Displaying 16 results from an estimated 16 matches for "adlibris".
2009 Jan 20
3
Forwarding calls and trasfer calls
...forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions.
I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf.
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2009 Jan 26
5
Start asterisk on boot
...the system is
# started until it is shut down again.
description "Asterisk daemon"
start on runlevel-2
stop on shutdown
respawn
exec //usr/sbin/asterisk -f
But it doesn't work.
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2009 Jan 08
2
Problem incomming from openser
...; to extension '0840303395' rejected because extension not found.
If I wait approx a minute and try again, the call will go trough.
We don't use REGISTER or anything like that.
What can be the problem
regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2009 Jan 13
2
404 not found from one ip-adress
...:16869 handle_request_invite: Call from '' to extension '0840303390' rejected because extension not found.
And the provider get an "404 not found" error on their side.
What can be the problem?
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2009 Feb 10
1
unistim and transfer calls
...calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error?
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2009 Mar 05
1
use more then one sip-provider to dial out
Hi
I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6?
/ralf
Ralf Tr?skman, IT
AdLibris AB, Box 3667, 103 59 Stockholm.
Bes?ksadress: Sveav?gen 56C, 111 34, Stockholm - Obs ny address!
Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99
ralf at adlibris.com<mailto:anders at adlibris.com>, www.adlibris.com<http://www.adlibris.co...
2009 Feb 05
1
musiconhold realtime queue
...time mysql. I am trying to set another musiconhold then "default" but I cant get it to work,
I have an musiconhold entry in my queue_table, but don't know what to put in there and where to put the file.
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2008 Dec 03
1
Asterisk user client for customer service
...up, like customer service can use?
We have today an avaya IP-office with phonemanager pro and I want something equal to phonemanager pro, where you can logon to ques and see how many calls is in that queue and so on.
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707458074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2008 Dec 04
2
set monitor_filename
...LENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID})
-- Executing [s at kundservice:1] Set("SIP/0850001175-b7942770", "MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12")
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707458074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2009 Jan 14
1
gxp2000 and no sound asterisk 1.6
...can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me.
How do you configure the grandstream 2000 to work on asterisk 1.6?
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2009 Feb 17
0
unistim channel problem
...ll worked yesterday.
I have the unistim module loaded.
Could it be that I have set keepaliave in unistim.conf to 500, I had to do that outerwise my phones would show server unreachable after approx 2 minutes.
What can I do?
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2009 Feb 19
0
sip phone cant hear the caller
...ne that calls me can hear me but I cant hear them, when I make the call I can hear them.
Im running asterisk 1.6 behind a firewall, I have port 10000-20000 for rtp and 5060 for sip forward to my asterisk.
Any tips?
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2009 Feb 23
0
problem with nortel 2002 disconecting
...ip2002 phones connected to asterisk 1.6, the problem I have is that the phone looses the connections with the server and then drops calls, we can reconnect but the customers don't like it.
Anyone has the same problem?
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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g
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2009 Feb 05
2
no need to dial areacode
Hi
To dial an outside line i have to dial 0. I want to have that when we dial local numbers, that is we are in the 08 area, I don't want to have to dial 08, how to set this up in asterisk 1.6?
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
ralf at adlibris.com<mailto:ralf at adlibris.com> www.adlibris.com<http://www.adlibris.com/>
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2009 Mar 03
1
Master.csv missing dstchannels values
Hi,
I'm currently building an report query for Asterisk (based on the Master.csv file).
In some rows there are missing values for dstchannel where lastapp is "Queue" and lastdata equals the queue name I'm interested in.
Pleas guide me, are the rows including the missing dstchannel values equal to calls hanged up by the caller in queue and never bridged to a agent?
Many
2009 Mar 04
0
Master.csv - disposition value (based on?)
I need to get the effective time for a call and therefore I wonder if the disposition field in the Master.csv are based on the effective call time with an agent or does this value also including the callers holdtime in queue?
Many thanks!
Regards
Tobias Ste?n
_____________________________________________________
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