[asterisk-users] meetme and dtmf
Hi Group, is it possible to read the DTMF tones from a caller while he is in a meetme conference? I would like to read the pressed key sequence and call a command like MeetMeAdmin or System Commands. I'm using Asterisk 1.8.7. Thanks for help Daniel -- _ -- 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] meetme identify user number
Hi Group, is in MeetMe any option to identify the own number (from the view of a caller)? I would like to write an option to set on the telephone an request for voice, if the room is muted. That request should display on our Conference Control Website and an Admin should unmute this person. Thanx for help. Daniel -- _ -- 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] keep dst cdr record if context change
Hello nice group, having a Problem with CDRs. If i change the context with Goto() Asterisk write the new exten in dst cdr field. How can i keep the old entry? Any ideas makes me very happy. Thanks for helping me. Daniel -- _ -- 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] keep dst cdr record if context change
Hi Danny, Thank you for answer. I'm using Asterisk 1.8.7.0 Daniel Am 30.03.2012 um 20:18 schrieb Danny Nicholas da...@debsinc.com: More information please - 1.8X or 10.X? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Knoll Sent: Friday, March 30, 2012 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] keep dst cdr record if context change Hello nice group, having a Problem with CDRs. If i change the context with Goto() Asterisk write the new exten in dst cdr field. How can i keep the old entry? Any ideas makes me very happy. Thanks for helping me. Daniel -- _ -- 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 -- _ -- 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] keep dst cdr record if context change
In your case i have 1 record and only the last one s is in the field dst, but i don't use s, i use numbers instead. Thats my situation :( Am 30.03.2012 um 21:21 schrieb Danny Nicholas: So you have a situation like so: [default] Exten = _X.,1,Answer Exten = _X.,n,Goto(foo,s,1) [foo[ Exten = s,1,playback(vm-goodbye) Exten = s,n,hangup() And you get two CDR records, 1 with default and 1 with foo? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Knoll Sent: Friday, March 30, 2012 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] keep dst cdr record if context change Hi Danny, Thank you for answer. I'm using Asterisk 1.8.7.0 Daniel Am 30.03.2012 um 20:18 schrieb Danny Nicholas da...@debsinc.com: More information please - 1.8X or 10.X? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Knoll Sent: Friday, March 30, 2012 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] keep dst cdr record if context change Hello nice group, having a Problem with CDRs. If i change the context with Goto() Asterisk write the new exten in dst cdr field. How can i keep the old entry? Any ideas makes me very happy. Thanks for helping me. Daniel -- _ -- 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 -- _ -- 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 -- _ -- 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] keep dst cdr record if context change
Looks nice, was also my first idea, but field dst is read only. I can't
overwrite this and get an error like this
ERROR[2474]: cdr.c:345 ast_cdr_setvar: Attempt to set the 'dst' read-only
variable!.
Am 30.03.2012 um 22:00 schrieb Warren Selby:
On Fri, Mar 30, 2012 at 2:21 PM, Danny Nicholas da...@debsinc.com wrote:
So you have a situation like so:
[default]
Exten = _X.,1,Answer
Exten = _X.,n,Goto(foo,s,1)
[foo[
Exten = s,1,playback(vm-goodbye)
Exten = s,n,hangup()
And you get two CDR records, 1 with default and 1 with foo?
No, he should be getting 1 record with s in the dst field.
To the OP: have you tried setting a channel variable to ${EXTEN} before your
Goto() command, and then in the h exten write it back into the cdr?
Something like:
[incoming]
exten = _X.,1,Verbose(New call coming in - verify routing)
exten = _X.,n,Set(finaldst=${EXTEN})
exten = _X.,n,Goto(mainmenu,s,1)
exten = h,1,Verbose(Hanging up)
exten = h,n,Set(CDR(dst)=${finaldst})
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: write system command output into a variable
I found a solution that works fine for me
Set(var1=${SHELL(shellcommand)})
Bye Daniel
Von: Daniel Knoll dan...@danielknoll.de
Datum: 16. April 2011 13:13:28 MESZ
An: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Betreff: write system command output into a variable
Hey Guys,
i would like to write the output from my bash script into a Variable, that is
readable by Asterisk
Using this:
Set(var1=${FILE(/dev/shm/tempfile.txt,0,6)})
is not very helpful because this command reading fixed character length. If i
read 6 characters and in the file only 3 i get 123
Can anyone help me ?
Thanx a lot for help
Daniel
--
_
-- 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] In which version is eventfilter working?
Hey Guys, In which Version of Asterisk is EventFilter: in manager.conf working? Higher than 1.6.2.10 or from the 1.8.0 Version? Thank for your answer Daniel -- _ -- 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] filtering AMI Event: RTCPSent
Hi Godson Gera, thank you for your answer. if i understand correctly, the EventMask filter all until i define all event that i need. This is not really helpful, because i must define the categories that i need. mhh is there another Solution for my Problem? Thanks a lot Daniel Am 09.12.2010 um 17:15 schrieb Godson Gera: On Thu, Dec 9, 2010 at 1:29 AM, Daniel Knoll dan...@danielknoll.de wrote: Hey Guys, for debugging i need to read the Events from AMI. But i have a lot of unwanted RTCPSent Events. How can i filter this Events in Asterisk 1.6.2.x Version of Asterisk? You can control to some extent on what kind of events are sent using EventMask check out this page http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Events So by setting proper eventmask you might be able to filter out RTCPSent but the mask may alos filter out other events that fall in the mask category. http://forums.digium.com/viewtopic.php?f=1t=9927start=0 -- Thanks Regards, Godson Gera Asterisk Consultant India -- _ -- 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 Daniel Knoll Liberdastr. 9 12047 Berlin fon +49 (0)179 20 16 50 8 mail dan...@danielknoll.de web www.danielknoll.de -- _ -- 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] filtering AMI Event: RTCPSent
Hey Guys, for debugging i need to read the Events from AMI. But i have a lot of unwanted RTCPSent Events. How can i filter this Events in Asterisk 1.6.2.x Version of Asterisk? Thanks a lot for your answers Daniel -- _ -- 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] don't leave meetme conf if key pressed
Hi @ all, what is the best way to to use features like MeetmeCount without leaving the conference. I use Meetme(,X) and MEETME_EXIT_CONTEXT=context, but the problem is that the caller leave the Conference :( Is it possible to press a key, without this obstacle? Thanx for your answers Daniel Knoll -- _ -- 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] meetme don't play conf-invalid if room does not exist
Has anyone a solution for me - with Meetme(,Ms)asterisk plays conf-invalid if a room not exist - with Meetme(123,Ms) asterisk plays not conf-invalid if the room not exist and asterisk hangup I am happy about any proposal. Thanks Daniel -- _ -- 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] more condition check for gotoif
Hello, is it possible to check more than one condition for GOTOIF in the dialplan? Or is the normal way to cascade the diaplan each GOTOIF? The Background is that I would like to check more than 2 values from a Variable, and then route the call based on the value. Thanks for your help. Daniel Knoll -- _ -- 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] MeetMe errorhandling
Hi Kai-Uwe,
thank you for your answer. but it doesn't work.
i use this dialplan.
exten = 22,n,Answer()
exten = 22,n,NoCDR()
exten = 22,n,WaitExten(2)
exten = 22,n,Set(CHANNEL(musicclass)=music)
exten = 22,n,Set(CHANNEL(language)=de)
exten = 22,n,Read(roomid,conf-getconfno,6,1)
exten = 22,n,MeetMe(${roomid},Ms)
exten = 22,n,Hangup()
exten = i,1,Playback(conf-invalid)
exten = i,n,Goto(22,1)
exten = t,1,Goto(22,1)
Sometimes, but only sometimes the caller jump into the i extension with the
same room number (which doesn't exist), but i can't comprehend.
i see, that MeetMe get the Roomnumer, then he drop the call
- Executing [...@provider:8] MeetMe(SIP/100-3b5c, 212,Ms) in new stack
== Parsing '/usr/local/asterisk-1.6.2.9/etc/asterisk/meetme.conf': == Found
== Spawn extension (provider, 22, 8) exited non-zero on 'SIP/100-3b5c'
Any Ideas?
Thanx,
Daniel
Am 06.09.2010 um 23:54 schrieb Kai-Uwe Jensen:
I use MeetMe(,Ms) in the Dialplan and if a Conference Room does't exist
Asterisk play (conf-invalid.slin)
If i use MeetMe(${room},Ms) (value from DTMF Read) and the Conference Room
doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk Hangup the
Call.
Use the i extension to control what happens when entering an invalid room
number. Simple example:
exten = 5000,Goto(confline,s,1)
[confline]
exten = s,1,Background(enter-conf-call-number)
exten = s,n,WaitExten(20)
exten = i,1,Playback(conf-invalid)
exten = i,n,Goto(s,1)
exten = t,1,Goto(s,1)
; Participants always dial a 7-digit conference number, optionally followed
; by the #-sign
exten = _XXX,1,MeetMe(${EXTEN},Mxwsp)
exten = _XXX,n,Hangup()
exten = _XXX#,1,Goto(${EXTEN:-8:7},1)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe errorhandling
Hi Paul,
i set Answer() .. just Cut the first, my fault.
is that the normal case, to treat errors like wrong conference Room?
Daniel
Am 07.09.2010 um 15:01 schrieb Paul Belanger:
On Tue, Sep 7, 2010 at 3:11 AM, Daniel Knoll dan...@danielknoll.de wrote:
Hi Kai-Uwe,
thank you for your answer. but it doesn't work.
i use this dialplan.
c exten = 22,n,NoCDR()
exten = 22,n,WaitExten(2)
exten = 22,n,Set(CHANNEL(musicclass)=music)
exten = 22,n,Set(CHANNEL(language)=de)
exten = 22,n,Read(roomid,conf-getconfno,6,1)
exten = 22,n,MeetMe(${roomid},Ms)
exten = 22,n,Hangup()
exten = i,1,Playback(conf-invalid)
exten = i,n,Goto(22,1)
exten = t,1,Goto(22,1)
Your dialplan is missing priory 1 for you Answer().
exten = 22,1,Answer()
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
--
_
-- 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] MeetMe errorhandling
Hi Group,
i have a MeetMe Question.
I use MeetMe(,Ms) in the Dialplan and if a Conference Room does't exist
Asterisk play (conf-invalid.slin)
If i use MeetMe(${room},Ms) (value from DTMF Read) and the Conference Room
doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk Hangup the
Call.
there is a solution for the kind my problem?
Thanx and bye
Daniel
--
_
-- 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] iax stresstest client
Hello Everybody, does anyone knows an opensource stresstest client for the IAX protocol, like sipp? Thanx for your answer. Daniel -- _ -- 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] alaw.h in app_meetme.c
Hi Group, short question. is it possible to use #include asterisk/alaw.h instead of #include asterisk/ulaw.h in app_meetme.c or is ulaw required in meetme? thanx for the answer. Daniel -- _ -- 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] MeetMe transcode / format problem
Hi Group, actual i have a transcode problem and i have no idea to solve this. All my wav files are alaw encoded and i allow only alaw codec. But sometimes the WriteFormat is slin and if i recall the same number the WriteFormat is alaw for the Channel. Why the channel has sometimes slin and sometimes alaw? NativeFormats: 0x8 (alaw) WriteFormat: 0x40 (slin) ReadFormat: 0x8 (alaw) WriteTranscode: Yes ReadTranscode: No After this I'm going into a conference Room and the Format completely change to slin. NativeFormats: 0x8 (alaw) WriteFormat: 0x40 (slin) ReadFormat: 0x40 (slin) How can i change the Format for Meetme to alaw which is the NativeFormat. Thanks for your help. Daniel PS: i unload the format_sln16.so and format_sln.so modules. -- _ -- 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] send Variable to remote system via AMI / Orginate
Hi all, is it possible to send a Variable to another System via IAX Protocoll by using AMI / Orginate Like this: Action: Originate Channel: IAX2/user1:passw...@192.168.1.2/6...@default Application: Meetme Data: 111,q Variable: var1=111 and the Remote System knows the Variable var1 ? In my Test it fails to transfer the Variable :( What I doing wrong? Thanx for your help Daniel -- _ -- 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] send Variable to remote system via AMI / Orginate
Hello Group,
I found a solution for my problem.
I use the CallerID Variable with the Orginate Action to send a value to another
System
Action: Originate
Channel: IAX2/user1:passw...@192.168.1.2/6...@default
Application: Meetme
Data: 111,q
CallerID: 111
and on the other Side:
exten = ,n,MeetMe(${CALLERID(num)},q)
Maybe it is helpful for all others.
Daniel
Am 12.07.2010 um 18:58 schrieb Daniel Knoll:
Hi all,
is it possible to send a Variable to another System via IAX Protocoll by
using AMI / Orginate
Like this:
Action: Originate
Channel: IAX2/user1:passw...@192.168.1.2/6...@default
Application: Meetme
Data: 111,q
Variable: var1=111
and the Remote System knows the Variable var1 ?
In my Test it fails to transfer the Variable :(
What I doing wrong?
Thanx for your help
Daniel
--
_
-- 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] strange issue while setting pin in MeetMe
Dear Group, after a compile Asterisk 1.6.2.6, i have strange issue. 1.) i get a recording message in Log, but i don't set the Option r 2.) if the room number has a entry pin, the caller get a voice Message to left a Name, i'd never set this Option. How can i disabled to play vm-rec-name / this function? Starting recording of MeetMe Conference 333 into file meetme-conf-rec-333-1278153806.209.wav. -- Recording -- SIP/dnsnet-in-0048 Playing 'vm-rec-name.slin' (language 'de') -- SIP/dnsnet-in-0048 Playing 'beep.slin' (language 'de') -- x=0, open writing: /usr/local/asterisk-1.6.2.6//var/spool/asterisk/meetme/meetme-username-333-1 format: sln, 0x84fad88 My Meetme Extension is: exten = ,1,Answer() exten = ,n,NoCDR() exten = ,n,WaitExten(2) exten = ,n,Set(CHANNEL(musicclass)=music) exten = ,n,Set(CHANNEL(language)=de) exten = ,n,MeetMe(,Msp) exten = ,n,Hangup() thanx for help. Daniel -- _ -- 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] joining 2 conferences together
Is it possible to join 2 meetme conferences (each on different server) together, that if i load balance the callers, they can see altogether something like a inter system communikation ? Thanx for your help. Daniel -- _ -- 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] load balance meetme
Hello wise Group, if i plan to load balance incoming SIP Calls for MeetMe Conference to 2 or more Server, i think it is a Problem, because each Server opened his own MeetMe Room/Channel. Is it possible to made some interconnect the dahdi or MeetMe Channels over many Servers? (Like PHP, it can store the SessionID in a distributed filesystem or mysql database) Or what is the best way, to load balance Meetme conferences? I read abut openSIPS, thats sounds nice, but additionally i would like a kind of asterisk channel interconnect for better load balance. Thanx for your help. Daniel -- _ -- 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] dahdi span
ok, thanx for your answer. Daniel Am 20.06.2010 um 19:17 schrieb Tilghman Lesher: On Saturday 19 June 2010 10:47:07 Daniel Knoll wrote: Hello Group, what does the Compiler Option mean LOTS_OF_SPANS ? The description is: More than 32 DAHDI spans Does this mean, more than 32 DAHDI Channels ? No, it means spans. If you only have analog DAHDI channels, then you should ignore this option. It is meant for those with PRI or SS7 links. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 Daniel Knoll Liberdastr. 9 12047 Berlin fon +49 (0)179 20 16 50 8 mail dan...@danielknoll.de web www.danielknoll.de -- _ -- 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] dahdi span
Hello Group, what does the Compiler Option mean LOTS_OF_SPANS ? The description is: More than 32 DAHDI spans Does this mean, more than 32 DAHDI Channels ? Thanx for help. Daniel -- _ -- 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] debug message: Internal timing is disabled
Hi all, i got a lot of this messages if only one caller is in a meetme conference and it playing a MusicOnHold Sound. If a second Caller entry the Conference the messages ended. DEBUG[11794] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=61 What does this message mean? Thanx for answers Daniel -- _ -- 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] bug with Moh on MeetMe ?
Hi Michael, Can you show us the output from: moh show classes and moh show files Command Or try it to set a new exten after setting the language with: exten = 12345,n,Set(CHANNEL(musicclass)=personalised) Daniel Am 13.06.2010 um 12:35 schrieb Mickael Monsieur: Hello, The MeetMe application refuses MusicOnHold personalized and skip always in the default! Have you any idea how to fix this? -- Executing [028883...@default:1] Set(SIP/109.10.214.1-0002, CHANNEL(language)=fr) in new stack -- Executing [028883...@default:2] Answer(SIP/109.10.214.1-0002, ) in new stack -- Executing [028883...@default:3] Playback(SIP/109.10.214.1-0002, welcome) in new stack -- SIP/109.10.214.1-0002 Playing 'welcome.alaw' (language 'fr') [Jun 13 12:30:00] NOTICE[13437]: channel.c:3012 __ast_read: Dropping incompatible voice frame on SIP/109.10.214.1-0002 of format ulaw since our native format has changed to 0x8 (alaw) -- Executing [028883...@default:4] MeetMeCount(SIP/109.10.214.1-0002, 100,COUNT) in new stack == Parsing '/etc/asterisk/meetme.conf': == Found -- Executing [028883...@default:5] GotoIf(SIP/109.10.214.1-0002, 0?100) in new stack -- Executing [028883...@default:6] MeetMe(SIP/109.10.214.1-0002, 100,1pdM(personnalised)) in new stack -- Created MeetMe conference 1023 for conference '100' -- Started music on hold, class 'personnalised', on SIP/109.10.214.1-0002 -- Stopped music on hold on SIP/109.10.214.1-0002 -- Started music on hold, class 'default', on SIP/109.10.214.1-0002 Thank you, Mickael. -- _ -- 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 Daniel Knoll Liberdastr. 9 12047 Berlin fon +49 (0)179 20 16 50 8 mail dan...@danielknoll.de web www.danielknoll.de -- _ -- 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] MeetMe problem
Hi Guys, sometimes if one caller or many callers are in a meetme Room and a new one join the room, then he or another caller into the same room where kickt from the room. It's very strange for me and in logs (full) I can't see anything. is it possible to log more from meetme.c ? can anyone help me and maybe someone has also the problem as i and have an solution. I use: asterisk-1.6.2.7 dahdi-linux-complete-2.3.0+2.3.0 asterisk-addons-1.6.2.1 Thanx a lot for any answers that helps me. Daniel -- _ -- 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] MeetMe problem
it's not so easy, because i use a mysql database (realtime) to write the room number from a webapp into the table. also i extend the meetme table for my web application :-/ any other things at least to show more logs from meetme or dahdi ? Daniel Am 12.06.2010 um 17:39 schrieb Thomas Perron: try using confbridge in lastest asterisk version On Sat, Jun 12, 2010 at 11:30 AM, Daniel Knoll dan...@danielknoll.de wrote: Hi Guys, sometimes if one caller or many callers are in a meetme Room and a new one join the room, then he or another caller into the same room where kickt from the room. It's very strange for me and in logs (full) I can't see anything. is it possible to log more from meetme.c ? can anyone help me and maybe someone has also the problem as i and have an solution. I use: asterisk-1.6.2.7 dahdi-linux-complete-2.3.0+2.3.0 asterisk-addons-1.6.2.1 Thanx a lot for any answers that helps me. Daniel -- _ -- 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 Daniel Knoll Liberdastr. 9 12047 Berlin fon +49 (0)179 20 16 50 8 mail dan...@danielknoll.de web www.danielknoll.de -- _ -- 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] call droped if second caller enter meetme conference
Hello Group, some strange problem i have on my setup. If a second caller entering a meetme conference dropping the first one. my setup is using asterisk 1.6.2.6 and dahdi 2.1.1.1 with realtime, the conference room numbers storing in a mysql database. the calls came from a sip provider. there are nothing in logfiles with sip debug on :( Has anyone the same problem and a solution for me? Thanx for all. Daniel Knoll -- _ -- 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] meetme changes between asterisk 1.6.2.6 and 1.6.2.7
Hi Guys, is it possible that the silence joining with the Option q in a MeetMe room damaged ? I updated to Version 1.6.2.7 (before 1.6.2.6) and now my silence Orginates to Play Voice into a Meetme Room will play a bleep after a Success Orginate I Orginate with this simple AMI request. Action: Originate Channel: Local/1122 Application: Meetme Data: 1234,q, Can anyone reproduce this ? Thanx for your help. Daniel Knoll-- _ -- 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] play a sound file directly to a caller channel
Hi Jim,
First i'm a little bit confused, because your code was a little bit difficult
to read, but now i understand.
it works fine in my Setup, Big Thanks for your help.
bye
Daniel
Am 16.05.2010 um 16:11 schrieb Jim Dickenson:
We do the following:
Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280_18-0002
ActionID: PlayBack
Async: true
exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()
exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 16, 2010, at 4:16 AM, Daniel Knoll wrote:
Hello User-List,
is it possible to play a sound file directly to a caller channel?
Like this AMI command
Action: Originate
Channel: SIP/20-1d41
Application: Playback
Data: /path/to/audio/file
I get an Error Message. My intension is to play a sound file to a caller and
the other callers don't hear this.
Can someone help me ?
Thanks a lot
Bye Daniel
--
_
-- 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
Daniel Knoll
Liberdastr. 9
12047 Berlin
fon +49 (0)179 20 16 50 8
mail dan...@danielknoll.de
web www.danielknoll.de
--
_
-- 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] play a sound file directly to a caller channel
Hello User-List, is it possible to play a sound file directly to a caller channel? Like this AMI command Action: Originate Channel: SIP/20-1d41 Application: Playback Data: /path/to/audio/file I get an Error Message. My intension is to play a sound file to a caller and the other callers don't hear this. Can someone help me ? Thanks a lot Bye Daniel -- _ -- 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
