Huh? Dude, that is just plain silly talk.
There is no excuse. That is not how forums work. When you need help you (remember that others may have the same issue) it is much better to start a post that is the subject of your issue. Then it shows up that way and when the issue is resolved it makes it much easier for those that follow to find answers.
It also shows a modicum of respect for the people that help out here.
sami.alio,
If you can’t read and follow the guidelines you’ll not be getting support from any of the major contributors here. Use common courtesy and read and follow the guidelines then you don’t belong here.
If you don’t like what you are getting for Free then please feel free to pay for support.
As Cosmicwombat, Skykingoh, and myself have said please learn to post, not hijack thread.
It’s pretty simple you’d not like others to bust into conversations you are having with something else totally irrelevant. That is exactly what you are doing here.
Please show some respect and post a new thread that is all we are asking.
I Installed a perfect smooth installation of asteriskNOW 1.5. everything seams to be working except nothing registers to the box.
can someone help ???
It just boggles the mind how many MORE-ONs have added MORE ON to this thread after OBVIOUSLY not reading ANYTHING that came above. Well, samsch, you can add yourself to the list of people who are NOT going to be helped.
Wow,
I actually read this thread to get more information on posting and such. I was amazed to find people asking questions here…LOL!
ctlw83,
Thanks for taking the time to read it and understand more about what we need to help you.
Yes some people just have no clue as you can see…
I installed follwing that:
http://www.freepbx.org/support/documentation/installation/install-process-for-centos-5-1.
After all, I test the installation by http://localhost/adnin as the installation steps showed, but the web page is forbiden to access. What and why is that ? What I have to do so I can use the web interface to manage the PBX.
Thank for your helps
yet another one who can’t read…
dear friends, i have a Centos 5.3 v2.6.18-128.1.10.el5, with freepbx 2.4.0, a Sangoma card A200 installed with 2fxs&2fxo. from local network, ipphone can talk with each other, but when i want to make out call, i have the message: all curcuits are busy now, please call again. with a rapid busy tone.
route name
: 9_outside
dial pattern
NXXNXXXXXX
NXXXXXX
trunk sequence call is configure to take zap/g3, first outline.
i’m confused now
thank’s
Don
dratte490, if you have read this entire thread you must have seen that this is NOT! a thread to ask questions in.
Read the first post, then read it again. Then create a NEW thread with all relevant information and I am sure that people will try to help you.
DO NOT REPLY TO THIS MESSAGE!!!
In order to understand why people don’t read the thread, you might try opening this thread in a browser and looking just at the first posting. Then look around (without scrolling down) - try to imagine what it looks like to someone who’s new to getting help from fora such as this.
The thing that struck me was that it isn’t obvious at all that there is a thread at all. Not until you get adventurous and scroll down to see /if/ there is anything at all. Clearly some people aren’t adventurous enough to do that. It just looks complete as is.
The title “So you have a problem and want help?” invites people to ask for help; the trouble is that it may not be obvious to a complete newbie that you don’t ask /exactly/ /here/, you have to figure out somewhere else to go.
Now you and I all know that these people shouldn’t have posted requests for specific help in this thread; I’m simply suggesting that, if you try to look here from the point of view of a complete newbie to fora, you might understand that newbies could still be bewildered and unsure of where/how to post to get help.
Hello everyone, i’ve been over this topic for days now and i haven’t got a solution im configuring a pbx with asterisknow and im using a SPA3102 linksys to make a SIP trunk for my PSTN and use the FXS port for a fax machine/sip extension, the problem is with the incoming on the PSTN and the dialing on the FXS
i can easily dial out with my pstn as a sip trunk and i can call easily to my sip extension configured on the FXS port of the SPA3102 but all my calls going in my PSTN are not answered, and when they are answered all i get is silence over the pstn caller.
and on the FXS, i can call the extension but when i pickup the phone to dial i have no dial tone!!!
i have and outbound and an inbound route configured for the pstn just like a tutotial i found at a forum like this, i’ve read like 8 diferent tutorials and none have worked.
so please if anyone can help me!!! im using the latest asterisknow 1.7 avalible on the page for x64
i followed letter by letter this post http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx
and still no solution.
Did you read what this post is about? It gives instructions on how to ask for help. How could you ignore the instructions and post in this thread?
Patience and learning is key to using Open Source.
Our small business is currently running “trixbox Pro EE v4.1.2-p25,” and is supported by our VOIP provider. We plan to switch to ‘FreePBX 2.9.0.7,’ but our vendor will not support it, nor do they have documentation on it for configuring our trunk. I have the config for our current setup, but the distro we want to move to (FreePBX 2.9.0.7) has a totally different config interface. What is the best way to go about this?
FYI…
AsteriskNOW 1.7.0 and FreePBX 2.9.0.7 (to be exact)
Unreal, do you see what this thread is about? How could you possibly hijack a thread that specifically discusses hijacking
From my understanding one server serves multiple phones and that server cannot have more than one ip address. Meaning in the configuration there is only one way to setup the server, which is setting up a single server. I haven’t seen multiple server configuration variables. Sorry…
btw: why do you need 2 servers? I server can serve unlimited amount of phones and it doesn’t matter where it resides, you’ll need to enter the server IP/SIP IP in the phone and that should wrap it up.
Don’t work with your current VoIP Provider if they don’t support it and applying patches here and there will not work either and it will break.
I got my systems working seamlessly with asterisk/freepbx latest version with SIP Station service/VoIP Service and loving it.
PS: Don’t forget to issue yum -y update; it will grab all the updates for asterisk and its dependencies.
i install Centos 6 Desktop 64 bit and freepbx and all applications.
but when i open the page
it give me
Not Found
The requested URL /admin was not found on this server.
Apache/2.2.15 (CentOS) Server at “XXXX my Server IP” Port 80
what i can check i think i was have problem in php.ini , i could not change the time zone becouse when i command vi php.ini i find the file empty even everything is installed ?
kindly need your support
Did you even read the top of this post on how to provide needed information on your system so we can assist? You would be better off starting a new thread rather than hijack this one.
Hello everyone, my first ever post here.
I have read this thread to get clue on how to post here, such as to know the rules this place gives, and I got stunned by all this people trying to get help posting completely_off-topic questions right here!
So I start my trip here with a question to moderators: I think this thread doesn’t need answers, so what about closing it? It’s a reference after all, not a discussion.
Thank you, see you around.