Re: [asterisk-users] ControlPlayback's options
On 2011-06-04 13:38, virendra bhati wrote:
Hi Johan Wilfer,
Thanks for your reply. On the basis of your provided code I made all
things into extensions.conf. But i have an small issue on which I need
your attention again.
in below context what's ${tz} ? Is this time zone value or else?
Yes, I store the calls timezone in tis variable before the code sample
you got.
and another things is what is the use of SayUnixTime(${time},${tz},d
'digits/of' B);
this function in such case?
I've implemented 5 as a pause-button on the phone. This context handles
this by playing a
prompt that you have pause the recording and time and date.
This is repeated untill the user presses a key on the keypad.
context conference_play_recordings_
conference_paused {
announce = {
Set(time=$[${epoch_start}+${position}/1000]);
while(true) {
WaitExten(1);
Background(conf_playrec_pause_part1);
SayUnixTime(${time},${tz},kM);
Background(conf_playrec_pause_part2);
SayUnixTime(${time},${tz},d 'digits/of' B);
Background(conf_playrec_pause_part3);
WaitExten(5);
}
}
one thing which is also confusing is that what is the meaning or use
of such lines in this application.
ControlPlayback(${filename},6,3,1,*#2456790,,,o(${position}))
${filename} is the file you want to play.
6 is 60 seconds to skip.
3 is to use 3 as forward 60 seconds
1 to to use 1 as rewind 60 seconds
*#2456790 is used as stop buttons (and handled by the dialplan)
o() is a option to go to a specific position in the file
${position} is the variable that hold the current position of the playback.
To get more details use the following command:
asterisk*CLI core show application ControlPlayback
Displays:
-= Info about application 'ControlPlayback' =-
[Synopsis]
Play a file with fast forward and rewind.
[Description]
This application will play back the given filename.
It sets the following channel variables upon completion:
${CPLAYBACKSTATUS}: Contains the status of the attempt as a text string
SUCCESS
USERSTOPPED
ERROR
${CPLAYBACKOFFSET}: Contains the offset in ms into the file where playback
was at when it stopped. '-1' is end of file.
${CPLAYBACKSTOPKEY}: If the playback is stopped by the user this variable
contains the key that was pressed.
[Syntax]
ControlPlayback(filename[,skipms[,ff[,rew[,stop[,pause[,restart[,options]]])
[Arguments]
skipms
This is number of milliseconds to skip when rewinding or fast-fo
rwarding.
ff
Fast-forward when this DTMF digit is received. (defaults to '#')
rew
Rewind when this DTMF digit is received. (defaults to '*')
stop
Stop playback when this DTMF digit is received.
pause
Pause playback when this DTMF digit is received.
restart
Restart playback when this DTMF digit is received.
options
o(time):
time - Start at time ms from the beginning of the
file.
[See Also]
Not available
/Johan
Please put some light on these too.
On Wed, Jun 1, 2011 at 1:50 AM, Johan Wilfer li...@jttech.se
mailto:li...@jttech.se wrote:
On 2011-05-30 14:32, virendra bhati wrote:
Hi List,
Asterisk 's *ControlPlayback* will used for play any recorded
file as an audio player. Is it possible that we can use it for
multiple forward and rewind ?
ex:-
original: ControlPlayback(filename,skipms,ff,rew,stop,pause)
expected
ControlPlayback(filename,skip1,skip2,skip3,forward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause)
:
Yes, you can use the CPLAYBACKSTATUS, CPLAYBACKOFFSET and
CPLAYBACKSTOPKEY variables to get this behavior.
All you have to do is to list the additional keys and stop keys
and implement this in your dialplan...
I've attached some ael I use for this to implement 1 and 3 as 1
minute rewind/forward. 4 and 6 as 5 minutes rewind/forward and 7
and 9 as 15 minutes.
5 I use as the pause key, and */# to switch recording.
Greetings,
Johan Wilfer
context conference_play_recordings_conference_connect {
playrec_intro = {
Set(position=0);
goto play,1;
}
play = {
while (true) {
if (${position}==-1) { goto recording_end,1; }
//rewind 5 seconds after every action (so the user doesn't
feel lost...)
Set(position=$[${position}-5000]);
if (${position} 0) { Set(position=0); }
ControlPlayback(${filename},6,3,1,*#2456790,,,o(${position}));
Set(position=${CPLAYBACKOFFSET});
if (${CPLAYBACKSTATUS}==ERROR) {
Playback(pbx_error_500);
Playback(pbx_endcall);
Wait(2);
Hangup();
}
//If stopped by user
if (${CPLAYBACKSTATUS}==USERSTOPPED) {
if (!${ISNULL(${CPLAYBACKSTOPKEY})}) { goto
${CPLAYBACKSTOPKEY},1; }
}
}
}
[asterisk-users] Blind transfer issue on Asterisk 1.8.4.2
Hi all, when doing a blind transfer using the keys defined in features.conf, we hear a confirmation of the attempt to blindly transfer, followed by an invalid extension message. The console says this: [Jun 4 22:30:31] VERBOSE[11301] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/570-0006 [Jun 4 22:30:31] VERBOSE[11301] file.c: -- SIP/518-0005 Playing 'pbx-transfer.gsm' (language 'nl') [Jun 4 22:30:32] WARNING[11301] features.c: Extension '53' does not exist in context 'transfer_context,570,1' [Jun 4 22:30:32] VERBOSE[11301] file.c: -- SIP/518-0005 Playing 'pbx-invalid.gsm' (language 'nl') [Jun 4 22:30:34] VERBOSE[11301] res_musiconhold.c: -- Stopped music on hold on SIP/570-0006 Mind you, the entered extension is 531, so it seems part of the entry is cut off. Sometimes it shows just 5. It is as if featuredigittimeout (set to 2000) is not taken into account. Thx!! B. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.6 - 511 Command not permitted causing high CPU usage
http://pastebin.com/vxGM2n5j We are getting those errors 100x per second in console when AGI set debug is on It is causing extremely high CPU usage, we've tried asterisk version 1.6.1.22 and 1.6.2.18 It seems the problem is worse in 1.6.2.18 Can someone advise how to fix this? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
On 3 June 2011 22:41, Hans Witvliet h...@a-domani.nl wrote: On Fri, 2011-06-03 at 09:07 +0100, Ishfaq Malik wrote: Are you suggesting that there are no bugs in 1.4 or 1.6? I presume that you are aware of the fact that it is impossible to prove the absence of bugs in any piece of software You might not have detected them yet. Furthermore behaviour that might have been coded on purpose, can be considered eroneously some time later. Currently there seems to be a fear of 1.8. We're about to put it into production and yes, we've had issues with it, mostly due to the fact we use RealTime, but before you change anything it is always advisable to test the hell out of it. To anyone who is thinking of moving to 1.8 the question is not, 'is it stable?'. The question is, 'have I comprehensively tested it to show that it is suitable for my needs?' If you put it into production, test at least the functions that you are going to use. There might (and probably will) problems in the code, but as long as it does not bother you, so what? See this thread here about Asterisk 1.8 - and Digium's view on the matter. http://nerdvittles.com/?p=743 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
digium, eat your dog food! On Sun, Jun 5, 2011 at 11:08 AM, dotnetdub dotnet...@gmail.com wrote: On 3 June 2011 22:41, Hans Witvliet h...@a-domani.nl wrote: On Fri, 2011-06-03 at 09:07 +0100, Ishfaq Malik wrote: Are you suggesting that there are no bugs in 1.4 or 1.6? I presume that you are aware of the fact that it is impossible to prove the absence of bugs in any piece of software You might not have detected them yet. Furthermore behaviour that might have been coded on purpose, can be considered eroneously some time later. Currently there seems to be a fear of 1.8. We're about to put it into production and yes, we've had issues with it, mostly due to the fact we use RealTime, but before you change anything it is always advisable to test the hell out of it. To anyone who is thinking of moving to 1.8 the question is not, 'is it stable?'. The question is, 'have I comprehensively tested it to show that it is suitable for my needs?' If you put it into production, test at least the functions that you are going to use. There might (and probably will) problems in the code, but as long as it does not bother you, so what? See this thread here about Asterisk 1.8 - and Digium's view on the matter. http://nerdvittles.com/?p=743 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.6 - 511 Command not permitted causing high CPU usage
On Sun, 5 Jun 2011, vip killa wrote: http://pastebin.com/vxGM2n5j We are getting those errors 100x per second in console when AGI set debug is on Can someone advise how to fix this? Thank you. Don't request 'WAIT FOR DIGIT 1000' from a dead channel. Don't ignore the error from 'WAIT FOR DIGIT 1000' Don't loop on the error. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] broken SVN asterisk 1.8 ?
Hey guys! I have just download latest SVN Revision 322051 and compile and install but my asterisk -V showing still old version :( is it broken ? /usr/sbin/asterisk -V Asterisk SVN-branch-1.8-r321926 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF issue in app_konference using with asterisk 1.8.3.2
Hi, I have a requirement where the DTMF entered by a member in konference is passed on to the other members. But the DTMF is not being recognized, when checked the events from manager API, I do see DTMF event being passed, but some how it is not passed to other members. This tells me - may be it is not an asterisk issue, but more a konference application issue. Is this not supported by app_konference or am i missing any thing specific in my configuration? I am using the flags (Mx) for the konference application. I am using asterisk 1.8.3.2 and app_konference 1.7. I also tried changing DTMF to 1 in the Makefile of konference application. Wondering if any other member in the group is having a similar issue with app_konference and asterisk 1.8.3.2 and any suggestions would be appreciated? Thanks, Krishna -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ControlPlayback's options
Hi John Wilfer,
Thanks for replay. Now all things is working on asterisk 1.6.2.18 version.
But When I try the same thing on Asterisk 1.4.X then facing problem.
Is this the problem of ControlPlayback 's option fields of asterisk 1.4.X
in this version have option P(jumping) not O(time) ?
Is there any way by which we will implement like by upload ControlPlayback
from asterisk 1.6 to 1.4 or else ?
ControlPlayback(filename[,skipms[,ff[,rew[,stop[,pause[,restart[,options]]])
On Sun, Jun 5, 2011 at 2:16 PM, Johan Wilfer li...@jttech.se wrote:
On 2011-06-04 13:38, virendra bhati wrote:
Hi Johan Wilfer,
Thanks for your reply. On the basis of your provided code I made all things
into extensions.conf. But i have an small issue on which I need your
attention again.
in below context what's ${tz} ? Is this time zone value or else?
Yes, I store the calls timezone in tis variable before the code sample you
got.
and another things is what is the use of SayUnixTime(${time},${tz},d
'digits/of' B);
this function in such case?
I've implemented 5 as a pause-button on the phone. This context handles
this by playing a
prompt that you have pause the recording and time and date.
This is repeated untill the user presses a key on the keypad.
context conference_play_recordings_
conference_paused {
announce = {
Set(time=$[${epoch_start}+${position}/1000]);
while(true) {
WaitExten(1);
Background(conf_playrec_pause_part1);
SayUnixTime(${time},${tz},kM);
Background(conf_playrec_pause_part2);
SayUnixTime(${time},${tz},d 'digits/of' B);
Background(conf_playrec_pause_part3);
WaitExten(5);
}
}
one thing which is also confusing is that what is the meaning or use of
such lines in this application.
ControlPlayback(${filename},6,3,1,*#2456790,,,o(${position}))
${filename} is the file you want to play.
6 is 60 seconds to skip.
3 is to use 3 as forward 60 seconds
1 to to use 1 as rewind 60 seconds
*#2456790 is used as stop buttons (and handled by the dialplan)
o() is a option to go to a specific position in the file
${position} is the variable that hold the current position of the playback.
To get more details use the following command:
asterisk*CLI core show application ControlPlayback
Displays:
-= Info about application 'ControlPlayback' =-
[Synopsis]
Play a file with fast forward and rewind.
[Description]
This application will play back the given filename.
It sets the following channel variables upon completion:
${CPLAYBACKSTATUS}: Contains the status of the attempt as a text string
SUCCESS
USERSTOPPED
ERROR
${CPLAYBACKOFFSET}: Contains the offset in ms into the file where playback
was at when it stopped. '-1' is end of file.
${CPLAYBACKSTOPKEY}: If the playback is stopped by the user this variable
contains the key that was pressed.
[Syntax]
ControlPlayback(filename[,skipms[,ff[,rew[,stop[,pause[,restart[,options]]])
[Arguments]
skipms
This is number of milliseconds to skip when rewinding or fast-fo
rwarding.
ff
Fast-forward when this DTMF digit is received. (defaults to '#')
rew
Rewind when this DTMF digit is received. (defaults to '*')
stop
Stop playback when this DTMF digit is received.
pause
Pause playback when this DTMF digit is received.
restart
Restart playback when this DTMF digit is received.
options
o(time):
time - Start at time ms from the beginning of the
file.
[See Also]
Not available
/Johan
Please put some light on these too.
On Wed, Jun 1, 2011 at 1:50 AM, Johan Wilfer li...@jttech.se wrote:
On 2011-05-30 14:32, virendra bhati wrote:
Hi List,
Asterisk 's *ControlPlayback* will used for play any recorded file as an
audio player. Is it possible that we can use it for multiple forward and
rewind ?
ex:-
original: ControlPlayback(filename,skipms,ff,rew,stop,pause)
expected
ControlPlayback(filename,skip1,skip2,skip3,forward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause)
:
Yes, you can use the CPLAYBACKSTATUS, CPLAYBACKOFFSET and
CPLAYBACKSTOPKEY variables to get this behavior.
All you have to do is to list the additional keys and stop keys and
implement this in your dialplan...
I've attached some ael I use for this to implement 1 and 3 as 1 minute
rewind/forward. 4 and 6 as 5 minutes rewind/forward and 7 and 9 as 15
minutes.
5 I use as the pause key, and */# to switch recording.
Greetings,
Johan Wilfer
context conference_play_recordings_conference_connect {
playrec_intro = {
Set(position=0);
goto play,1;
}
play = {
while (true) {
if (${position}==-1) { goto recording_end,1; }
//rewind 5 seconds after every action (so the user doesn't feel
lost...)
Set(position=$[${position}-5000]);
if (${position} 0) { Set(position=0); }
[asterisk-users] Asterisk users Calculation
Dears I already read most of post on asterisk group and (http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning) But I could not find a calculator 1-Is there a calculator I can download for that 2-What I the maximum simultaneous calls that can asterisk handle using CPU 3.0 MHZ and 4GB ram With rtp g729 and there is no codec transcoding , 3-And what is the number of simultaneous calls if I use direct RTP (Canreinvite=no /Directrt=yes) Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com image001.png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk users Calculation
On Sun, 5 Jun 2011, Khaled W. Chehab wrote: 1-Is there a calculator I can download for that 2-What I the maximum simultaneous calls that can asterisk handle using CPU 3.0 MHZ and 4GB ram With rtp g729 and there is no codec transcoding 3-And what is the number of simultaneous calls if I use direct RTP (Canreinvite=no /Directrt=yes) 1) No. Because every case is a bit different and nobody has taken the time to research and document it. 2) In the 'hundreds.' I have a 5 yr old 3.4 Xeon server with 2GB of ram running all kinds of AGIs that handles 300 simultaneous ULAW calls without issue and without any 'tuning.' The Asterisk process uses less than 100MB so more GBs means nothing. 3) Probably in the thousands depending on what those calls are doing. (Just guessing here because I have no experience with this configuration.) Would a SIP server like OpenSIPS be a better platform choice? More details will yield better responses. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] broken SVN asterisk 1.8 ?
On Sun, Jun 05, 2011 at 04:18:25PM +, satish patel wrote: Hey guys! I have just download latest SVN Revision 322051 and compile and install but my asterisk -V showing still old version :( is it broken ? /usr/sbin/asterisk -V Asterisk SVN-branch-1.8-r321926 asterisk -V shows the last changed revision in the build. To see the difference, try: cd asterisk-src-dir svnversion svnversion -c -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk users Calculation
May I add...I still have documented cases of asterisk 1.4.x running ulaw with no transcoding and running 2k+ concurrent calls on a CentOS 4(5?, fuzzy) machine with 2ghz CPU and 2gb ram Sent from my iPhone On Jun 5, 2011, at 3:02 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 5 Jun 2011, Khaled W. Chehab wrote: 1-Is there a calculator I can download for that 2-What I the maximum simultaneous calls that can asterisk handle using CPU 3.0 MHZ and 4GB ram With rtp g729 and there is no codec transcoding 3-And what is the number of simultaneous calls if I use direct RTP (Canreinvite=no /Directrt=yes) 1) No. Because every case is a bit different and nobody has taken the time to research and document it. 2) In the 'hundreds.' I have a 5 yr old 3.4 Xeon server with 2GB of ram running all kinds of AGIs that handles 300 simultaneous ULAW calls without issue and without any 'tuning.' The Asterisk process uses less than 100MB so more GBs means nothing. 3) Probably in the thousands depending on what those calls are doing. (Just guessing here because I have no experience with this configuration.) Would a SIP server like OpenSIPS be a better platform choice? More details will yield better responses. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] broken SVN asterisk 1.8 ?
Thanks but they should change svn revesion number change in file. -- Sent from my iPhone On Jun 5, 2011, at 7:13 PM, Barry Miller asterisk-us...@notanet.net wrote: On Sun, Jun 05, 2011 at 04:18:25PM +, satish patel wrote: Hey guys! I have just download latest SVN Revision 322051 and compile and install but my asterisk -V showing still old version :( is it broken ? /usr/sbin/asterisk -V Asterisk SVN-branch-1.8-r321926 asterisk -V shows the last changed revision in the build. To see the difference, try: cd asterisk-src-dir svnversion svnversion -c -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI STREAM FILE not working?
Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten = 5150,1,Answer() same = n,Set(CHANNEL(language)=en_AU) same = n,AGI(testagi.sh) same = n,Hangup console output: -- Executing [5150@AllPhones:1] Answer(SIP/PBX-0024, ) in new stack -- Executing [5150@AllPhones:2] Set(SIP/PBX-0024, CHANNEL(language)=en_AU) in new stack -- Executing [5150@AllPhones:3] AGI(SIP/PBX-0024, testagi.sh) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/testagi.sh -- Playing 'welcome' (escape_digits=1) (sample_offset 0) -- SIP/PBX-0024AGI Script testagi.sh completed, returning 0 -- Executing [5150@AllPhones:4] Hangup(SIP/PBX-0024, ) in new stack == Spawn extension (AllPhones, 5150, 4) exited non-zero on 'SIP/PBX-0024' But nothing happens...as in even when it says that it's playing the file (as verified in the asterisk 'full' log), I hear nothing on the phone What gives? spent 2 hrs Googling but nothing! :( Thx \A -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk users Calculation
It would be a great help to others(including me) if those using 1.8.X can provide some details on hardware configurations,features they have implemented on it and some sort of load testing results. Thanks, [SATISH] On Mon, Jun 6, 2011 at 6:28 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: May I add...I still have documented cases of asterisk 1.4.x running ulaw with no transcoding and running 2k+ concurrent calls on a CentOS 4(5?, fuzzy) machine with 2ghz CPU and 2gb ram -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] issues.asterisk.org
i'm trying to review issues that i'm monitoring and/or have reported at http://issues.asterisk.org when I click on 'My View' or 'View Issues' I get an error: APPLICATION ERROR #401 Database query failed. Error received from database was #1142: DELETE command denied to user 'mantisreadonly'@'localhost' for table 'mantis_tokens_table' for the query: DELETE FROM mantis_tokens_table WHERE '2011-06-06 00:03:56' expiry. Are tickets that I had set up for monitoring on mantis going to be automatically monitored in jira ? similarly, are tickets that I reported in mantis going to show as me being the reporter in jira? or are the tickets going to stay in mantis until they are resolved and never make it into jira ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issues.asterisk.org
On 6/6/2011 1:08 AM, Jeremy Kister wrote: similarly, are tickets that I reported in mantis going to show as me being the reporter in jira? or are the tickets going to stay in mantis until they are resolved and never make it into jira ? after some more clicking, i see the answer to this one; nevermind. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
