similar to: Asterisk@home .05 release questions on setup.

Displaying 20 results from an estimated 7000 matches similar to: "Asterisk@home .05 release questions on setup."

2004 Dec 26
7
IAX Registration Refused
I tried to connect my * to IAXtel, but i always get this errors. chan_iax2.c:5849 socket_read: Registration of 'mnetwork' rejected: Registration Refused On dial a iax number i get: chan_iax2.c:5526 socket_read: Call rejected by 69.73.19.178: No authority found chan_iax2.c:5528 socket_read: Immediately destroying 3, having received reject chan_iax2.c:2411 iax2_hangup: We're hanging
2004 Aug 29
2
Sip device not login or register calls to that device go to busy voicemail not un-available
I feel this is in error some place. If I call a sip device that is not registered or not connected at the time. Asterisk will send that call to voicemail to busy not unavailable. Is there a way to correct this? Ariel Batista Kasi International - Computer Networking Ph: 305-574-6721 Fx: 305-574-0212 -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Jun 06
4
*@home .conf files request
hi all, can anyone emailme the .conf of asterisk at home, i cant download the full size tar or iso because of a network problem that fu*** every big file download.... and i just wanna learn not change my distro bye and thanks! -- Luis Diaz - Un obsesivo con proyectos! :oP
2005 Mar 29
2
Asterisk@Home 0.7 released Question/Problem
I'm new to this and have tried to find the answer in the discussions and docs but to no avail. I even read the posting saying the password issue has already been discussed. So, at he risk of being exiled, here goes. Question 1: I've installed 0.7 and can log into the asterisk server from windows by typing http://192.168.1.11 I can log in with wwwadmin and the password I set myself
2006 Mar 28
3
aah 2.7 / BRI
First encounter with * Just downloaded & installed aah-2.7 Started up AMP, but i can not find any reference towards isdn. I presume there has to be some configuration done for my Eicon-Diva-pro. Does aah actually support isdn-bri? On the mail-archive i found some references, but these are rather old ( they speak about the coming release of aah-2.1) aah-handbook (version 1.6) doesn't
2005 Aug 31
1
Asterisk@Home: How to changed AMP User Login and Password
Hi, I can't figure out how to change User Login and Password for AMP. By default it is user:admin and password:maint. Anybody knows how to do it? Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/c8ef57b2/attachment.htm
2006 Mar 27
5
FreePBX & AAH
Does anyone know if FreePBX can be installed on a Linux box that was built using Asterisk@Home. I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only
2006 May 17
2
Asterisk@home default password doesn't match
Hi all, This is my first post! I'm newbie, yesterday I installed Asterisk@home, and I got lot of Kernel panics, after trying to reinstall it, this messages dissapeared. My problem now is that the default password for maint user in AMP is not working... I got this error message when I try to connect via another computer in the same network, after trying to log -> maint/password: FORBIDDEN
2005 Jul 25
2
Operating AAH v1.1
Hi, Just set up AAH 1.1 using an HFC BRI line and 5 IP phones as per http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone The dialplan was configured through AMP and has nothing fancy in it. As a first time user of not only Asterisk, but also a PBX, there are some operator teething problems. After much googling & searching of voip-info.org, I cannot find any answers to these
2005 May 18
4
Pickup other ringing phone
Hi everyone, Is there a simple way of answering a different ringing extension from a sip phone using AAH? I have absolutely zero technical know-how when it comes to modifying conf files etc. Still working on figuring it all out. ;) That brings me to my second question... where the hell does one find an extensive manual of sorts that explains all conf files and what the strings all mean etc?
2003 Dec 17
5
Readline & readline-devel installation on RH9
I have a new user question. Sorry I know most of you are Linux experts I am not! I am just getting my feet wet with this. And I am sorry to ask this stupid question. I was following an installation post from Wiki that said when using RH 9 you need to make sure that you have the following installed first and you should check them with the following command. Are there any other items I need to
2004 Jan 22
5
Snom 200 phones not working.
I have 2 Snom 200 and would like to get them to work properly with Asterisk. With the Firmware 2.02t I am able to use the phone. But only one line configured. With there newer firmware 2.03o it will not allow me to make calls. But I can get calls on the unit. Again the 2nd line is not able to be registered. Is this an issue with Asterisk or Snom? I could use some example configuration
2005 Mar 04
1
defold usernames in asterisk@home version 6
OK. So check out the Wiki here.... http://www.voip-info.org/tiki-index.php?page=Asterisk The archive of this list can be search via google by entering... site:lists.digium.com <some parameter> www.digium.com has a link to all the materials for getting started in the Documentation section of the website. Those are really quite good so I would start there. Most were written prior to
2003 Oct 13
4
IAXTEL/ Dial problem
Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed XXXXX where my passwords are. CLI> Executing Dial("Zap/14-1", "IAX/abatista:xxxxxx@iaxtel.com/917009965342@iaxtel") in new stack -- Calling using options
2003 Sep 17
4
Programming 976 numbers from dialing out.
I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! exten => 1900XXXXXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => 1XXX976XXXX,1,Congestion exten => 91900XXXXXXX,1,Congestion exten =>
2003 Nov 24
4
Sip phones!
I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor
2005 Mar 28
13
Asterisk@Home 0.7 released
We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2
2003 Dec 22
2
Sipura 2000 configuration.
Ok here is another problem I have run into. I have a Sipura 2000 and I have been able to configure line 1 with only one small problem. But I can't get the line 2 working with asterisk. Here are samples of my sip.conf and extensions.conf. If I disable line 1 I can then get line 2 working. Is there a sample configuration for the Sipura to get both ports working with Asterisk. Sip.conf
2003 Oct 07
3
Line going to Zombie
I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 & also soft hangup
2005 Mar 18
2
PSTN > Voicemail
This is probably a stupid question.. How do I login to voicemail from the PSTN? I can dial *98 from within the system, but when dialing from the PSTN I have it set up to ring a dial group, then to an extensions vmail. During the extensions vmail prompts, I dial *98 and it sends me to the directory. -------------- next part -------------- An HTML attachment was scrubbed... URL: