf? The URL to spanDSP in the guide wasn't working. I also use a traditional internet line to recieve calls and hopefully i will get Fax working soon. This is so confusing. Thanks, Arne

Thanks. I'm able to getting the asterisk calling back to my cellphone. But when I get to the authentication I get this message when I start to dial in my password: NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received Is this a DTMF failure of some sort? Thanks again. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com

Thank you very much. For some reason "emailsubject" was not included in my example config. Well, it's working great now. Last question, I promise :P. Is it possible to change the date format? I want it in Norwegian. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Barry Flanagan Sendt: 22.

What is the output of your logfile when you try to ping a public ip? Besides, you should change your internal ip addresses to private addresses (rfc 1918): 10.0.0.0 - 10.255.255.255 (10/8 prefix) 172.16.0.0 - 172.31.255.255 (172.16/12 prefix) 192.168.0.0 - 192.168.255.255 (192.168/16 prefix) best regards, Kenneth. -----Opprinnelig melding----- Fra: Marta

It's fixed now. In "/etc/mail/ssmtp.conf", this ("FromLineOverride=YES") line was commented out. Removing that comment did the trick :) Now I only need to change the e-mail's title. Is that possible? -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Barry Flanagan Sendt: 22.

Yeah I did change those. I'm using 1.0.8 (Or was it 9?). It seams that the system overrides these settings? -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Barry Flanagan Sendt: 21. februar 2006 14:54 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Re: Fromstring when

I thought the same thing before I made my reply but zapata.conf seems to be the only config file that deals with echo at all. From what I understand of 'echotrain' is that at the beginning of the call it sends a short signal out that measure echo in an attempt to try and cancel it. I was wondering if you tried using it and if so was it of any help? Sincerely, Steve But is Zapata.conf

As mentioned earlier I did try that. Someone suggested that there might be an issue with sendmail not "trusting" the asterisk user. And the default behaviour of that is to not allow modification of the "fromstring" and "serveremail". So if you have any idea how to fix that in Gentoo I would really appreciate it. Thanks -----Opprinnelig melding----- Fra:

Just one more question. In /etc/passwd there's a line with "asterisk" and "added by portage" in it. Can I just change this without screwing up everything? Or is there a command to change user info or something? As you can see, I'm not so good in Linux. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com

Yes your chap-secret file could be the issue, it should look like this: Your_usrname * Your_password * That?s it :) But if that don?t work include this file to the list: /etc/ppp/pptpd-options It should look something like this: ipparam PoPToP lock mtu 1490 mru 1490 ms-wins 172.x.y.1 ms-dns 172.x.y.1 multilink proxyarp auth +chapms-v2 ipcp-accept-local ipcp-accept-remote lcp-echo-failure 30

Thank you for response. So a "standard" quad bri card can serve the KX-TD1232, no need for anyt special fxs similar equipment. Would like not to invest much more money into the Panasonic, I wan't it out :P BR Amund -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Peter Svensson Sendt: 14. juni

Hello AstLinux seems quite suited for my use. Can you configure more incoming port via a web interface? I'd like to install it to a "normal" hdd. Can that cause any problems? BR Amund Nygaard -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Kristian Kielhofner Sendt: 4. juli 2005 03:23 Til:

There is no static interval. But i found out that it was my IP-Phone Service Provider that was having serviceproblems today. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Giorgio Incantalupo Sendt: 26. august 2005 11:33 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users]

Yeah sorry about that. But I didnt see my message in the list, so I thought it didn't ame through. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Dave Cotton Sendt: 14. september 2005 19:45 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] RxFax problems On Wed,

I also have some other trouble. How the I send the caller to voicemail (next extension) if the Member => SIP/phone stops answering for a defined period of time. I cant figure out if this would work (from queues.conf): ; If you wish to remove callers from the queue if there are no agents present, then set ; this to yes. Note that this is for use with dynamic queue members! ; ; leavewhenempty

Yeah I do use ring groups at the moment. But the problem is that I can't control "the flow". Let's take your example. dial(SIP/dev1&SIP/dev2&SIP/dev3) If I dial these 3 numbers, and dev2 is already one the phone. How do I check for that? I only want one of the three phones active at the time. But if no telephone is busy, they all should ring until the call is

Oh. So how can I do this? If I write something in PHP, how do I make it output to an Asterisk variabel? I need to set a variable in asterisk to TRUE or FALSE based on the result of the PHP-script. ________________________________ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Morgan Gilroy Sendt: 8. februar 2006 15:28 Til:

Jeff, the tip on http://stat.bell-labs.com/RS-DBI/download/index.html (at the bottom of the page) may help. Trond Rafoss -----Opprinnelig melding----- Fra: Jeff Lee [mailto:JLee at acamllc.com] Sendt: 10. desember 2003 23:34 Til: 'r-help at stat.math.ethz.ch' Emne: [R] How to start RMySQL I am a newbie in R and want to use MySQL database. Here is what I have and done. R 1.8.0 mysql

Thanks to Roger Bivand, my problem was solved. binutils 2.14 was updatet to binutils 2.15, as indicated under. Geir Helge Systad Norwegian Institute for Nature Research Arctic Ecology, Polarmilj??senteret, N-9296 Troms??, Norway Adress aug-04 to feb-05: Albert Pettersonsvei 13, N-5750 Odda fax +47 85 03 82 14 phone +47 53 64 24 94 mobil +47 91 63 70 55 E-mail: geir.systad at nina.no