In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I have server with a total of 6 Analog ports...using TDM04B and TDM02B.
I have 3 Lines that are DIDs and 3 are Main/Roll Over lines and I have
worked through getting the DIDs to work and route to the
extensions...now what I need to do is
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Thank you, but this is how I see your mail. How can I see it right?
http://lists.digium.com/pipermail/asterisk-users/2006-February/146742.html
Thank you!
--
Tomislav Parcina
[EMAIL PROTECTED]
You can do this with agents, no need for a queue.
Define agents in agents.conf
In your dialplan, instead of Dial(SIP/bedroom) use
Dial(Agent/200)
Let the phones login as agent :)
OK, I know I have to Dial(Agent/200), but how will I login agents if I don't
use queue? If phone log's in as
AMP doesn't do miracles! Look at its dialplan.
I believe he doesn't, but I don't have AMP installed. Next week I think I'll
have enough free time to try it. Will [EMAIL PROTECTED] do the trick?
--
Tomislav Parcina
[EMAIL PROTECTED]
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In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
We have got some ATA for only $55 if you are interested?
Sam
Yes Sam, I'm interested. If they work with FAX I'll definitely buy one of them
for testing.
--
Tomislav
[EMAIL PROTECTED]
___
Since you have no Digium hardware (and thus no connection to POTS or
PRI)... are you routing your phone calls via VoIP? If so, it is not
recommended to run FAX via VoIP. The two don't mix. FAX is not able to
handle packet loss like VoIP. Also, any codec other than uLaw will not
even come
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
AFIK, fax is supported and installed with with app_txfax app_rxfax
If this proves to be true why would you need the ATA?
I'm working on this one. I have to install app_rxfax but I have failed. Soon,
I'll try again (hopefully next week).
Why don't you simply rotate the logs with logrotate ?
How to do that?
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Tomislav Parcina
[EMAIL PROTECTED]
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In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
This can easily be accomplished with AMP using the Users and Devices mode.
http://voipspeak.net/index.php?/content/view/49/28/
How can this be done without AMP? Using personal queue's and agents? I need
information's to get better
Subject: RE: SIP Register
From: Tomislav Parcina [EMAIL PROTECTED]
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
First impressions telling me you want to check your phone settings. What
phone are you using and what are the config settings?
Hi Mark, thank you for your reply.
I'm
Subject: RE: Queue - check agent
From: Tomislav Parcina [EMAIL PROTECTED]
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Hello,
I might be wrong here, but I thought that in Queues.conf, if you defined a
queue with joinempty=no, or joinempty=strict then no calls will be placed in
Why do you think it's phone problem and not Asterisk? Asterisk is the
one that contents my provider. * is the one who should decide what
information's to send to my VoIP provider... Anyway, I'm inexperienced
with this and I'm just trying to understand what is happening and where
could be
When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why?
When a phone call isn't from queue then att transfer works fine.
In features conf I have *1 for recording, *2 for att transfer and #1 for blind.
In queue blind transfer works. For disconnect I have #0.
I guess that
I'm having trouble making calls over my VoIP provider. I do successfully
register, and when I try to establish a phone call Asterisk sends wrong
username and password. Instead of sending username and pass that I have
provided, he send username and pass of the SIP phone that is registered to *
Hi list!
How to send a call directly to voicemail recording?
When I put this
exten = 313,n,VoiceMail,u221
Or this
exten = 313,n,VoiceMail,b221
In my dial plan, calling person first hears greeting message (busy or
unviable). I would like to avoid greeting message (I would play something with
with a call.
Thank you for your time.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
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When I try to make att transfer (*2) of call that was in queue the call get's
disconnected. Blind transfer (#1) works fine. In dial plan I don't have any h
or H (hangup call with *). In features.conf I have this line disconnect = *0.
What could be the reason why call hang's up?
--
Tomislav
,Katarina Ivanisevic
agent = 404,404,Marija Bilic
agent = 405,405,Ana Kaliterna
Will this work? Will agents 401 and 402 be in both groups? If I join
every group to another queue, will one agent be in both queue's?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e
$
Thank you for any suggestions.
P.S.
If this is second time you see this message, then sorry for resending, but I
didn't see it on list...
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
)
Do I have to add one more line (311,n) which will define what will
happen with call if no agents are logged in that queue? Can I do that on
any other way?
Thank you for your time!
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
with GE have Gigabit Ethernet port.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
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Asterisk-Users mailing list
will use for double
check of CDR and billing.
Any information's are welcome.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
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With asterisk-1.0.9 version when I wanted to enable logging of CDR in MySQL I
needed to make softlink
ln -s /usr/src/asterisk-1.0.9/asterisk.h
/usr/src/asterisk-addons-1.0.9/asterisk.h
Now, on version 1.2.0 I don't have that file. Do I need it?
Thank you for your time.
--
Tomislav Parčina
Are there any Asterisk Users Newsgroup? For me it's much easier to follow
newsgroup then to read all e-mails. Especially with news readers with so many
features.
So, if anybody knows for any newsgroup that has big discussions about Asterisk,
please let me know.
--
Tomislav Parčina
Lama
this one.
I would appreciate any hint or help.
Thank you for your time.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
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allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=h263p
allow=h261
canreinvite=no
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
done something wrong or is there I haven't done? Please help.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
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is in firewall.
Thank you.
Tomislav
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tomislav Parčina
Sent: 10. studeni 2005 9:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP NAT register
I'm unable
; for SIP2 dial on FXO2... I hope I have make my self clear :))
Thank you for your time.
--
Tomislav Parčina
Lama d.o.o.
www.lama.hr
tparcina#lama.hr
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Asterisk-Users
Soner,
Thank you! This is what I needed.
Tomislav
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Soner Tari
Sent: 4. studeni 2005 12:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Every SIP on
load...)
Other things that I will welcome
- operators panel
- voicemail (to listen your voicemails)
Thank you for your time.
--
Tomislav Parčina
Lama d.o.o.
www.lama.hr
tparcina#lama.hr
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Thank you, this is definitely an option. Right now I'm trying to make something
work on my Linux installation (FC4). And I like to install as much things on my
own, so that I really can se how that stuff works.
Tomislav
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Does anybody have detailed instruction how to Install AMP? I have tried to
install it using Installation Guide on their pages but I'm unable to satisfy
AMP's PERL module dependencies.
Thank you for your time.
Tomislav
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