Displaying 20 results from an estimated 1000 matches similar to: "Asterisk 1.4 and Fax"
2009 Oct 20
1
OutCALL
Hi everyone,
Does anyone have the documentation for OutCall?
http://code.google.com/p/outcall/
The link isn't working.
Thanks
Dan
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2009 Oct 18
7
Asterisk Monitoring
Hello,
I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls.
Many thanks
Dan
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2011 Mar 17
1
Status of Queue Members
Hi,
I'm trying to work out an issue with call queues.
I need the calls that are in a queue to be kicked out if all members are unavailable (for example if all SIP members are having network problems).
I tried leavewhenempty = yes but that only seems works when all queue members specifically log out of a queue.
I've looked at autopause, but we need it to automatically un-pause once it
2011 Apr 12
0
No subject
a phone system, and plug it into a SIP Adapter like the PAP2T.
Never done it myself, so I can't recommend a suitable intercom. Hopefully s=
omeone else can.
Dan Journo
Kesher Communications (UK)
Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<h=
ttp://www.keshercommunications.com/hostedpbx.html>
2011 Apr 11
3
changing port 5060 to 5061
please send me the ways to change asterisk port from 5060 to 5061 i
need to configure it because we are already using 5060 port in router
then we cant use it again we have to configure other sip server so
please suggest me a way..........................
On 4/10/11, asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com> wrote:
> Send asterisk-users mailing
2011 Apr 06
4
Call recording - methodology
Hello Everyone;
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I
have to modify it to make it easier to use, I do not mind.
Does anyone know of any opensource or otherwise solutions out there that
I can try out?
Thanks much.
Glen
2011 Aug 02
3
MixMonitor and attended transfers
Hi
I'm using asterisk 1.8.3.2 (with a couple of patches)
I have the following scenario...
SIP call comes in and gets answered by extension A (MixMonitor is
executed as part of this inbound dial plan of the number being called)
Extension A puts call on hold and calls extension B
Extension A then does an attended transfer of incoming call to extension
B
I'm finding that the recording
2014 Feb 12
1
Gigaset R630H and Asterisk
Hi,
Is anyone aware of an issue with Gigaset DECT handsets (R630H and N510P) and Asterisk?
A client has them, and whenever they try a blind transfer, something goes wrong.
Agent 1 starts and completes the blind transfer.
Agent 2 answers the transferring call.
Agent 2 hears asterisk music on hold, but the caller can hear the agent.
Any ideas?
Thanks
Dan Journo
Kesher Communications (UK)
2010 Oct 16
6
Remote Unix Connection
Hi,
Does anyone know where this is suddenly coming from?
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
Thanks
Dan
p.s. sorry about the last post. hit the mouse by mistake and it sent the email.
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2010 Feb 14
3
Asterisk Redundancy
Hello,
My host just had a faulty power supply and therefore, my Asterisk server was down for 7 hours.
It was a Sunday so no one was making calls, however if it happened during the week, I'd have problems.
I was trying to find a whitepaper or advice on how to set up two Asterisk servers to provide some redundancy.
I've been googling "asterisk redundancy" but all I've found
2009 Oct 14
3
Extension Paging
Hi,
We have SPA921 handsets which apparently support Paging, however i can't
find any information on configuring Asterisk to make a page call.
Does anyone have any information on Paging?
Many thanks
Dan Journo
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2011 Mar 09
4
doorphone?
Hi,
could anybody suggest a usable doorphone and magnetic door opener
"hardphone" system for me, please? Of course should be connectable to
asterisk. I am in the EU, should be available here.
thank you,
Csaba
2010 Sep 14
9
Random File Name
Hi,
Im looking at using MixMonitor to record calls and I know that I need to set the filename first.
However, with the number of calls coming in, hard coding the filename isnt an option.
So I need to do something like this:-
MixMonitor(RANDOMNUMBER.wav)
But can't find a way to generate a random number.
I thought that maybe I could use a unique variable that already exists for the current
2011 Apr 08
2
Call Recording using MixMonitor - close, but would like some more words of wisdom.
Dan et al;
This looks like a perfect solution.
However, I have one issue. If I initiate the macro manually (put it in
the proper context/dialplan) it works. I see the *.wav file being
created and growing in the /var/spool/asterisk/monitor directory.
If I try to implement it adding the MixMonApp =>
*1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I
cannot get it to
2009 Nov 03
3
Problem with ChanIsAvail
Hi all,
I am having a problem with ChanIsAvail. It always returns the same
result, regardless of whether an extension is available or not.
It always returns 0 Unknown Status.
This is my dialplan.
exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s)
exten => _2XX,2,Verbose(0, ${AVAILSTATUS})
exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5)
exten =>
2009 Oct 14
8
Asterisk in the Cloud
Hi,
I was wondering if anyone is successfully running Asterisk in a cloud
environment.
If you could state which cloud you are using, I'd appreciate it.
Many thanks
Dan Journo
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2010 Nov 03
5
ADSL Load Balancing
Hi,
I've got a client with two ADSL connections for redundancy.
Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections?
Or to use one connection as the main one, and automatically fail over if the first connection drops?
Or does this kind of thing need a serious network switch?
Thanks
Dan
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2010 Aug 24
8
Include and Realtime
Hi,
I think I already know the answer to this question, but is there any way to do the following using realtime? Or do I have to create a full dialplan for each client without using includes?
[client1_phones]
include => client1_internal
include => client1_outgoing_calls
include => test_calls
include => parkedcalls
[client2_phones]
include => client2_internal
include =>
2009 Dec 06
3
Call Limits
Hello,
I'm trying to figure out how to limit the number of concurrent calls a client can make.
I have a client that has 6 SIP accounts. One for each SIP phone.
I want to limit it so that they can only make 2 outgoing calls at a time so that I can bill them "per channel" rather than "per extension".
A separate (but not so important) issue is that I want them to be able to
2010 Oct 13
11
DMTF Mode
Hi,
Which DTMF mode do people mostly use?
I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user.
So if I call a company that has a menu system, I can't use the menu.
Thanks
Dan
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