similar to: From Database, PHP-Webinterface -> TO flatfileconfiguration

There was some discussion in the past about which one is the best Content Management System that can be used in conjunction with Asterisk. Mambo was supposed to be the best out there under GPL. The guys who developed Mambo have a new product now - Joomla. I am using this and it appears to be better than Mambo in many respects. Read the gist about Joomla below. ------------- If you've read

Hi there. Is it possible to have different sip users have the same CallerId number in sip.conf. I need this because we got multiple companies on this Asterisk box. Company A's internal numbers: CID: User: 1000 - User 1 2000 - User 2 3000 - User 3 4000 - User 4 Company B's internal numbers: CID: User: 1000 - User 5 2000 - User 6 3000 - User 7 4000 - User 8 Is this allowed? Regards

Hi list, I have a couple of questions related to asterisk billing and the generation of cdr logs. I've searched the wiki but have not found my answers, hopefully you guys can help. 1) When are asterisk CDR logs _normally_ generated? When the call arrives, when the call hangs up, or both? I have looked at the records created and it seems to only generate it at the time the call is

Hi there. I currently have a GotoIf statement that goes to a special extension priority if the CID match with one of the numbers in my "list" of CIDs. The way I've done it now is by multiple OR operators. There must be a better way. Anyone got some suggestions? This is basicly what I want. "If CID Exists in $File, goto s,10". So when I want to add a new CID I

Hi there. We're going to develop a call centre app for internal use in our office. The call centre is probably going to be a java-based client installed on a windows machine that our secretary can use. Features should be a way to see incoming calls, answer them, and then transfer the calls to our different users/groups/divisions. If it also could be possible to have a way to see if the user

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

Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail(${mailboxnr}@context) Thanks.

Hi. How does this work? What if this SIP/account was a member (agent) of a queue? Ex: dial(SIP/account,20,tT). Would the dialstatus be set as busy when one of the phones is actively talking, or will the other phones continue to ring? You may have seen my other submissions to this list. I'm looking for a way to make the other phones in a group unavailable when one of them is busy. Because

Ok. I've been searching the wiki and google for a long time now. HOW do I enable asterisk to send mail when users get new messeages in there mailbox? Do i need to change mailcmd in voicemail.conf? Regards, Arne morten

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

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

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.

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

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.

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

I often get a Maximum retries error while making outgoing calls. Why does this happend? Most of the time a reload solves the problem, but not all the time? What to do? Aug 26 09:52:46 WARNING[6613]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 32166-1B75-151E-E6DA-E37AEEAA2882@10.100.4.252 for seqno 1 (Non-critical Response) Regards, Arne Morten.

Hello. I just emerged spanDSP with all the packages needed. After a bit of configuration i was read to test. But i get this errormessage stating that application rxfax was not found. I could't fint rxfax i teh modules directory. I use asterisk 1.0.7. I did reset the server after emerging SpanDSP I use gentoo kernel 2.6 I don't know what else to do. Regards, Arne Morten

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:

Does the incomming call context in extensions.conf always have to be [default]? Can't i define different context for incomming like i can for Outgoing in the sip.conf? My default conf is getting very large. Regards, Arne morten

Hi. I'm having trouble controlling the user info when sending e-mails from asterisk via sendmail to a Microsoft exchange server. When I receive the email the sender is always "asterisk@TheDomainISpecify.com" and the name of the sender is always "Added by portage for asterisk". I want to change both sender-address and the name of the sender. I'm using Gentoo for my