[Asterisk-Users] Help on directed Call Pickup

[Cristian Manoni]

Is Directed Call Pickup supported in asterisk? 
(http://voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup)

The group call pickup works well, but I can't make directed call pickup:
example:
Someone is calling 202 (my phone, GS BT101), i stay near ext. 230 (Cisco 
7905g), i want capture this call:
i compose the pickup ext (*8)  followed by my ext (202); after composed 
*8202 i hear busy tone and 202 continue ringing

I searched in wiki and mailing lists but there is no more documentation
Can someone help me?
Thanks
Cristian 

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[Asterisk-Users] how to force agent logout

[Cristian Manoni]

Hi all
in my callcenter i have 10 agents that make the agentlogin and wait the
calls with the strategy leastrecent earing music.
sometimes the agent logout fail when an agent hangup the phone to make a
break.
The result is that the agent after the break can't log in again because is
alreadyon

the only solution i find to permit the agent login again is to restart
asterisk

The problem appens with asterisk 0.7.2 with app_queue patched.
the bug is http://bugs.digium.com/bug_view_page.php?bug_id=0001039

my question is:
how can i log out the agent manually to permit he can login again?
There is a command line way to logout a single agent?

Thanks
cristian

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[Asterisk-Users] Choppy Problem!!

[Cristian Manoni]

Help me
i'm managing a call center with asterisk, GS 102 and diva server 4 bri.

i have big problem with big choppy sound, In the direction External
user --- Agent

Please aid me!!!


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[Asterisk-Users] Asterisk Crash

[Cristian Manoni]

Hi all
i have an asterisk crash every time the number of channel CAPI arrives at
490
CAPI[contr1/541774629]/490.

Feb  2 11:36:37 WARNING[10251]: Unable to allocate socket: Too many open
files
Feb  2 11:36:37 WARNING[10251]: Unable to create RTP session: Too many open
files
Feb  2 11:36:37 WARNING[541717]: Failed to create pipe: Too many open files

after this log asterisk shut down.

Is this a bug?

Help Me!!!

Cristian

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[Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

[Cristian Manoni]

Hi All
i have continuos error:
Unable to handle DTMF tone 'f' for 'SIP
on the asterisk console.
after this the call hang up.

I have a BGT 101 that make and receive call from the capi channel

Thanks
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[Asterisk-Users] CLIR and isdn4linux

[Cristian Manoni]

hi 
I have a passive isdn port configured in modem.conf
in extention.conf i use this two channel (ttyI0 and ttyI1) with the string:
exten = _NX,1,Dial,Modem/g1:${EXTEN}|60|r

how can i hide my msn?
is it possible to activate the clir with the @ before the ${EXTEN}?
 
thanks
Cristian
 
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