Re: [asterisk-users] AMI Permissions, "all" means different things?
2012-09-07 16:13, David M. Lee skrev:
On Sep 7, 2012, at 1:49 AM, Johan Wilfer wrote:
Hi!
I'm trying to limit the permissions for a AMI-account. But I'm a little bit confused by
the permissions. The commands I use are (output from "manager show commands",
btw: privilege col seems cropped?):
Yes, sadly it is.
Action PrivilegeSynopsis
Redirect call,all Redirect (transfer) a call.
Originateoriginate,allOriginate a call.
Getvar call,reporting, Gets a channel variable.
If I put this in my manager.conf:
[pbx_ami]
secret = ***
deny=0.0.0.0/0.0.0.0
permit = x.x.x.x/255.255.255.255
write=originate,call
read=
I get this ("manager show user pbx_ami"):
username: pbx_ami
secret:
acl: yes
read perm:
write perm: call,originate,all
displayconnects: yes
Where does the "all" permission come from?
Probably just a bug in the 'manager show user' command. The user doesn't have
all the permissions, so 'all' shouldn't show up in the list. If it's not
already in the issue tracker, please file a bug[1].
[1]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
However, If I change the row in manager.conf to "write=originate,call,all" the
output is:
username: pbx_ami
secret:
acl: yes
read perm:
write perm:
system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate,agi,cc,aoc,test,all
displayconnects: yes
Can someone explain this please?
This is at least looks correct. The 'all' permission is a superset of, well,
all the permissions. The 'write=all' line in manager.conf assigns all of these
permissions to the user.
Thanks!
--
Johan Wilfer
Thank you David for the feedback.
I reported the following bugs:
https://issues.asterisk.org/jira/browse/ASTERISK-20397 (all bug)
https://issues.asterisk.org/jira/browse/ASTERISK-20396 (cropped col)
--
Johan Wilfer
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[asterisk-users] Asterisk as a translating proxy only?
Hello, We want to use Asterisk as a proxy to translate between Skinny/SCCP and SIP, with as little as possible work required in between. Does Asterisk have a way for custom programs to read and write raw packets? If we can get the input data in a readable format and output it in the required format, that may do the trick. Thank you for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Async AGI
Hi Danny, I am running it asynchronously because adhearsion needs it. Regards, Pavel Date: Wed, 5 Sep 2012 11:28:46 -0500 From: "Danny Nicholas" Subject: Re: [asterisk-users] Async AGI To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Message-ID: <00cd01cd8b83$862a05d0$927e1170$@debsinc.com> Content-Type: text/plain; charset="us-ascii" As I understand it, when the AGI is asynchronous, you lose the control of checking for completion, etc. Since you are running prior to answer(), why run it asynchronously. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pavel Siderov Sent: Wednesday, September 05, 2012 11:26 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Async AGI Hi, Is there a way to execute next priority in the dialplan if you have called agi:async? I want to play warning message if adhearsion is down. Currently I wasn't able to make it work. The dialplan execution ends after the first priority. [incomming] exten => _X.,1,AGI(agi:async) exten => _X.,2,Answer exten => _X.,3,Playback(some-message) exten => _X.,4,Hangup Regards, Pavel -- _ -- 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] Queue and reinvite
Hi, I have 10 agents who are pstn lines in queue and would like that when they answer the rtp should go directly Is it at all possible in queues? If yes what could be bothering it from happening? Thanks, Israel -- _ -- 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 crashing when recording ConfBridge calls (10.7.1)
Hi list,
since I enabled call recording in my ConfBridge bridge profile, after
about every 5th held conference (sometimes after 3, sometimes after 10,
it depends), when the last users exits the conference, Asterisk crashes.
Is this a known error? I'm using Asterisk 10.7.1 on CentOS 64bit with
the official RPMs.
lb1*CLI>
[Sep 10 15:06:18] WARNING[21582]: file.c:496 filehelper: File
/var/spool/asterisk/confbridge/confbridge-name-22*68*39*09-1347282201.46.sln
detected to have zero size.
-- Playing
'/var/lib/asterisk/sounds/chatfire_konferenz_verlassen_6.slin' (language
'en')
-- Playing 'confbridge-leave.gsm'
(language 'en')
== MixMonitor close filestream (mixed)
== End MixMonitor Recording
ConfBridgeRecorder/conf-22*68*39*09-uid-1446923861
lb1*CLI>
[Sep 10 15:06:19] WARNING[21582]: channel.c:2880 ast_hangup: Hard hangup
called by thread 1115089216 on Bridge/0xaf10ba8-output, while fd is
blocked by thread 1115597120 in procedure ast_waitfor_nandfds!
Expect a failure
lb1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
And here's the snippet from confbridge.conf:
[chatfire-internal-bridge]
type=bridge
record_conference=yes
sound_has_joined=/var/lib/asterisk/sounds/chatfire_betritt_konferenz_6
sound_has_left=/var/lib/asterisk/sounds/chatfire_konferenz_verlassen_6
sound_only_person=/var/lib/asterisk/sounds/chatfire_erster_anrufer_6
sound_there_are=/var/lib/asterisk/sounds/chatfire_es_befinden_sich_derzeit_6
sound_other_in_party=/var/lib/asterisk/sounds/chatfire_anrufer_in_der_besprechung_6
sound_get_pin=/var/lib/asterisk/sounds/chatfire_intro_6
And the dialplan:
; internal conference only
exten => 06,1,Answer()
exten => 06,n,Set(CHANNEL(language)=chatfire)
exten => 06,n,Set(CONFBRIDGE(user,template)=chatfire-internal-user)
exten => 06,n,Set(CONFBRIDGE(bridge,template)=chatfire-internal-bridge)
exten =>
06,n,Set(CONFBRIDGE(bridge,record_file)=/tmp/chatfire-${STRFTIME(,,%C%y%m%d%H%M)}.wav)
exten => 06,n,ConfBridge(22*68*39*09,,,chatfire-internal-menu)
exten => 06,n,Hangup()
Help! :)
Thank you,
Markus
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Re: [asterisk-users] Async AGI
That being the case, you are going to need to do a dialplan modification
something like this:
Exten => _X.,1.Set(answerstatus=false)
exten => _X.,2,AGI(agi:async)
exten => _X.,3,Gotoif($["${answerstatus}"=="false"?7:4)
exten => _X.,4,Answer
exten => _X.,5,Playback(some-message)
exten => _X.,6,Hangup
exten => _X.,7,Verbose(adhesion failed)
exten => _X.,8,Playback(failure-message)
exten => _X.,9,Hangup
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pavel Siderov
Sent: Monday, September 10, 2012 3:57 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Async AGI
Hi Danny,
I am running it asynchronously because adhearsion needs it.
Regards,
Pavel
Date: Wed, 5 Sep 2012 11:28:46 -0500
From: "Danny Nicholas"
Subject: Re: [asterisk-users] Async AGI
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Message-ID: <00cd01cd8b83$862a05d0$927e1170$@debsinc.com>
Content-Type: text/plain; charset="us-ascii"
As I understand it, when the AGI is asynchronous, you lose the control of
checking for completion, etc. Since you are running prior to answer(), why
run it asynchronously.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pavel Siderov
Sent: Wednesday, September 05, 2012 11:26 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Async AGI
Hi,
Is there a way to execute next priority in the dialplan if you have called
agi:async? I want to play warning message if adhearsion is down. Currently I
wasn't able to make it work. The dialplan execution ends after the first
priority.
[incomming]
exten => _X.,1,AGI(agi:async)
exten => _X.,2,Answer
exten => _X.,3,Playback(some-message)
exten => _X.,4,Hangup
Regards,
Pavel
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Re: [asterisk-users] Asterisk crashing when recording ConfBridge calls (10.7.1)
- Original Message - > From: "Markus" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Monday, September 10, 2012 8:19:47 AM > Subject: [asterisk-users] Asterisk crashing when recording ConfBridge calls > (10.7.1) > > Hi list, > > since I enabled call recording in my ConfBridge bridge profile, after > about every 5th held conference (sometimes after 3, sometimes after > 10, > it depends), when the last users exits the conference, Asterisk > crashes. > Is this a known error? I'm using Asterisk 10.7.1 on CentOS 64bit with > the official RPMs. > No, that is not a known bug. Please open an issue in the issue tracker. https://issues.asterisk.org/jira -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://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
Re: [asterisk-users] AMI Permissions, "all" means different things?
On Sep 10, 2012, at 2:38 AM, Johan Wilfer wrote: > Thank you David for the feedback. > > I reported the following bugs: > > https://issues.asterisk.org/jira/browse/ASTERISK-20397 (all bug) > https://issues.asterisk.org/jira/browse/ASTERISK-20396 (cropped col) Thanks! -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
[asterisk-users] MixMonitor inserting extra 20ms packets of silence (1.4.43)
I'm trying to diagnose potential causes of an issue with MixMonitor in 1.4.43 (I doubt it's very version-specific). I won't have hands on the kit until the end of the week, but I have listened to some recordings. It doesn't happen on every call - only sometimes. Basically, it is a call from one SIP phone extension to another, and the dialplan sets up MixMonitor on the calling channel before doing the Dial. The call sounds fine to both parties in the call, but when listening back to the recording, the speech is slowed down and broken - a bit like a robot or dalek voice. Examining the recorded audio in Goldwave I can see that every 20ms (the size of a SIP packet), an extra 20ms of silence is being inserted in the recording file. This happens most of the time, punctuated by occasional bursts of clear audio before it starts happening again. Has anyone seen this kind of thing before? Better still, seen it and solved it? The "SIP phones" are actually Soundwin ATAs. There is no zaptel or dahdi timing source in the system. Are there any known issues with MixMonitor that could cause this behaviour? Any pointers would be appreciated - thanks! Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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
Re: [asterisk-users] Async AGI
Danny as I mentioned in the first message the dialplan execution stops after AGI(agi:async). So the provided example doesn't help. Regards, Pavel -- _ -- 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] Async AGI
Perhaps the AGI is improperly terminating and should stop and return an error instead of just "dying" like it does now. Just catch the failing condition and return a variable to the dialplan so life will be good. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pavel Siderov Sent: Monday, September 10, 2012 11:17 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Async AGI Danny as I mentioned in the first message the dialplan execution stops after AGI(agi:async). So the provided example doesn't help. Regards, Pavel -- _ -- 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] Maximum messages in voicemail
Greetings list, I've seen a few errors recently in our logs along the lines of: [Sep 10 17:41:41] WARNING[6719] app_voicemail.c: Save failed. Not moving message: destination folder full. maxmsg in voicemail.conf is set to 1000. I've checked the mailboxes on the server in question, and the maximum number of messages in any account is 535. The filesystem in question is less than 50% full; likewise available inodes are plentiful. Does asterisk have a hardcoded maxmsg figure that supercedes voicemail.conf at a certain point? Is there any way of telling which mailbox the warning relates to? Anything else worth checking? Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] MixMonitor inserting extra 20ms packets of silence (1.4.43)
2012-09-10 18:13, Tony Mountifield skrev: I'm trying to diagnose potential causes of an issue with MixMonitor in 1.4.43 (I doubt it's very version-specific). I won't have hands on the kit until the end of the week, but I have listened to some recordings. It doesn't happen on every call - only sometimes. Basically, it is a call from one SIP phone extension to another, and the dialplan sets up MixMonitor on the calling channel before doing the Dial. The call sounds fine to both parties in the call, but when listening back to the recording, the speech is slowed down and broken - a bit like a robot or dalek voice. Examining the recorded audio in Goldwave I can see that every 20ms (the size of a SIP packet), an extra 20ms of silence is being inserted in the recording file. This happens most of the time, punctuated by occasional bursts of clear audio before it starts happening again. Has anyone seen this kind of thing before? Better still, seen it and solved it? The "SIP phones" are actually Soundwin ATAs. There is no zaptel or dahdi timing source in the system. Are there any known issues with MixMonitor that could cause this behaviour? Any pointers would be appreciated - thanks! Tony Maybe a I'm reaching here but.. I had some very strange issues with broken quality with Monitor and a NFS mount. This was 1.4, but several years ago. I ended up not using NFS in the end. -- Johan Wilfer -- _ -- 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] Maximum messages in voicemail
What flavor of asterisk? Realtime or just files? Post your voicemail.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: Monday, September 10, 2012 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Maximum messages in voicemail Greetings list, I've seen a few errors recently in our logs along the lines of: [Sep 10 17:41:41] WARNING[6719] app_voicemail.c: Save failed. Not moving message: destination folder full. maxmsg in voicemail.conf is set to 1000. I've checked the mailboxes on the server in question, and the maximum number of messages in any account is 535. The filesystem in question is less than 50% full; likewise available inodes are plentiful. Does asterisk have a hardcoded maxmsg figure that supercedes voicemail.conf at a certain point? Is there any way of telling which mailbox the warning relates to? Anything else worth checking? Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Maximum messages in voicemail
On 10/9/12 6:48 pm, Danny Nicholas wrote: What flavor of asterisk? Realtime or just files? Post your voicemail.conf. Flat files, latest 1.4.x Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Async AGI
On Sep 10, 2012, at 11:16 AM, Pavel Siderov wrote: > Danny as I mentioned in the first message the dialplan execution stops after > AGI(agi:async). So the provided example doesn't help. > > Regards, > Pavel Pavel, This may be due to the asynchronous nature of Async AGI. The AGI(agi:async) command will continue to run until the channel is hung up, the 'asyncagi break' command is sent over AMI, or some sort of fatal error is encountered. When a call comes in, the AGI(agi:async) command sill send out a start event over AMI. But if Adhearsion isn't there to hear it, it's also not there to tell Asterisk what to do with the channel. So it will sit there, waiting for a break or hangup. I'm sure the Adhearsion guys have run into this sort of thing before. You may want to ask on their forum[1] for how they handle the situation. [1]: https://groups.google.com/forum/?fromgroups=#!forum/adhearsion -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
