Re: [asterisk-users] Is Asterisk a Linux only system?
On Feb 12, 2015, at 9:11 AM, Joshua Colp jc...@digium.com wrote: Justin Sherrill wrote: I would love to run Asterisk on a BSD system. I do not know of any developers actively working on Asterisk on a BSD platform, though my knowledge isn't comprehensive. I'm also unaware of anyone developing on BSD like that. Linux of course and a smattering of folks on OSX doing the odd thing. /me is an odd developer occasionally doing odd things on OS X. If anyone wants to improve Asterisk on any non-Linux system, a good place to start is to run it with some of the common developer flags enabled (pass --enable-dev-mode to configure, enable DO_CRASH, enable the TEST_FRAMEWORK and run the tests). The increased warning level on GCC catches a few portability bugs, that are usually straightforward to fix. The tests can be harder, since you don’t know if it’s a problem in the test itself, or in Asterisk. Unfortunately, I doubt the Python test suite would run on non-Linux. I don’t even bother trying to run it on Ubuntu; I have a CentOS VM specifically for running the test suite to avoid platform problems. -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com http://www.digium.com/ www.asterisk.org http://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
Re: [asterisk-users] Is Asterisk a Linux only system?
On Feb 12, 2015, at 10:42 AM, Matthew Jordan mjor...@digium.com wrote: On Thu, Feb 12, 2015 at 10:38 AM, David M. Lee d...@digium.com mailto:d...@digium.com wrote: Unfortunately, I doubt the Python test suite would run on non-Linux. I don’t even bother trying to run it on Ubuntu; I have a CentOS VM specifically for running the test suite to avoid platform problems. It runs just fine on Debian based systems. Most issues you will run into are just making sure the dependencies are set up correctly. It does require Python 2.6+ (recommended: Python 2.7 just in case something has slipped in that we missed.) I was sure it would. But the instructions are for CentOS/EL, and I used up all of my sense of adventure just getting Asterisk running on OS X. -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com http://www.digium.com/ www.asterisk.org http://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
Re: [asterisk-users] AMI timeouts
On Jul 11, 2013, at 2:40 AM, Alexander Frolkin a...@eldamar.org.uk wrote: The Java process has an AMI connection to Asterisk which it keeps open continuously. When sending an AMI request, the process will wait a certain amount of time for a response (I think the timeout is 5 seconds) and if it hasn't received it, will retry the request up to 5 times. Timeout based retries on AMI is probably going to be a problem. AMI uses TCP, which has its own retry/retransmission logic. While the TCP connection is alive, you can safely assume that the message is either on its way to Asterisk, Asterisk is processing the message, or the response is heading back to you. Now if the TCP connection fails after you've sent a request, but before you receive a response, then you really have no way of knowing if the request was received/processed/rejected/etc. At that point, the best you can do is reestablish the AMI connection and retry the action. Occasionally, the Java process logs show AMI requests timing out (after 5 tries). What I see in packet captures of the AMI traffic is: 1. Java process sends a request (e.g., add member to queue) Do you see the TCP ACK coming back from Asterisk? 2. Retries 5 seconds later 3. Retries again 5 seconds later 4. Retries again 5 seconds later 5. Retries again 5 seconds later 6. Logs a timeout 7. After a few more seconds, Asterisk replies to all five requests in one go (in a single packet; so, e.g., for add member to queue it would reply success, then four failures because the member is already added); however at this stage, the Java process has given up During the quiet period while you're waiting for the response, do you receive events over that AMI connection? Are there other actions that you're attempting to execute? Is there any consistency as to which commands are getting delayed? It feels like Asterisk queues up the AMI responses and then periodically sends out all the responses in the queue in one go. Is something like this going on? Does the frequency at which Asterisk flushes the queue depend on load? Are there any tunable in the config for this? No, there's no response queue in Asterisk. For the action's I've looked at, it pretty much immediately processes the request and sends the response. There are any number of reasons why the response would be delayed, but the 25 seconds delay you're seeing is excessive for any of the reasons I can think of. The resource you're using AMI to access may be busy doing something else. Or the request is simply taking that long to process. Packet loss could cause delays in getting responses, but usually not for the lengths of times you're talking about. I know it's not a lot of info, but hopefully you can turn up some logging or packet captures to narrow down what's going on. Thanks in advance, Alex -- 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
Re: [asterisk-users] Asterisk 11.3.0 - Mask for new file not correct
On Apr 29, 2013, at 10:51 AM, Ludovic Boué wrote: The fact is we want to use the RECORDED_FILE function from Application_Record module and create a file with 666 permissions. But when I check the created file, rights are not what I expected. [root@STD1-SRVASTSVI-03 pseudos]$ ll -rw-r--r-- 1 asterisk asterisk 51244 mars 29 16:04 Pseudo_2_.wav I checked the doc on https://wiki.asterisk.org/wiki/display/AST/Application_Record but I didn't find anything about umask permissions. I checked Doxygen, I can see file creation permissions is set to 666 #define AST_FILE_MODE 0666 http://doxygen.asterisk.org/trunk/asterisk_8h.html#a6293b2dae52a2b470494df672a26c42 What can I do to fix that or debug? The AST_FILE_MODE works by the same rules as mode parameter in open(2): The effective permissions are modified by the process's umask in the usual way: The permissions of the created file are (mode ~umask).[1] My guess is that the umask of your asterisk process is 022, which is very typical. You'll have to play around with your umask settings and file permissions to get things the way you want them. [1]: http://linux.die.net/man/2/open Ludovic BOUÉ -- 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
Re: [asterisk-users] pipe character in CDR user field
On Nov 29, 2012, at 3:54 AM, Vieri wrote:
I'm trying to set a CDR userfield to a custom value. This value may contain a
'|' but it's really just part of the value.
However, Asterisk keeps warning me about the application delimiter not being
a pipe.
It's NOT an application delimiter (it's just part of a variable value) so I'm
expecting Asterisk not to warn me about it.
Is it expected behavior?
Well, you certainly didn't expect it. I would not have expected it either :-D
Why?
So, back in the early days, Asterisk inconsistently used different delimiters
depending upon context. Sometimes it used pipes, sometimes commas.
Inconsistency is never good, so we picked a winner (commas) and put in code to
look for the loser (pipes) in the dial plan and warn if they show up. ('We' in
this case would be Asterisk developers of long ago, by the way).
The code that does this checking[1] isn't the smartest code in the world. It
basically looks at the data passed to the application, and if it contains a
pipe, and no comma, and warnings are enabled, it warns you.
So you could disable warnings, but that would turn off other warnings that
might be useful.
Another option would be a small hack in your dial plan: add a comma.
MSet(CDR(userfield)=|usr_r=vieri,PIPE_HACK=true) ; Asterisk warns if it
sees a pipe without a comma.
Since you're not trying to use a pipe as a delimiter, displaying the warning is
a mistake in Asterisk. The whole pipe/comma thing happened so long ago[2], it's
time to just lose the warning altogether.
[1]:
https://code.asterisk.org/code/browse/asterisk/branches/11/main/pbx.c?u=3r=376690#to1583
[2]: https://code.asterisk.org/code/changelog/asterisk?cs=188210
See the following log:
SIP/4053-007bAGI Rx EXEC Set CDR(userfield)=|usr_r=vieri
-- AGI Script Executing Application: (Set) Options:
(CDR(userfield)=|usr_r=vieri)
[Nov 29 10:53:08] WARNING[4815]: pbx.c:1563 pbx_exec: The application
delimiter is now the comma, not the pipe. Did you forget to convert your
dialplan? (Set(CDR(userfield)=|usr_r=vieri))
SIP/4053-007bAGI Tx 200 result=0
SIP/4053-007dAGI Rx EXEC Set CDR(userfield)=\|usr_r=vieri\
-- AGI Script Executing Application: (Set) Options:
(CDR(userfield)=|usr_r=vieri)
[Nov 29 10:54:57] WARNING[4838]: pbx.c:1563 pbx_exec: The application
delimiter is now the comma, not the pipe. Did you forget to convert your
dialplan? (Set(CDR(userfield)=|usr_r=vieri))
SIP/4053-007dAGI Tx 200 result=0
Thanks,
Vieri
--
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
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Re: [asterisk-users] Questions about extension.conf
On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote: That is a good answer. Thanks. Any reason why it is not documented? It's documented on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities Ron -- 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
Re: [asterisk-users] AGI and AMI stuff.
On Nov 15, 2012, at 10:54 AM, Ken D'Ambrosio wrote: Hey, all. I'm interested in doing some simple, very specific web pages for some of my users -- things like call groups, setting forwarding, and for the receptionist to transfer calls and see calls. Probably do this in Ruby or PHP, though I'm open-minded. Anyway, if someone could point me to some documentation -- dead tree, electronic, whatever -- that gives some fairly in-depth detail on this, I'd be most appreciative. AGI commands and AMI actions and events are documented on the wiki - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Command+Reference The wiki also has command references for 1.8 and 10. Asterisk: The Definitive Guide has good chapters on the protocols: * Book - http://shop.oreilly.com/product/9780596517342.do * AMI - http://ofps.oreilly.com/titles/9780596517342/asterisk-AMI.html * AGI - http://ofps.oreilly.com/titles/9780596517342/AGI.html There are client libraries that handle the protocol details for you. StarPy for Python is a fairly low-level wrapper around AMI/AGI. Adhearsion for Ruby is a fairly high level wrapper for building voice applications. I believe some exist for PHP, but I know nothing about them. Thanks! -Ken Good luck! -- 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
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
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
Re: [asterisk-users] AMI Permissions, all means different things?
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: Set acl: yes read perm: none 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: Set acl: yes read perm: none 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 -- 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
