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Re: Question about lines into
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Hello,

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

Posted on: 2009/10/29 9:46
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Re: Question about lines into
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Quote:

I use a card with 4 analog inputs of the lines entering

On each port has a different line with different number
My question can I control if the line number 1 ring I want to refer it to - a particular IVR
Line 2 rings and then refer it to - a certain extension
Line 3 rings and then refer it to assume a voice box

In conclusion, my question Is it possible to control the routing of all incoming lines
on its own line and line


Hello,

You can route incoming calls using DIDs ("Calls Managemente => Direct Inward Dial (DID)" menu). To do this you must have callerid enabled and working.


Posted on: 2009/10/28 9:51
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Re: How to set outbound route to dial-out
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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.



Posted on: 2009/9/11 9:27
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Re: Find Me Follow Me
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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.


Posted on: 2009/6/16 11:59
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Re: Configuration of Easyasterisk
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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.


Posted on: 2009/2/23 12:08
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Re: Configuration of Easyasterisk
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Quote:

nazrul_70 wrote:
Econet*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
604/604 (Unspecified) D 0 Unmonitored
603/603 (Unspecified) D 0 Unmonitored
601/601 192.168.2.3 D 46008 Unmonitored
600/600 192.168.2.5 D 23206 Unmonitored
100/100 (Unspecified) D 0 Unmonitored
5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 3 offline]
Econet*CLI>


Ok, now 601 extension is correctly registered.


Posted on: 2009/2/23 12:03
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Re: Configuration of Easyasterisk
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Quote:

nazrul_70 wrote:
Econet*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
604/604 (Unspecified) D 0 Unmonitored
603/603 (Unspecified) D 0 Unmonitored
601/601 (Unspecified) D 0 Unmonitored
600/600 192.168.2.5 D 23206 Unmonitored
100/100 (Unspecified) D 0 Unmonitored
5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline]
Econet*CLI>


As you can see your 601 extension is not registered to asterisk. Please check this configuration.


Posted on: 2009/2/23 12:03
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Re: Configuration of Easyasterisk
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Hello,

Quote:

-- Executing [s@macro-stdexten-vmbu:15] Dial("mISDN/7-u0", "SIP/601|30|to") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-stdexten-vmbu:16] Goto("mISDN/7-u0", "s-CHANUNAVAIL|1" ) in new stack
-- Goto (macro-stdexten-vmbu,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-stdexten-vmbu:1] VoiceMail("mISDN/7-u0", " u601") in new stack
-- <mISDN/7-u0> Playing 'vm-theperson' (language 'en')


You receive correctly the call from your ISDN Line but is seems your 601 local extension is not online so Voicemail starts. Does this extension is correctly registered to asterisk PBX? Please type "sip show peers" in asterisk console and post the result.

Posted on: 2009/2/4 14:10
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Re: Configuration of Easyasterisk
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Quote:

nazrul_70 wrote:
Hi admin,
Telco advised NT mode and Point to Point.


You need to configure your adapter ports as TE Point to Point. Please post console log when you receive a call.

Posted on: 2009/1/14 9:26
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Re: Configuration of Easyasterisk
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Quote:

nazrul_70 wrote:
Hi admin,
I have follow as your advised.now i can make local extension to extension call.
i am unable to call out from extension to mobile through ISDN BRI Line.


Please check if the configuration of your ISDN adapter (point to point / point to multipoint) meets che configuration of your BRI Line (your telephon company should know this).

Quote:

Niether incoming call on the number which provided by Telecom.
I have some screenshoot,i have no idea how to upload for you to View.

what is the procedure to see call hit from Bri line for incoming call?


Login in console as root and enter in asterisk console typing:
asterisk -r

Please post your console log.

Posted on: 2009/1/12 22:58
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