Displaying 20 results from an estimated 230 matches similar to: "asterisk-1.8.0 compilation error"
2011 Apr 21
2
[asterisk-user] Can't get hostname on asterisk dialplan by ENV()
Hi Friend,
Can't get hostname environment variable on asterisk dialplan.
Help me about how to get hostname environment variable on asterisk dialplan.
I have written "export HOSTNAME" in /root/.bash_profile and when i execute
"echo $HOSTNAME" then get right hostname but not success through asterisk
dialplan.
Get environment variable path right value through below
2009 Dec 07
1
Error : SIP/2.0 401 Unauthorized
Hi Friends,
need to help.
*I have problem about sip : SIP/2.0 401 Unauthorized*
Is it require to nathelper module in kamailio ?
*what can i write kamailio.cfg file when kamailio and Asterisk on same
network?*
Scenario is like as :
-----------------------------
1) kamailio server on 172.18.100.74
kamailio.cfg ( nathelpler module )
-----------------
loadmodule "nathelper.so"
2011 Apr 30
12
HA Asterisk
Hi,
I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf,
but its not yet production ready. Can someone please pitch in about HA
feature in Asterisk ?
(HA -> High Availability.) Also, What would be the pros and cons of using
AsteriskNow over Asterisk ? Are the versions same in Asterisk and
AsteriskNow ? We have been evaluating Asterisk for our Voice Application and
it seems
2010 Nov 18
2
exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
Hi Friends,
i have installed and configure asterisk-1.8.0.
When i have tried asterisk start get below errors and not able to start
asterisk.
*FD 32767 exceeds the maximum size of ast_fdset!*
Thanks in advance.
--
Best Regards,
Rajnikant Vanza
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2011 Jun 08
2
No IVR listen at device end......SIP phone is working fine
Hi List,
When we make calls into asterisk with the help of our mobile, landline
number, Cisco 79XX series then we didn't able to here any IVR which is
playing into asterisk server. But when we dial from SIP softphone then all
is working fine and we are able to here the IVR sound files.
What is the problem in this case please help me..
--
-----
Thanks and regards
Virendra Bhati
2010 Sep 21
0
Dialplan extension pattern matching for '/' character
Hi Friends,
LOCAL/*89/9875784578
I want to match above dialstring into dialplan context.
How can i match dialplan extension pattern matching for "*89/9875784578"
with including '/' character.
Thanks in advance.
--
Best Regards,
Rajnikant Vanza
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2015 Apr 17
0
How to Answer QUEUE call through AMI
Hi All,
Anybody know about to using AMI to agent answer call that placed in QUEUE.
Asterisk Version : 1.8.9.1
Queue Name : agent-support
Agent logged in Extension : SIP/8001,SIP/8002 ...
strategy : ringall
1) How to intimate or send information to available agent to inside queue
incoming call ringing.
2) How to agent answer incoming call that placed in queue.
3) AMI for agent Hold/Unhold call.
2004 Aug 18
0
connections always mapped to guest
Hi,
I am trying to access a fileserver(NT4 based from a client machine
(Windows 2k Pro) and using SAMBA (version 3.0.2a) as a PDC through
username raj.It shows me the correct share on the file server but
doesn't allow me to change properties of any folder even though I am the
admin.
The releavnt configuration settings is correct..
I suspect that it gives me guest ACL.
When I see my debug 10
2014 Jul 18
0
How to get 2 CDR Records of 2 outgoing calls bridge
Hi all,
I need 2 CDR Records of below 2 numbers for outgoing calls, detail is given
as below:
*96XXXXXXXX88XXXXXXXX*
*=> Call file : outbound call generate through below file*
Test.call
======
Channel: local/s at outgoing/n
WaitTime: 45
Context: outgoing_ivrs
Extension: s
Priority: 1
Set: contact_no=96XXXXXXXX
extensions.conf
============
[outgoing]
exten => s,1,NoOP(----- First LEG
2010 Oct 21
1
Asterisk 1.8.0-rc5: Blind transfer failed, SIP REFER Method
Hi,
I setup an asterisk system (version 1.8.0-rc5). While using a SIP only
environment I discovered a problem using blind transfer. The phones are
SNOM or Aastra and are using the SIP REFER Method.
The following is working:
User A calls user B, B accepts the call, user A than transfers to user C
The following is NOT working:
User A calls user B, B accepts the call, user B than transfers to user
2009 Jun 17
0
WriteXLS - New Version 1.8.0
The updated package has been submitted to CRAN and will propagate to
mirrors over the next day or so.
It is maintained on R-Forge at http://r-forge.r-project.org/projects/writexls
, where downloads are available as well.
Package: WriteXLS
Version: 1.8.0
Description: Cross-platform perl based R function to create Excel (XLS)
files from one or more data frames. Each data frame will be written
2009 Jun 17
0
WriteXLS - New Version 1.8.0
The updated package has been submitted to CRAN and will propagate to
mirrors over the next day or so.
It is maintained on R-Forge at http://r-forge.r-project.org/projects/writexls
, where downloads are available as well.
Package: WriteXLS
Version: 1.8.0
Description: Cross-platform perl based R function to create Excel (XLS)
files from one or more data frames. Each data frame will be written
2003 Nov 12
1
value of strptime in R 1.8.0
Dear R-people!
I am using R 1.8.0, under Windows XP.
What I want to do is a date conversion of a character column of a data frame
and assign the result as a new column.
Simple example:
> x <- data.frame(a=c("yesterday","today","tomorrow"), b=I(c("20031111",
"20031112", "20031113")))
# convert x$b from character to date:
>
2009 Apr 13
1
value of strptime in R 1.8.0
Dear R friends,
I have a data frame, I need to get a time interval between the two columns.
The times are recorded in 24 hour clock. My data frame is called
version.one.
my commands are:
t.s.one<-paste(version.one[,9])
t.s.two<-paste(version.one[,61])
x<-strptime(t.s.one,format="%H:%M")
x
y<-strptime(t.s.two ,format="%H:%M")
y
z<-difftime(y,x, units =
2010 Nov 01
1
DISA problem in 1.8.0
When I call into my Asterisk box via my VoIP line (using gsm codec) and then
try to make an outgoing DISA call over PSTN I get the following:
[Nov 1 15:12:54] WARNING[17694]: chan_dahdi.c:8930 dahdi_write: Cannot
handle frames in gsm format
[Nov 1 15:12:54] WARNING[17694]: app_dial.c:1401 wait_for_answer: Unable to
forward voice or dtmf
Obviously, it looks like asterisk is not converting the
2003 Nov 24
0
Re: [RMySQL] unable to establish connection since R-1.8.0 upgrade
Hi Laurent,
The issue is a very simple one to correct --- you only need to
re-install RMySQL with the R version you're currently using.
The issue is that the internal representation of objects in the
methods package changed in 1.8.0, and thus packages that use S4
classes and methods and save their images (like RMySQL) and were
installed prior to 1.8.0 need to be re-installed to be used with
2010 Oct 22
0
CEL ODBC problem in 1.8.0
Hi,
I have been experimenting with CEL in a trunk version of asterisk for some time and have upgraded my test machine to 1.8.0 today.
Made a few calls and it looks like the eventtype field is missing in the CEL insert query when using ODBC. I see the following errors on the console:
[Oct 22 21:46:09] WARNING[952]: res_odbc.c:634 ast_odbc_prepare_and_execute: SQL Execute returned an error -1:
2012 Jul 27
0
[ANNOUNCE] xf86-input-mouse 1.8.0
xf86-input-mouse is the Xorg server mouse driver for non-evdev OS'es.
This release drops support for Xorg 1.6.x, and now requires Xorg 1.7 or later.
On Solaris, it fixes a compatibility issue with the Xorg 1.13 ABI changes,
allowing this driver to build for the upcoming Xorg 1.13 release.
It also exposes "Device Node" (read-only), "Mouse Middle Button Emulation"
2010 Sep 23
2
rtp problem with 1.8.0-rdc1
Hi. I am having a very strange problem --aren't they all -- with the
release candidate. I have softphone which talks to asterisk from behind
nat -- the asterisk is on a public ip -- and when I hit mute on the
softphone, all rtp traffic ceases! Now, a version which does work is
r281875, this does not happen in that vrsion, but right after that this
strange thing starts and is not fixed in
2010 Oct 21
0
Asterisk 1.8.0 Now Available!
The Asterisk Development Team is proud to announce the release of Asterisk
1.8.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4. For more information about
support time lines for Asterisk releases, see the