similar to: SV: Turn off echo-cancellation when fax is detected?

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:

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

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.

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 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

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

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,

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

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

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:

How can I do this? I've set faxdetect=both in zapata.conf. Does this cancel echo-cancellation (echo-training) when a fax is detected or is this just for using exten=>fax, ... in extensions.conf.? I'm having trouble getting spanDSP - RxFax to recieve faxes. I am using Asterisk 1.0.8 and the fax number is registered in sip.conf Thanks, Arne Morten -------------- next

What about faxing yourself if you don't have a scanner? -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Johan van Tongeren Sendt: 5. september 2005 09:11 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] sending fax [macro-fax-dialing] exten =>

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

Hi again, Well - I didn't see beta8a-2.3.3 in custom dir. Will try. Also I tried to contact Sangoma - they are very fast to answer but main problem is time difference - it's 6 hours between Canada and Europe. Br, dmitry Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK-2600 Glostrup Denmark Phone: +45 70 25 74 74 Fax:???? +45 70 25 73 74 Web: www.comx.dk

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

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

That's just the thing, and it sucks, because the VoIP implementation actually works very good. Jon _____ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af list mail Sendt: 8. juni 2006 02:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] I can hear only one way when I use

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