easyAsterisk.org :: Forum

Call from Computer to a FritzBox 7112 [by paule22]

Sat, 27 Mar 2010 15:07:52 +0200

General:: Call from Computer to a FritzBox 7112
Hello,

is it possible to make a call with zoiper softphone from on PC to a FritzBox with a wireless network connection?
How can I do it?

Thanks
paule22

Re: Question about lines into [by admin]

Thu, 29 Oct 2009 08:46:07 +0200

General:: Question about lines into
Hello,

All incoming calls are routed to the destination defined in "Attendant Console", except those for which you have configured a DID.

Re: How to set outbound route to dial-out [by admin]

Fri, 11 Sep 2009 07:27:21 +0200

General:: How to set outbound route to dial-out
Quote:

robert wrote:
i have read the parts of trunk and outbound route in the user manuel several times, and i did exactly according to user manule,but i can not dial-out,when i uphook the phone and press the number i want to call,asterisk displays hangup.However, i can get incoming calls.

in custom contexts,i add from-internal
here is my dialplan:
exten =>_X.,1,Dial(dahdi/5/${EXTEN})
exten =>_X.,2,Hangup()

Hello,

In order to use dial out functions integrated in easyAsterisk you don't need to configure custom contexts. You have to proceed as follows:

- Configure hardware settings if you use dahdi or mISDN adapters.
- Configure trunks.
- Create a new outbound routing rule and set a dial-out prefix.
- Configure routing rules (patterns) for the outbound route, remember to assign one or more trunks for every pattern you create.
- Configure client permissions for this route ("allow all local extensions to use" is ok and allow all local phone extensions to use the route).
- If on the left column of easyAsterisk interface you see a red button asking you to reload or restart the asterisk server, press it.
- Now you can use dial out using the route just created. Remember to add the dial-out prefix to the extension you want to dial. For example, if your dial-out prefix is "0" and you need to dial "1234567", you have to digit "01234567" and press dial from the phone.

Re: Find Me Follow Me [by admin]

Tue, 16 Jun 2009 09:59:57 +0200

Future Developments:: Find Me Follow Me
Quote:

rryding wrote:
Has any though been given to implementing a find-me follow-me function in the user section?

thanks ross

Hello,

Thanks for your suggestion, we'll consider implementing follow me in future developments.

Re: Configuration of Easyasterisk [by admin]

Mon, 23 Feb 2009 11:08:28 +0200

Administration:: Configuration of Easyasterisk
Quote:

-- Executing [s@macro-queue:2] Answer("mISDN/7-u908", "") in new stack
-- Executing [s@macro-queue:3] Wait("mISDN/7-u908", "1") in new stack
== Spawn extension (macro-queue, s, 3) exited non-zero on 'mISDN/7-u908' in ma cro 'queue'
== Spawn extension (macro-queue, s, 3) exited non-zero on 'mISDN/7-u908'
Econet*CLI>

It seems you correctly receive calls from isdn but there is a problem with the queue configuration. Please try to configure your attendant console to point directly to your phone extension and try if it works.

Re: have somebody used trixbox ? [by admin]

Wed, 24 Dec 2008 11:01:02 +0200

General:: have somebody used trixbox ?
Quote:

yx85107453 wrote:
nobody used trixbox ?

Hello,
Trixbox integrate freepbx while easyAsterisk use a proprietary web interface. easyAsterisk web gui is very easy to use. You can try both and choose what you like.

Re: Problems with incomming calls [by admin]

Wed, 10 Sep 2008 09:25:58 +0200

Installation:: Problems with incomming calls
Quote:

micbal wrote:
I can call and be called by the analog number.
The VOIP account can call out but can not be called.

Please check your sip trunk configuration. You need to set the "type" as "friend" and enable registration. If you execute "sip show registry" in asterisk console you can verify if the server is correctly registered to budgetphone. Remember also to set correctly "Attendant Console".

Re: Registering SIP phones [by arian]

Wed, 13 Aug 2008 10:13:06 +0200

Administration:: Registering SIP phones
Username and secret are available only on dynamic host configuration.
Go to each extension and set host on dynamic, change username and secret as for phone configuration.
On Linksys SPA 922, select Admin Login, Ext1, UserID and password in Subscriber information section.
Don't forget to reboot the phone.

Re: Return values from Perl to asterisk [by sujith]

Sun, 10 Aug 2008 05:02:19 +0200

General:: Return values from Perl to asterisk
Hi all, I'm stuck with how to pass the return value to asterisk. That is, a value has to be passed from Perl script to asterisk. The variable returned in Perl has to be passed to asterisk, where if the return value is 1 from Perl , i have to do something and if return value is 0, i have to do something else. I have written code like this.
--------------
sujith

Re: does Aeasyasterisk works with A2billing [by admin]

Thu, 31 Jul 2008 07:23:52 +0200

General:: does Aeasyasterisk works with A2billing
easyAsterisk does not include A2billing integration. You can install it manually including sip and iax configuration files used by A2billing from "General Settings => SIP/IAX2 Protocol => Advanced" menu.