So you have a problem and want help?
You probably just posted a question (or are planning on posting one) on the forum and are looking for a kind person to give you a hand solving your issue... But before you do consider this.
when you state you have a problem think like a technical person please. How would you answer this question if somebody you didn't know asked you the same question? Do you have enough information to work with to start?
60% of all questions posted here are missing the all important details at the start, so most times the first reply they get is to give more information...
What kind of information is needed?
We will want to know the versions of everthing...
FreePBX, asterisk don't answer the latest as that is not good if somebody 6 months from now reads this trying to fix the same issue.
is your system one that you made from a disto build? if so who's and version you used/upgraded to.
Not a distro build then it's a hand built system. We'll need to know who's OS, what version, who's directions did you use to build it (URL please), did you follow them to the letter, if not, why not, and what didn't you do if any thing? Also very important did you install anything extra? some times when you add something extra to a install it might ahve changed a default you didn't know about and that can cause huge hidden issues.
The reason we ask for all of this information is it helps determine if you are really using the latest current version (updates could have come out at 3am and either you or your helper didn't know) or maybe you are bringing up a old issue that might have already been addressed in a later version.
With hand built systems it is very, very common to have permissions issues if the directions are not followed to the letter or did you do something different then expected like didn't make Apache run as the asterisk user for example.
Some distro's these days don't provide the same versions that are available directly off this site (and at the same time not make that clear to the installer either). Some distro's have gone so far as to have "forked" things (thus screwing things up for their own purposes) changing all kinds of things like version numbers, mirror sites, not including all the modules you'd expect if you used a real copy from FreePBX. Some distro's will pre-install certain modules while others will not.
So please remember that when you asking for help to provide the details needed at the start, that way you can get your problem addressed quicker then without the details and the next thing we do is ask for them.
__________________
Technical Corner
New Forum Topics
- Upgrade from 1.6.0 to 1.6.2 from source
- Language problem in queue
- CDR Report wildcards?
- CDR Report wildcards?
- Calling system on IAX thrgough Asterisk to phonesystem on SIP.
- System needs a reload every day to work.
- Symlink Problems with Trixbox 2.8
- * key not accepted on oubound calls
- Queues No Longer Utilizing Follow-Me Settings after 2.8 Upgrade
- Extensions appears ON but after 20 sec they go OFF in polycom 601
- Cheat Sheet for User Interaction
- Permit and Deny fields
Feel free to use anything
Feel free to use anything you'd like from my "dentists" post over at the trixbox forums if you think you can make use it. Last I saw it was stickied in the "help" forum.
I remember that post, for
I remember that post, for those interested look here http://trixbox.org/forums/trixbox-forums/help/we-are-not-dentists-we-are.... Yes many have a dislike for trixbox but that post covers it very nicely.
My favorite issue is people who bump there own posts. Normally the reason they don't get answers are because of one of two reasons:
1. The question is so open ended, or lacking details nobody wants to touch it or a 2 second search using google will find the answer.
2. You have a very small nitch question and it will take time for somebody in the know to see and respond to it.
One thing I would have added...
And if, despite the above advice, in one of the replies you get someone asking you a specific question (or making an assumption - the implied question is, "Is this assumption correct?"), ANSWER THE QUESTION. Don't go off onto a tangent about a dozen other things and then forget to answer the question, or give a vague or ambiguous answer to the question. If you don't know the answer (or don't know how to obtain the answer) then at least reply and state that. If you don't think the question is relevant then say why you think it isn't - the person who asked the question may choose not to help you anymore, but others might (particularly if the question truly wasn't relevant).
Usually when someone asks a question they are trying to narrow down the possible causes for your issue, or maybe explore your options for fixing the problem. It makes it a lot harder when you don't answer the question, and the "helpers" tend to wander away to other threads, to help people who will answer the questions posed.
Having said that, that does NOT mean you are obligated to try every piece of advice you are given. It is YOUR system, after all, and if something doesn't "smell right" about a piece of advice, it might not be. Not everyone who attempts to answer questions in these forums is an expert, and some people give out good advice while others freely dispense very questionable advice (as in, it might solve the problem for the moment, but cause other problems later on) - and none of us are experts on everything!
agreed!
i used to work in a support office and i hated those "my computers not working please call" requests where all that was wrong was their e-mail wouldn't open!!!!
thanks for the post, i hope i was detailed enough in my help post! http://www.freepbx.org/forum/freepbx/users/freepbx-and-sugarcrm
sorry wrong place
sorry
Setting up Server to Server over the internet
I currently have 2 Asterisk servers one in California and one in Washington.
AsteriskNOW 1.5
My question is or if I could be pointed in the direction.
How do I set them up to work together?
Thank you
MBB
Wow bmoyce did you read any
Wow bmoyce did you read any of this thread or simply hijack it with your question.
Suggest before posting again you google for the many articles on linking Asterisk/FreePBX based systems.
setting up x-lite with AsteriskNOW 1.5
I installed asterisknow 1.5 which has centos 5.3 final version. I setup two extensions on the server. On another system I installed x-lite and gave it the username and secret of one of the stations. When I dial the other extension, it takes a while with the message that it is calling and then it comes back with message to try later. How should I resolve this problem, Iam new to this asterisk staff. Your help will be appreciated.
ubuntukips, Please re-read
ubuntukips,
Please re-read the above postings. Like the one directly above yours.
after I installed freepbx 2.5.1, all module in web interface wer
after I installed freepbx 2.5.1, all module in web interface were gone after reboot the server. I use asterisk 1.6.0.1, CentOS 5.2 and freepbx 2.5.1
Please help me to solve it....
tithtola, So you could not
tithtola,
So you could not read either? Please go to the top and re-read the whole thread.
You couldn't make it up
You couldn't make it up
IVR Dialing group etc missing
I am very new to Asterisk. I have downloaded Astersiknow-1.5.0 ISO and set it up in a system. I have added two extensions and tested with x-lite. It is working well. The problem i am facing is , in the FreePBX admin menu, i can't see the options like IVR, Follow me, Conference , Call Back etc. How this could be enabled. Any help will be greatly appretiated
rajesh_amma, Did you even
rajesh_amma,
Did you even read any of the messages above?
Do not hijack a thread about one thing with another it is the fastest way to NOT get any help.
Got Resolved
Really Sorry Mr. Fskrotzki,
I got resolved the issue with the help of Module Admin
Hello All Please i need your
Hello All
Please i need your help
i want to initiate a session through my small network ( 1 linux + 2 LAptops with soft phones )
so i add 2 extensions but it not worked
so
can you tell me how to configure the SIP protocol
the follwing errors i had in the main page of free PBX :
retrieve_conf failed, config not appliedReload failed because retrieve_conf encountered an error: 1
Added 2 days, 16 hours, 36 minutes ago
(freepbx.RCONFFAIL)
and :
Asterisk Manager Connection FailureFailed to connect to the Asterisk manager through port: 5038
Added 2 days, 16 hours, 36 minutes ago
(retrieve_conf.FATAL)
Sami - You have to be
Sami - You have to be kidding? You did not read a single entry in this post before hijacking this thread?
I am dumbfounded did you scroll to the bottom and ignore all the instructions on how to ask for help?
This is a sure way to get ignored.
this is not a nice way to
this is not a nice way to reply
when some one needs help
he has the execuse to put his comment whereever he wants
because he needs help
Any way thank you for your help !!!
Quote: this is not a nice
This entire thread is about how to ask for help and where to ask for it.
There is no "excuse" for putting your question where it does not belong and not providing the information requested. It is, as pointed out earlier in this thread rude.
If you post with the information requested in a new thread someone will help you. As is stands posting in the wrong thread with insufficient information will get you no assistance.
Quote: when some one needs
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.
Robert Keller - VoIPologist
FreePBX Training Opportunities - Click Here
Get Official Paid Support - Click Here
sami.alio, If you can't read
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.
AsteriskNOW
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
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.
LOL
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
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...
Help on FreePBX
I installed follwing that:
http://www.freepbx.org/support/documentation/installation/install-proces....
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
yet another one who can't read...
can't make outbound calls
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
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!!!!!
Mikael Carlsson, FreePBX Development Team
Get Official Paid Support - Click Here
Why people don't read
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.
i need help!!! asterisknow 1.7 x64 and a SPA3102
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-31...
and still no solution.
Quote: magnumve Did you
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.