Displaying 20 results from an estimated 21 matches for "lour".
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2012 Oct 05
2
SendFAX - multi-page TIFF
Hi,
Does anyone had the problem of asterisk SendFax + spandsp sending only
the first page of a multi-page TIFF file?
Seams to be related to spandsp ECM config.
Any thoughts about it?
Thanks,
Gabriel
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2012 Feb 02
2
externip nat audio sip trunk issue problem
Hi all,
I've tried search this problem on the list... no luck...
The case is:
without externip/localnet config on sip.conf [general] my SIP trunk works,
but with no audio NAT problem (asterisk sends the private 192 address to
the outside...)
when I configure externip/localnet correctly my SIP trunk simply disappear!
Checking the signalling with tcpdump shows me that Im sending the
2009 Jul 18
3
Count Available Queue members
Hi all,
Someone know how can I check for available members on a queue Before I
queue the call, so I can do something else with it? Note that is not the
case for joinempty
Thanks,
Gabriel Ortiz
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2009 Jan 17
1
canreinvite per route
Can I activate/deactive the canreinvite SIP flag on the dial plan?
The idea is to allow reinvite only for exten <-> exten calls, and not for
outbound calls
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2011 Feb 15
2
Dialplan end of pattern matching question
Hi,
I've noticed an unusual behavior on the dialplan execution: assume this
DP:
exten => _6XXX,1,NoOp(test1)
exten => _XXXX,1,NoOp(test2)
exten => _XXXX,2,NoOp(test3)
If I call 6000 then test1 and test3 NoOps get executed, even though the
pattern is different.
I've always thought that if I call 6000 it would match the 6XXX pattern,
that only has 1 priority, that would get
2017 Mar 22
2
Large astDB - millions of tuples - issues?
Hi all,
Does anyone uses astDB for a large amount of data, in special for
implementing black lists with millions of numbers (i'd like about 2 or 3
million)?
That would be held in memory right? Is this (memory consumption) the only
problem I could face?
Att.
Gabriel
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2009 Jan 16
1
Dialing from E1/T1
Hi,
A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN
trought another E1. When the legacy user dial to the PSTN the call pass
trought Asterisk.
All works OK, the only problem is the delay on the Asterisk server when it
receives the digits from the 1st E1 link. It will only make the call when
the digit timeout expires.
Is there a way to make something like
2009 Mar 31
1
Queues in memory after startup
Hi all,
After * starts the command "queue show" would not show any of the realtime
queues, but just the ones that are in the queues.conf file. In this state de
AMI would not send any "QueueMemberStatus" for that queues until a call is
received by that realtime queue.
Anyone knows any whay to load this information in *'s memory without the
need of the queue receiving a
2009 Aug 17
1
Goto mask
Hi all,
When I have 2 masks that would like to execute the same logic, there is
the way to use the Goto (or any other) command without changing the
${EXTEN}?
Eg. DID range is 1200-1349 -> call Macro(disca), what mask to use? (I just
got it with 2 masks, but I didn't wanted to duplicate the dialplan for both)
[test]
exten => _12XX,1,Set(DIR=3)
exten =>
2009 Sep 11
1
Voicemail by email with HTML
Hi all,
I'm trying to send an email with the voicemail details and I want to send
a HTML link on it to make a click2call to the voicemail main, but the email
is send with 'text/plain' encoding and thus it will not show the link, but
the HTML in plain text on the body of the email,
How can I change the enconding to 'text/html' so the link will get
displayed correctly?
2009 Nov 06
1
AMI Originate and Variable header
Hi all,
I'm trying to use the CDR() function on the "Variable" header of the
Originate AMI action, but it isn't working.
Anyone knows anything about this problem?
asterisk 1.4.26
Thanks,
Gabriel Ortiz
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2014 Oct 01
1
CALLERID(num) and CDR(clid) - originate
Hello,
A question on channel originating (call files and AMI Originate):
How can I change the CALLERID(num) var (because of the E1 provider
needs), but having another n?mber (the original one) stored on the "clid"
CDR field on the database?
A channel agnostic solution would be the best one, without having to deal
with the problem based on what type of Tech used for the outgoing
2010 Aug 10
1
Playback during call
Hi all,
How can I playback a file within an active call?
I've tried with ChanSpy whisper mode like this (using AMI):
Action: Originate
Channel: Local/9999 at default
Priority: 0
Variable: MSG=test
Application: ChanSpy
Data: SIP/1234-123
Async: 1
and in the dialplan:
[default]
exten => 9999,1,Answer()
exten => 9999,n,Wait(2)
exten => 9999,n,Playback(${MSG})
Where
2007 May 19
2
Ser vs. DUNDi
With all of the recent talk on the list about DUNDi, I have a question. From
the outset it appears that SER is often used for high availability solutions
and as a tool for almost clustering Asterisk boxes behind it. It appears to
me that DUNDi is providing a lot of this as well. Now I know DUNDi is not an
application by itself to proxy SIP requests but can I hear any information
out there that
2008 Jun 30
4
Rebuild of kernel 2.6.9-67.0.20.EL failure
Hello list.
I'm trying to rebuild the 2.6.9.67.0.20.EL kernel, but it fails even without
modifications.
How did I try it?
Created a (non-root) build environment (not a mock )
Installed the kernel.scr.rpm and did a
rpmbuild -ba --target=`uname -m` kernel-2.6.spec 2> prep-err.log | tee
prep-out.log
The build failed at the end:
Processing files: kernel-xenU-devel-2.6.9-67.0.20.EL
Checking
2009 Jan 20
0
channel var for Call on hold?
Hi all,
Does asterisk (I'm using 1.4.19) sets any channel variable with the holded
chan when it does an atxfer? I tried to see if it does on the source, but i
didn't find any clue, neither enabling console debug.
Thanks,
Gabriel
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2009 May 06
0
How to get SIP resposnse codes
Hi all,
I need to know the SIP response code from within the dial plan, someone
could point me on how to?
Gabriel Ortiz
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2009 Jun 08
0
remote queue members
Hi all,
I'd like to know the best way to deal with queue member that are reached
trough a SIP trunk. Let me explain:
1) "Master" asterisk box with my call queue
2) "Slave" asterisk box with a channel bank interface
the two boxes are connected trough a SIP trunk, and the dialplan in the 1st
box connect (for eg.) Local/0030 at gw to the 30rd channel on the 2nd box. All
2009 Jul 22
0
Attended transfer and 'pbx-invalid' - 1.4.26
Hi,
I've created a tiny dialplan to test the return of a call on transfers,
like this: (I had to use the DEVSTATE backport here)
[phones]
exten => _12XX,1,Dial(SIP/${EXTEN},6,tT)
exten => _12XX,n,GotoIf($[ "x${BLINDTRANSFER}" = "x" ]?noBT)
exten => _12XX,n,Set(DIALRET=${CUT(BLINDTRANSFER,-,1)});
exten => _12XX,n,Goto(dRet)
exten => _12XX,n(noBT),GotoIf($[
2013 Apr 17
1
core console debug on single file
Hi all,
I have console debugging enabled in logger.conf:
console => notice,warning,error,debug
Then a issue de command:
core set debug 100 manager.c
To see only debugging messages from AMI.
But It shows nothing!!!
And then if I do:
core set debug 1
Then I can see managar.c debug info, BUT if lots of other debug from all
other files.
How to see only manager.c (or any other