Re: [asterisk-users] Case-sensitivity of Dialplan variables.
On Tue, 2012-10-02 at 17:11 -0700, Ira wrote: At 02:19 PM 10/1/2012, you wrote: So respond here and let me know what you think. I got a couple of replies on the -dev list and they said that this would be good to put out on the -users list too. Mark Michelson In true Republican fashion, I'm going to vote for case-insensitivity. Given that many of the users were not programmers and didn't likely grow up in a case sensitive world I'd also vote for case insensitivity. I fall into that category, I grew up with dBase, Clipper and VB and case issues get me all the time when I program in C. I would vote for case-sensitivity. True, i grew up in the early day's of PDP11, flex, uniflex and so-on, where case-sensitivity was default. I think it is a bad habit to write something else, from what you expect. More important is, that you get a un-avoidable error, when you try to read a variable, that isn't initialised (due to mixed case). Like in the old fortran/pascal/C days, where you just get a compilation error, that you had to solve before you could continue There is already too much insensitivity in this world, let's get rid of (at least) case insensitivity! hans -- _ -- 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] Case-sensitivity of Dialplan variables.
At 07:59 PM 10/2/2012, you wrote: Given that many of the users were not programmers and didn't likely grow up in a case sensitive world I'd also vote for case insensitivity. I fall into that category, I grew up with dBase, Clipper and VB and case issues get me all the time when I program in C. Allowing case insensitivity does not stop someone from using case consistently and While I guess there could be a reason why you'd want to use the word hash in the forms hash, Hash and HASH and have them be 3 different items, I'm guessing that the people trying to get their feet wet moving from Asterisk-Now to Asterisk would be confused to say the least if someone did that in example code. While true that most users are probably not programmers, most people administering Asterisk would be system / network admins, correct? System admins and networking admins are used to working in environments such as Linux where variables and file names are case sensitive. If someone is moving from a GUI interface to CLI, then they would/should know that case sensitivity is important and therefore the change shouldn't pose a problem. I'm not a system / network admin, at least not for Linux. I have one Linux machine, it runs Asterisk and Samba. I can usually make Asterisk do what I want. Samba works but I have little to no idea why. I run yum update occasionally and I run V11 trunk or whatever the proper name would be for the development version. If there was a compiler and declared variables then case makes perfect sense. Without that, I'd never get a C program to work. I know people want case sensitivity, it's the right way to do it, but how does it help Asterisk? Does anyone have configurations that would be broken by case insensitivity? If not, then what is the upside of enforcing case sensitivity? Ira -- _ -- 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] Case-sensitivity of Dialplan variables.
Il 01/10/2012 23.15, Mark Michelson ha scritto: https://issues.asterisk.org/jira/browse/ASTERISK-20163 The issue involves case-sensitivity of channel and global variables in the dialplan. +1 for case-sensitive variables everywhere. I'm glad to see that this inconsistency issue will be finally addressed. Whatever the final result it is important that there is **consistency**. I believe also that from a programming perspective may be a small gain in parsing efficiency using case-sensitive variables. -- TeeBX VoIP communication platform (coming soon) http://code.google.com/p/teebx/ --- Lightweight++ Business Friendly++ Open++ -- _ -- 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] Parameterize asterisk config files
On Tue, 2012-10-02 at 19:34 -0400, Paul Belanger wrote: On 12-10-02 06:39 PM, Mitch Claborn wrote: Asterisk 1.8 on Ubuntu We store the configuration files in CVS. We have a development, QA and production environments. 90% of the config files are the same across all 3 environments, but there are some differences in sip.conf and extensions.conf (environment specific voip providers and/or analog/digital lines). I'd like to be able to use the same config files in CVS and have the differences resolved at run time, based on host name of the asterisk server. Any ideas how to do this? I looked at STS, but it appears to be Mac only. One idea would be to use something like #include sip-$$$hostname$$$.conf and then use sed or similar in the startup script to replace $$$hostname$$$ with the actual host name. Then each host/environment would have it's own include file as needed. Another idea would be to write a simple perl or other program to pre-process the files and put some markers in the files themselves. ; onlyif host=abc ; /onlyif The pre-processor would delete lines between the tags that didn't match the currently running host. If you are going to astricon you'll want to show up for my talk. This is basically what Leif and I will be talking about. I use puppet to help manage our 3 environments (test, stage and production). Along side it I use a the following configuration setup[1] plus some Debian packaging scripts[2]. With this, I can quickly spin up instances which are provisioned to a base. Then, depending on puppet manifests[3] for each node, it defines how the system is then provisioned. If more per-site settings are required, I'll roll them into Debian packages (we use Ubuntu 12.04) and have each site subscribe to a customer repo. [1] https://github.com/kickstandproject/asterisk/tree/master/debian/ast_config [2] https://github.com/kickstandproject/astricon-2012-presentation/tree/master/debian [3] https://github.com/kickstandproject/puppet-modules/tree/master/modules/asterisk/manifests This is how we do it as well, and then control the puppet config files using GIT Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] Call Termination Provider Madness
Op 03-10-12 01:17, Chris Nighswonger schreef: On Tue, Oct 2, 2012 at 5:30 PM, Chris Bagnall aster...@lists.minotaur.cc wrote: On 2/10/12 6:51 pm, Carlos Alvarez wrote: Your traffic level, number of concurrent calls, etc would help us know what sort of carrier you should be talking to. Equally important, your geographic location, and the geographic location to which most of your calls are made will be useful in helping list members advise you. We do ~4000+ min of outbound calling per month and just about that inbound. Not a large volume. We have four DID's (one of which is 800). Our calling patterns are mostly the lower 48 with a smattering international. We are located in NC. RTP is the problem in the FW. I just cannot see opening all RTP ports to $universal. But I'm probably missing a key piece of information. :-) Kind Regards, Chris Chris, Have a look at your /etc/asterisk/rtp.conf file. In it you specify the UDP portrange your asterisk will use for RTP traffic. change the rtpstart and rtpend to your needs and set them open in your FW. Do not make the range too small each active call will normally take one RTP channel incoming and one RTP channel outgoing. I have mine set to for example: rtpstart=1 and rtpend=10100. This should be enough for 100 simultanious calls. Regards, Michel. -- _ -- 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] Case-sensitivity of Dialplan variables.
On Monday 01 October 2012, Danny Nicholas wrote: I propose that dialplan variables need to be made consistent in their evaluation. We need to choose either to be always case-sensitive or always case-insensitive. The problem is, I don't know which of these changes would have a larger effect on people. This is where I would like your feedback. Which way should it go? Case-sensitive always is the better option. It's just more intuitive. Most other things in Unix-like environments are case-sensitive, so you come to expect it. More things are likely to be broken by going case-insensitive. If someone, somewhere is relying on STUFF, Stuff and stuff being distinct, then they probably have a very good reason for doing so. Besides, if we start accommodating that sort of sloppiness, where will it lead next? Will we stop distinguishing between I, 1 and L, or 0 and O because they look similar? How about letters which are next to one another on the keyboard? Perhaps, bearing in mind the imperfect rollover action on many keyboards, we should treat stuff and sutff the same?/hysterics -- AJS Answers come *after* questions. -- _ -- 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] Call me now outbound calls in a queue
The problem is that you need to have a process waiting for a free agent and then doing the reschedule. Instead of writing your own, you could try our WombatDialer (that is currently free as in beer, as it is being community tested) to automate such a task. It has a nice HTTP API and it would do exactly what you are looking for. See http://wombatdialer.com/ l. 2012/9/28 Mitch Claborn mitch...@claborn.net That approach only works if there are any agents that are not busy on a call - I could pick one, take them out of the queue then connect the call. If all agents are busy, I need to be able to insert the request into the queue so that it gets processed in sequence with the inbound calls. Mitch -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
[asterisk-users] asterisk 1.8 parking not working
Hi guys, I've upgraded my pbx from asterisk 1.4 to 1.8 but parking does not work anymore. Tried asterisk-1.8.11.0 and then, after reading about a (fixed) problem in CHANGELOG tried asterisk-1.8.16.0, without success. My features.conf is: [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in, need to INCLUDE this in extensions.conf parkingtime = 45 ; Number of seconds a call can be parked for (default is 45) My extensions.conf is: [inbound] include=parkedcalls ... [outbound] include=parkedcalls as written on the manual (Oreilly ver.3). They seem right to me but when I transfer a call to exten 700 I get an invalid extension message as if Asterisk wouldnt' recognize 700 as a special extrension. Any idea? Thank you. Giorgio -- _ -- 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] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
The problem I see with this approach is that you usually do not just want to dial out 10 calls at a time, but you will want to keep track of what happened to them and (in case) reschedule them. So you will likely need to monitor them over AMI to make sure they went through, and you need to implement some rescheduling logic. [Shameless plug starts here] This was the reason why we started working on Wombat a while ago - to offer something that would handle all this (and more) but leaving you the Asterisk touch of being free to program the call handling at the dialplan level, so you would get the best of both worlds. Did I already mention the current beta versions are free? :) [Shameless plug ends here] I am not saying that this is the only correct solution (or it is a correct solution at all) but our almost ten years of Asterisk call-center experience show that what starts out as something quick and simple to plug a hole ends up being a platform :) Just my two Swiss cents, l. 2012/9/28 A J Stiles asterisk_l...@earthshod.co.uk On Friday 28 September 2012, Patrick Archibald wrote: Hi, Is there a way to move 100 .call files in to /var/spool/asterisk/outgoing/ at once and have Asterisk call at maximum 10 at a time? Yes: Move them in batches of 10. Could be as simple as last if ++$n_files 9; if the script is in Perl. You know how many calls you can deal with at once; it's up to you to stay within your own limits. Asterisk just tries its damnedest to do whatever it's been told, without imposing any sort of judgement as to whether it's sane or wholesome. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR
Another option that seems to be very good for handling logs where you write quite a lot is Cassandra - http://cassandra.apache.org/ - but of course you lose the SQL layer on top - unless you go for something like http://blog.mariadb.org/announcing-the-cassandra-storage-engine/ This may not be completely off topic here because you get high data security / crash protection and parallel cluster writes, so you could insert tens/hundreds of thousands of events per second on a suitably dimensioned cluster for an Asterisk server cluster of similar size :) l. 2012/9/28 Leif Madsen leif.mad...@asteriskdocs.org On 27/09/12 11:45 AM, Matt Hamilton wrote: Date: Thu, 27 Sep 2012 10:23:35 +0200 From: lenz.lo...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR I'd go for MyISAM and would set up a remote replica if data integrity is important. If you have like 1000 calls of (say) 30 seconds avg length, and you create 10 events per call, you would expect an event every three seconds. This is about 300 inserts per second. Say 600 at peaks. This should be feasible with server-grade hardware without much difficulty. Also as you always INSERT it behaves as a log file (no seeking, no locking) if the table is optimized. l. We decided to go with MyISAM since it supports concurrent inserts (as you suggested). Data integrity (a slight loss of call records) is something we can live by. Right now we use DRBD for replication, but I guess with MyISAM it doesn't make much sense if the db crashes. We are looking into other options as well. This may or may not be relevant, but you can also check out MySQL/Galera[0] for clustering solutions. Not sure if that gets you closer or further from your goal though :) It uses a modified InnoDB to allow a multi-master MySQL cluster. I used a chef cookbook to deploy it[1]. [0] http://www.codership.com/content/using-galera-cluster [1] http://support.severalnines.com/entries/21453521-opscode-s-chef-mysql-galera-and-clustercontrol -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Receptionist console (software)
On Wed, 2012-10-03 at 12:48 +0300, James Mutuku wrote: Any recommendations I can use. I am looking on having software based not a handset. We have recently been playing with fop2 and find it very good. It does require a phone but you could use a softphone installed on the receptionists computer rather than a physical desk phone. Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] Call me now outbound calls in a queue
On Wednesday 03 October 2012, Lenz Emilitri wrote: The problem is that you need to have a process waiting for a free agent and then doing the reschedule. Instead of writing your own, you could try our WombatDialer (that is currently free as in beer, as it is being community tested) to automate such a task. It has a nice HTTP API and it would do exactly what you are looking for. See http://wombatdialer.com/ l. Beware of *anything* free as in beer. Unless it includes the Source Code, place it gently down on the floor and then run, don't walk, in the opposite direction as fast as your legs can carry you. -- AJS Answers come *after* questions. -- _ -- 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] Questions on converting to ConfBridge
Why are you wanting to use CLI commands instead of AMI? The available
AMI actions for ConfBridge can do listing/locking/muting/kicking etc as
you want.
Because I can't easily manually do an AMI command, but instead have to
write code to do it. It's important to me to be able to clean up things
from the command-line if something is stuck or broken.
As for dialplan applications to do the various things - what are you
trying to achieve using them?
And IVR application that people can call into and manipulate people in
conference rooms. Note that this depends on dialplan commands *and*
having a number index for them. It's unclear how I'd do this with
confbridge.
Here's the dialplan I'm using.
exten = 210/_[12]XX,1,NoOp ; Valid if internal.
exten = 210,s,Gosub(Authenticate,s,1()); Else authenticate.
same = n,Mset(C=conferenceha/roomdigits/2digits/0,E=adacore/not-exist)
same = n,Mset(STATS_INC(conf_mgr)=1,__G=conf_op) ; Count the usage.
same = n(r),Macro(Get-Speech,${G},${EFN}adacore/conf_mgr,2,10,100,w)
same = n,GotoIf(${S_T}?${S_T},1:r); Retry or do action.
exten = _[lLK]20Z,1,GotoIf(${MEETME_INFO(parties,20${EXTEN:-1})}?:err)
exten = _L20Z,n,Mset(V=lock,T=locked,E=isadacore/already-locked)
exten = _l20Z,s,Mset(V=unlock,T=unlocked,E=isadacore/already-unlocked)
exten = _K20Z,s,Mset(V=terminate,T=terminated) ; To terminate.
exten =
_[lLK]20Z,n,Set(CFN=adacore/you-want-toadacore/${V}${C}digits/${EXTEN:-1})
exten = _[lLK]20Z,n,Gosub(Is-That-Correct,s,1) ; ... and see if correct.
exten = _[lLK]20Z,n,GotoIf($[${GOSUB_RETVAL}=2]?210,r) ; Retry it not.
exten = _L20Z,n,GotoIf(${MEETME_INFO(lock,20${EXTEN:-1})}?err) ; Bad status.
exten = _l20Z,s,GotoIf(${MEETME_INFO(lock,20${EXTEN:-1})}?:err) ; Likewise.
exten = _K20Z,s,NoOp ; No test needed for termination.
exten = _[lLK]20Z,n,MeetMeAdmin(${EXTEN:-3},${EXTEN:0:1}) ; Perform op.
exten = _[lLK]20Z,n,Set(EFN=${C}digits/${EXTEN:-1}isnowadacore/${T})
exten = _[lLK]20Z,n,Goto(210,r); Ask for another operation.
exten = _[lLK]20Z,n(err),Set(EFN=im-sorry${C}digits/${EXTEN:-1}${E})
exten = _[lLK]20Z,n,Goto(210,r); See if another operation is wanted.
exten = _s20Z,1,Goto(s20${EXTEN:-1}${MEETME_INFO(parties,${EXTEN:-3})},1)
exten = _s20Z.,1,Playback(${C}digits/${EXTEN:3:1}) ; Say that conference ...
exten = _s20Z0,n,Playback(adacore/not-exist) ; ... doesn't exist,
exten = _s20Z1,s,Swift(has one participant); ... or has one person,
exten = _s20Z.,s,Swift(has ${EXTEN:4} participants) ; ... or more.
exten = _s20Z.,n,ExecIf(${MEETME_INFO(lock,${EXTEN:1})}?Swift(and is locked)
exten = _s20Z.,n,Set(M=$[CEIL(MEETME_INFO(activity,${EXTEN:1:3})/60)])
exten = _s20Z[1-9]!,n,Swift(and has been active for ${M} minutes)
exten = _s20Z.,s,NoOp ; In other cases, do nothing.
exten = _s20Z.,n,Goto(210,r) ; Go back for another operation.
exten = _j20Z,1,Set(CFN=you-wish-to-join${C}digits/${EXTEN:-1})
same = n,Gosub(Is-That-Correct,s,1) ; See if correct.
same = n,GotoIf($[${GOSUB_RETVAL}=2]?210,r) ; Retry it not.
same = n,SpeechDestroy; Else free speech channel.
same = n,Goto(${EXTEN:-3},1) ; And go there.
exten =
_[pP]20Z,1,GotoIf($[MEETME_INFO(parties,${EXTEN:1})=0]?s20${EXTEN:-1}0,1)
same = n,Swift(participants in) ; Say the header and ...
same = n,Playback(${C}digits/${EXTEN:3:1}) ; ... conference number.
same = n,ExecIf($[x${G:0:3}=xtmp]?System(rm -f ${GRAMS}/${G}.gram))
same = n,Set(__G=tmp/r${RAND(1,9)}) ; Grammar filename part.
same = n,AGI(conflist.php,${EXTEN:1},${GRAMS}/${G}.gram,${EXTEN:0:1})
same = n,Goto(210,r) ; And go back.
exten = _m20ZXX.,1,Mset(Q=adacore/unmute,OP=adacore/unmuted)
exten = _M20ZXX.,s,Mset(Q=adacore/mute,OP=adacore/muted)
exten = _k20ZXX.,s,Mset(Q=adacore/remove,OP=removed)
exten = _[Mmk]20ZXX.,n,Playback(adacore/you-want-to${Q}) ; Start question.
same = n,Swift(${EXTEN:6}); Say who ...
same = n,Gosub(Is-That-Correct,s,1) ; ... and see if correct.
same = n,GotoIf($[${GOSUB_RETVAL}=2]?210,r) ; Retry it not.
same = n,Set(U=${IF($[${EXTEN:4:1}=0]?${EXTEN:5:1}:${EXTEN:4:2})})
same = n,MeetMeAdmin(${EXTEN:1:3},${EXTEN:0:1},${U}) ; Do operation.
same = n,Swift(${EXTEN:6}); Say name ...
same = n,Playback(${OP}) ; ... and what we did.
same = n,Goto(210,r) ; Go back for another try.
exten = What,1,Set(EFN=adacore/confop_what) ; Say what options are available.
same = n,Goto(210,r) ; And go back and prompt again.
exten = Done,1,Playback(vm-goodbye); Here to hangup.
Here's the grammar:
#ABNF 1.0 UTF-8;
language en-US;
mode voice;
tag-format semantics/1.0.2006;
root $conf_op;
$Operation = lock {out = L;}
| unlock {out = l;}
| (end | kill | terminate) {out = K;}
| join { out = j;}
| [(say | get)] status [of] {
Re: [asterisk-users] Case-sensitivity of Dialplan variables.
While true that most users are probably not programmers, most people administering Asterisk would be system / network admins, correct? System admins and networking admins are used to working in environments such as Linux where variables and file names are case sensitive. I'm in favor of case-sensitivity for the same reason. -- _ -- 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] Receptionist console (software)
Check out isymphony and fop2. -- ringfree.biz Twitter: ringfreebiz 828-575-0030 On Oct 3, 2012, at 5:49 AM, James Mutuku listmut...@gmail.com wrote: Any recommendations I can use. I am looking on having software based not a handset. -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.me.ke -- _ -- 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] DAHDI help please
Thank you!
So, this code in dahdi-base.c should work?
void dahdi_rbsbits(struct dahdi_chan *chan, int cursig)
{
unsigned long flags;
if (cursig == chan-rxsig)
return;
if ((chan-flags DAHDI_FLAG_SIGFREEZE)) return;
spin_lock_irqsave(chan-lock, flags);
switch(chan-sig) {
case DAHDI_SIG_EM:
if (!(cursig DAHDI_XBIT)) {
__dahdi_hooksig_pvt(chan, DAHDI_RXSIG_START);
break;
}
/* Fall through */
Please forgive my ignorance.
As long as folks like you are there to help, folks like me will not remain
ignorant for long!
Thank you again!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Wednesday, October 03, 2012 1:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI help please
On Tue, Oct 02, 2012 at 11:22:31PM -0400, Pat Collins wrote:
Shaun,
To make more sense of the code, I changed
#define DAHDI_XBIT(3 2) to
#define DAHDI_XBIT(0)
Sadly, incoming calls do not work. Not sure exactly how to START or
RING when the RX AB bits are 00 Any ideas?
Thanks again for your help!
The board drivers call dahdi_rbsbits() when they want to report a change in
the state of the RBS bits for a channel. If you look in the code there you
will see where events are generated depending on the signalling type.
I should have pointed out that function in my previous email.
Cheers,
Shaun
--
Shaun Ruffell
Digium, Inc. | Linux Kernel 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] asterisk 1.8 parking not working
- Original Message - From: gincantalupo gincantal...@fgasoftware.com To: asterisk-users@lists.digium.com Sent: Wednesday, October 3, 2012 4:29:28 AM Subject: [asterisk-users] asterisk 1.8 parking not working Hi guys, I've upgraded my pbx from asterisk 1.4 to 1.8 but parking does not work anymore. Tried asterisk-1.8.11.0 and then, after reading about a (fixed) problem in CHANGELOG tried asterisk-1.8.16.0, without success. My features.conf is: [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in, need to INCLUDE this in extensions.conf parkingtime = 45 ; Number of seconds a call can be parked for (default is 45) My extensions.conf is: [inbound] include=parkedcalls ... [outbound] include=parkedcalls as written on the manual (Oreilly ver.3). They seem right to me but when I transfer a call to exten 700 I get an invalid extension message as if Asterisk wouldnt' recognize 700 as a special extrension. Any idea? Nope. Using Asterisk 1.8, I set up a rather limited and contrived test using the following dialplan and a Local channel: [default] exten = 100,1,NoOp() same = n,Answer() same = n,Echo() include = parkedcalls Originating the Local channel into extension 100 and 700 resulted in the channel being parked without any problems: *CLI channel originate Local/100@default extension 700@default -- Executing [100@default:1] NoOp(Local/100@default-;2, ) in new stack -- Executing [100@default:2] Answer(Local/100@default-;2, ) in new stack *CLI -- Executing [700@default:1] Park(Local/100@default-;1, ) in new stack == Parked Local/100@default-;1 on 701 (lot default). Will timeout back to extension [default] s, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls -- Local/100@default-;1 Playing 'digits/7.gsm' (language 'en') -- Executing [100@default:3] Echo(Local/100@default-;2, ) in new stack When you perform a dialplan show, does it show the parkinglots context? [ Context 'parkedcalls' created by 'features' ] '700' = 1. Park() [features] What does a DEBUG log file illustrate when you attempt to place a channel in extension 700? How you are attempting to place the channel in extension 700? -- 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
[asterisk-users] Passing a variable downstream to an IAX server
Apologies if this is a really stupid n00b question, but I don't seem to be able to find an answer anywhere. Is it possible somehow to pass a channel variable to another Asterisk server down an IAX trunk? I have 2 machines; one which is fitted with an ISDN card to make outside calls via an E1 line, and another with a GSM card to make outside calls via the mobile network. The GSM machine gets some calls passed onto it by a command like Dial(IAX2/user:p...@host.cc/number) . This works nicely, but I want to know: is there any way to pass on a variable (already set in the dialplan before the Dial() occurs) to the downstream server? It's a single numeric value that I want to pass on, so I guess in the worst case it could be sent within the actual number e.g. by appending * and the data to the extension number and then using CUT() on the downstream server. But is there a less-ugly method that I am just missing? -- AJS Answers come *after* questions. -- _ -- 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] Call Termination Provider Madness
- Original Message - Have a look at your /etc/asterisk/rtp.conf file. In it you specify the UDP portrange your asterisk will use for RTP traffic. change the rtpstart and rtpend to your needs and set them open in your FW. Do not make the range too small each active call will normally take one RTP channel incoming and one RTP channel outgoing. I have mine set to for example: rtpstart=1 and rtpend=10100. This should be enough for 100 simultanious calls. 2 RTP ports per session (inbound/outbound media)... that would mean 50 simultaneous calls, no? --Tim -- _ -- 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] Peer blocking CDR and recording?
No idea? ): 2012/10/1 Stefan at WPF stefan.at@googlemail.com Today I called some support hotline, for this support hotline no CDR was created, also the call wasn't recorded, though there's a MixMonitor in my dialplan, automatically recording every call. Out of curiosity I set core set verbose 10 in the asterisk console. I then dialed the support hotline again - no single sign of this number being dialed! On every other number I see that I am dialing it / see the dialplan execution! How is that possible? I can provide the number for testing on request, there's a speech computer first where you can simply cancel the call without annoying anyone... -- _ -- 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] Peer blocking CDR and recording?
- Original Message - No idea? ): How about showing your dialplan, and the log or console output from when you make the call? I have a hard time believing this number is special in any way... --Tim -- _ -- 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] Passing a variable downstream to an IAX server
Look in the archives. This was covered sometime in July or August. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Wednesday, October 03, 2012 7:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Passing a variable downstream to an IAX server Apologies if this is a really stupid n00b question, but I don't seem to be able to find an answer anywhere. Is it possible somehow to pass a channel variable to another Asterisk server down an IAX trunk? I have 2 machines; one which is fitted with an ISDN card to make outside calls via an E1 line, and another with a GSM card to make outside calls via the mobile network. The GSM machine gets some calls passed onto it by a command like Dial(IAX2/user:p...@host.cc/number) . This works nicely, but I want to know: is there any way to pass on a variable (already set in the dialplan before the Dial() occurs) to the downstream server? It's a single numeric value that I want to pass on, so I guess in the worst case it could be sent within the actual number e.g. by appending * and the data to the extension number and then using CUT() on the downstream server. But is there a less-ugly method that I am just missing? -- AJS Answers come *after* questions. -- _ -- 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] Call Termination Provider Madness
Op 03-10-12 15:08, Tim Nelson schreef: - Original Message - Have a look at your /etc/asterisk/rtp.conf file. In it you specify the UDP portrange your asterisk will use for RTP traffic. change the rtpstart and rtpend to your needs and set them open in your FW. Do not make the range too small each active call will normally take one RTP channel incoming and one RTP channel outgoing. I have mine set to for example: rtpstart=1 and rtpend=10100. This should be enough for 100 simultanious calls. 2 RTP ports per session (inbound/outbound media)... that would mean 50 simultaneous calls, no? --Tim -- Tim, As Far as I known are the outbound RTP ports determined by the other end. It is also UDP traffic so the inbound stream could be destined for port 1 and the outbound could be coming from port 1. So still 100 simultanious calls. 1 -- XXX (outbound) 1 - XXX (inbound) for one call. Michel. -- _ -- 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] Case-sensitivity of Dialplan variables.
- Original Message -
From: Ira i...@extrasensory.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 3, 2012 3:21:50 AM
Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables.
At 07:59 PM 10/2/2012, you wrote:
While true that most users are probably not programmers, most people
administering Asterisk would be system / network admins,
correct? System admins and networking admins are used to working in
environments such as Linux where variables and file names are case
sensitive.
If someone is moving from a GUI interface to CLI, then they
would/should know that case sensitivity is important and therefore
the change shouldn't pose a problem.
I'm not a system / network admin, at least not for Linux. I have one
Linux machine, it runs Asterisk and Samba. I can usually make
Asterisk do what I want. Samba works but I have little to no idea
why. I run yum update occasionally and I run V11 trunk or whatever
the proper name would be for the development version.
I can think of some situations where case sensitivity could be a problem. I
hope I am not out in left field with my thinking. Asterisk can be found in
companies that have several offices. Asterisk could be used in a cluster.
Asterisk may be administered by many different folks at a company and probably
more than one Asterisk box. If those individuals are expecting the variables
to be case sensitive, it becomes a problem trying to debug problems in the dial
plan. They may not know that an individual in one office is doing things one
way because they are not expecting variables to be case sensitive while another
individual is expecting things to be case sensitive. It really can create a
lot of trouble and confusion in bigger deployments versus a single individual
administering his own box.
If there was a compiler and declared variables then case makes
perfect sense. Without that, I'd never get a C program to work.
I know people want case sensitivity, it's the right way to do it,
but how does it help Asterisk?
This helps Asterisk by following a more or less established standard that
everyone expects. I believe that this case-insensitivity in the dial plan
actually came as a surprise to some who had never stumbled across it before.
Again, those with experience in unix/linux environments have been trained that
variables are case sensitive and they do not have to be programming in C to
know that.
Does anyone have configurations that would be broken by case
insensitivity?
Some people might have broken dial plans and that is why this was brought up on
the list in order to gain attention and feedback. But, it will only break for
the next release. It won't affect current releases. Instead, Mark is planning
on documenting the current behavior on the Asterisk wiki. From what I am
observing so far, it looks like it may only affect a small number of people.
My feeling is that the majority may have already been using variables as if
they were case sensitive already. That was how variables were documented on
the Asterisk wiki... as being case sensitive.
If not, then what is the upside of enforcing case sensitivity?
The upside is that we have consistency. This helps to keep bug reporrts to a
minimum and in my opinion helps the end user not to create problems for
themselves. The example mentioned in the issue being worked on, is say, an
application is expecting the variable ${MIXMONITOR_FILENAME}. A user thinks,
Hey, the dial plan is case insensitive and uses ${mixmonitor_filename} or
${MixMonitor_FileName} to set the file name. They find out that the variable
is being ignored. They later check the variable ${MIXMONITOR_FILENAME} (notice
all uppercase) in the dial plan and it shows him that it is set. They then
think there is a bug in Asterisk... well, the problem is that they didn't set
the variable according to what app_mixmonitor is expecting. The application IS
case sensitive when it comes to variables. So, this is the confusion that can
be caused by having one part of Asterisk be case sensitive and another part of
Asterisk NOT be case sensitive.
I hope this explanation helps those reading this to understand better what is
trying to be resolved here. At least, this is the way I am understanding the
reason for the proposal presented to the list.
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[asterisk-users] SIP, Polycom, Asterisk - VPN
I am trying to configure the following scenario but have failed. Currently, I have an Asterisk box sitting on a Static Public IP address in my office. I have a remote office with 3 Polycom IP335s that are registering back to my local office's publically address Asterisk box. The remote office Polycom phones are getting IP information from an RV042 and using the local ISP for internet access. I want to set up a VPN on the remote side. Has anyone done this? Does it make sense to do this? Thanks, --E -- _ -- 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] Passing a variable downstream to an IAX server
On Wed, Oct 3, 2012 at 5:44 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: Apologies if this is a really stupid n00b question, but I don't seem to be able to find an answer anywhere. Is it possible somehow to pass a channel variable to another Asterisk server down an IAX trunk? I have 2 machines; one which is fitted with an ISDN card to make outside calls via an E1 line, and another with a GSM card to make outside calls via the mobile network. The GSM machine gets some calls passed onto it by a command like Dial(IAX2/user:p...@host.cc/number) . This works nicely, but I want to know: is there any way to pass on a variable (already set in the dialplan before the Dial() occurs) to the downstream server? It's a single numeric value that I want to pass on, so I guess in the worst case it could be sent within the actual number e.g. by appending * and the data to the extension number and then using CUT() on the downstream server. But is there a less-ugly method that I am just missing? -- AJS Answers come *after* questions. -- _ -- 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 Glad I found you asking a question! Check a function IAXVAR. I think Asterisk version matters for it. --Satish Barot -- _ -- 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] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 4:37 AM, Michel Verbraak mic...@verbraak.org wrote: Have a look at your /etc/asterisk/rtp.conf file. In it you specify the UDP portrange your asterisk will use for RTP traffic. change the rtpstart and rtpend to your needs and set them open in your FW. Do not make the range too small each active call will normally take one RTP channel incoming and one RTP channel outgoing. I have mine set to for example: rtpstart=1 and rtpend=10100. This should be enough for 100 simultanious calls. Thanks to everyone for the help in this regard. Its amazing how much I still do not know after nearly 30 years of wrestling with computers. :-) A lack of understanding about the nature of RTP led me to limit traffic inbound from specific IPs which, of course, led to inbound call weirdness. At this point I only have ~40 extensions, so I took Michel's advise and set my RTP range to 1-10100. The default 1 ports was a bit more surface area than I want to expose. Kind Regards, Chris -- _ -- 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] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 7:35 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: At this point I only have ~40 extensions, so I took Michel's advise and set my RTP range to 1-10100. The default 1 ports was a bit more surface area than I want to expose. If you think 100 or 10k RTP ports going to your voice server makes ANY difference in security, you really need to re-think this and study more. Not to be a dick or anything, but really, think about it. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Call Termination Provider Madness
Two points; #1 the more ports you have open, the greater your possible exposure. #2 AFAIK you should have 4 RTP ports for each line you wish to use (although you can tweak some parameter to make it 2) so 1-10100 should actually be 10001-10120. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Wednesday, October 03, 2012 9:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Termination Provider Madness On Wed, Oct 3, 2012 at 7:35 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: At this point I only have ~40 extensions, so I took Michel's advise and set my RTP range to 1-10100. The default 1 ports was a bit more surface area than I want to expose. If you think 100 or 10k RTP ports going to your voice server makes ANY difference in security, you really need to re-think this and study more. Not to be a dick or anything, but really, think about it. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Questions on converting to ConfBridge
On 02/10/12 06:07 PM, Richard Kenner wrote: I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. There also doesn't seem to be a way to lock conferences or mute or kick out users from the dialplan. What am I missing? You're missing the custom DTMF based menus in confbridge.conf, which allows you to set menus separately for admins and users of the conference bridge. This menu allows you to control kicking, muting, etc of users within the conference bridge. No need to manipulate from the dialplan anymore. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] SIP, Polycom, Asterisk - VPN
The easiest way to accomplish this is probably going to be to set up an asterisk server in the remote office and just use IAX to talk between the two boxes. We do VPN here for two phones but I can't really tell you all that you need to know. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Wednesday, October 03, 2012 9:22 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] SIP, Polycom, Asterisk - VPN I am trying to configure the following scenario but have failed. Currently, I have an Asterisk box sitting on a Static Public IP address in my office. I have a remote office with 3 Polycom IP335s that are registering back to my local office's publically address Asterisk box. The remote office Polycom phones are getting IP information from an RV042 and using the local ISP for internet access. I want to set up a VPN on the remote side. Has anyone done this? Does it make sense to do this? Thanks, --E -- _ -- 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 1.8 parking not working
Hi Matthew,
this is the result of dialplan show:
's' =1. NoOp()
[app_dial]
[ Context 'parkedcalls' created by 'features' ]
'700' = 1. Park()
[features]
[ Context 'macro-hash-automon' created by 'pbx_config' ]
and this is the log:
[2012-10-03 16:50:25] VERBOSE[32765] app_dial.c: -- SIP/8-008c
answered SIP/107-008b
-- transferring with ##700 (on the phone) --
[2012-10-03 16:51:02] VERBOSE[32765] res_musiconhold.c: -- Started
music on hold, class 'default', on SIP/107-008b
[2012-10-03 16:51:02] VERBOSE[32765] file.c: -- SIP/8-008c
Playing 'pbx-transfer.gsm' (language 'it')
[2012-10-03 16:51:06] VERBOSE[32765] features.c: -- Blind
transferring SIP/107-008b to '700' (context inbound) priority 1
[2012-10-03 16:51:06] VERBOSE[32765] res_musiconhold.c: -- Stopped
music on hold on SIP/107-008b
[2012-10-03 16:51:06] VERBOSE[32765] res_agi.c: --
SIP/107-008bAGI Script /var/lib/asterisk/script.py completed,
returning 0
[2012-10-03 16:51:06] VERBOSE[32765] pbx.c: -- Executing
[700@inbound:1] NoOp(SIP/107-008b, 22 - Running in inbound at
700) in new stack
[2012-10-03 16:51:06] VERBOSE[32765] pbx.c: -- Executing
[700@inbound:2] Set(SIP/107-008b, AGISIGHUP=no) in new stack
[2012-10-03 16:51:06] VERBOSE[32765] pbx.c: -- Executing
[700@inbound:3] AGI(SIP/107-008b, /var/lib/asterisk/script.py)
in new stack
[2012-10-03 16:51:06] VERBOSE[32765] res_agi.c: -- Launched AGI
Script /var/lib/asterisk/script.py
As you can see 700 is matching with
; Forward any other extension to the standard AGI
exten = _X.,1,Noop(22 - Running in ${CONTEXT} at ${EXTEN})
exten = _X.,n,Set(AGISIGHUP=no)
exten = _X.,n,AGI(/var/lib/asterisk/hash3/bin/exten2.py)
exten = _X.,n,HangUp
and not with the extension 700.
I think this is a bug. It is not the first timeI remember having
reported a bug on asterisk 1.4, same topic...parking
This is what you can find inside Asterisk 1.8 changelog:
--- Fix blind transfer parking issues if the dialed extension is not
recognized as a parking extension.
That's why I upgraded from 1.8.11 to 1.8.16. But I think the issue has
not been solved
Giorgio
On 10/03/2012 01:49 PM, Matthew Jordan wrote:
include = parkedcalls
--
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Re: [asterisk-users] SIP, Polycom, Asterisk - VPN
Thanks for the reply! Why IAX over SIP? In what environment/setup are you using a VPN for the phones? --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, October 03, 2012 10:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SIP, Polycom, Asterisk - VPN The easiest way to accomplish this is probably going to be to set up an asterisk server in the remote office and just use IAX to talk between the two boxes. We do VPN here for two phones but I can't really tell you all that you need to know. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Wednesday, October 03, 2012 9:22 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] SIP, Polycom, Asterisk - VPN I am trying to configure the following scenario but have failed. Currently, I have an Asterisk box sitting on a Static Public IP address in my office. I have a remote office with 3 Polycom IP335s that are registering back to my local office's publically address Asterisk box. The remote office Polycom phones are getting IP information from an RV042 and using the local ISP for internet access. I want to set up a VPN on the remote side. Has anyone done this? Does it make sense to do this? Thanks, --E -- _ -- 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] SIP, Polycom, Asterisk - VPN
IAX uses one port; SIP uses 2-4 per call. We use Polycom 550's to talk to an Asterisk 10.X box. Nothing special on the Asterisk side; just have to get your VPN to talk to the Asterisk network. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Wednesday, October 03, 2012 10:12 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SIP, Polycom, Asterisk - VPN Thanks for the reply! Why IAX over SIP? In what environment/setup are you using a VPN for the phones? --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, October 03, 2012 10:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SIP, Polycom, Asterisk - VPN The easiest way to accomplish this is probably going to be to set up an asterisk server in the remote office and just use IAX to talk between the two boxes. We do VPN here for two phones but I can't really tell you all that you need to know. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Wednesday, October 03, 2012 9:22 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] SIP, Polycom, Asterisk - VPN I am trying to configure the following scenario but have failed. Currently, I have an Asterisk box sitting on a Static Public IP address in my office. I have a remote office with 3 Polycom IP335s that are registering back to my local office's publically address Asterisk box. The remote office Polycom phones are getting IP information from an RV042 and using the local ISP for internet access. I want to set up a VPN on the remote side. Has anyone done this? Does it make sense to do this? Thanks, --E -- _ -- 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] SIP, Polycom, Asterisk - VPN
Danny Nicholas wrote: IAX uses one port; SIP uses 2-4 per call. We use Polycom 550’s to talk to an Asterisk 10.X box. Nothing special on the Asterisk side; just have to get your VPN to talk to the Asterisk network. To be slightly pedantic SIP UDP generally uses a single port (5060). RTP generally uses two ports (one for RTP and one for RTCP). If you have multiple media streams (one audio and one video) those are two RTP sessions, so a total of 4 ports. If you have only an audio stream then that is a total of 2 ports. -- Joshua Colp Digium, Inc. | Senior 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] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 10:45 AM, Carlos Alvarez car...@televolve.com wrote: On Wed, Oct 3, 2012 at 7:35 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: At this point I only have ~40 extensions, so I took Michel's advise and set my RTP range to 1-10100. The default 1 ports was a bit more surface area than I want to expose. If you think 100 or 10k RTP ports going to your voice server makes ANY difference in security, you really need to re-think this and study more. Hi Carlos, I'm speaking of surface area. Ask any general if he would rather have to defend a 1000 mile front or a 1 mile front. You are right that an open port is an open port, but trying keeping the crowd out of 1 doors is *much* harder than trying to keep them out of 100 doors. Kind Regards, Chris -- _ -- 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] SIP, Polycom, Asterisk - VPN
On Wed, Oct 3, 2012 at 10:22 AM, eherr email.eherr9...@gmail.com wrote: I am trying to configure the following scenario but have failed. ** ** Currently, I have an Asterisk box sitting on a Static Public IP address in my office. ** ** I have a remote office with 3 Polycom IP335s that are registering back to my local office’s publically address Asterisk box. ** ** The remote office Polycom phones are getting IP information from an RV042 and using the local ISP for internet access. ** ** I want to set up a VPN on the remote side. ** ** Has anyone done this? Does it make sense to do this? ** ** Thanks, --E Setup OpenVPN between the two sites. A small solid state appliance can handle this easily. Don't worry about IAX2 as was suggested, SIP is just fine. I have used the WRT54G wireless router with one of the Linux firmwares. I have even run Asterisk on these little gems. Some SNOM phones have a Linux/OpenVPN firmware and you can actually bridge the WAN/LAN ports and use the phone as a gateway. Thanks, Steve Totaro -- _ -- 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 1.8 parking not working
- Original Message -
From: gincantalupo gincantal...@fgasoftware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 3, 2012 10:07:22 AM
Subject: Re: [asterisk-users] asterisk 1.8 parking not working
Hi Matthew,
snip
and this is the log:
[2012-10-03 16:50:25] VERBOSE[32765] app_dial.c: -- SIP/8-008c
answered SIP/107-008b
-- transferring with ##700 (on the phone) --
[2012-10-03 16:51:02] VERBOSE[32765] res_musiconhold.c: -- Started
music on hold, class 'default', on SIP/107-008b
[2012-10-03 16:51:02] VERBOSE[32765] file.c: -- SIP/8-008c
Playing 'pbx-transfer.gsm' (language 'it')
[2012-10-03 16:51:06] VERBOSE[32765] features.c: -- Blind
transferring SIP/107-008b to '700' (context inbound) priority 1
[2012-10-03 16:51:06] VERBOSE[32765] res_musiconhold.c: -- Stopped
music on hold on SIP/107-008b
[2012-10-03 16:51:06] VERBOSE[32765] res_agi.c: --
SIP/107-008bAGI Script /var/lib/asterisk/script.py completed,
returning 0
[2012-10-03 16:51:06] VERBOSE[32765] pbx.c: -- Executing
[700@inbound:1] NoOp(SIP/107-008b, 22 - Running in inbound at
700) in new stack
[2012-10-03 16:51:06] VERBOSE[32765] pbx.c: -- Executing
[700@inbound:2] Set(SIP/107-008b, AGISIGHUP=no) in new stack
[2012-10-03 16:51:06] VERBOSE[32765] pbx.c: -- Executing
[700@inbound:3] AGI(SIP/107-008b,
/var/lib/asterisk/script.py) in new stack
[2012-10-03 16:51:06] VERBOSE[32765] res_agi.c: -- Launched AGI
Script /var/lib/asterisk/script.py
As you can see 700 is matching with
; Forward any other extension to the standard AGI
exten = _X.,1,Noop(22 - Running in ${CONTEXT} at ${EXTEN})
exten = _X.,n,Set(AGISIGHUP=no)
exten = _X.,n,AGI(/var/lib/asterisk/hash3/bin/exten2.py)
exten = _X.,n,HangUp
and not with the extension 700.
I think this is a bug. It is not the first timeI remember having
reported a bug on asterisk 1.4, same topic...parking
This is what you can find inside Asterisk 1.8 changelog:
--- Fix blind transfer parking issues if the dialed extension is not
recognized as a parking extension.
That's why I upgraded from 1.8.11 to 1.8.16. But I think the issue
has not been solved
No, this is not a bug. Pattern match extensions within a given context are
given preference over included extensions from other contexts, which the parking
lot extension in context parkinglots is by its very nature.
From extensions.conf.sample (the key portion is the fourth sentence):
; Contexts contain several lines, one for each step of each extension. One may
; include another context in the current one as well, optionally with a date
; and time. Included contexts are included in the order they are listed.
; Switches may also be included within a context. The order of matching within
; a context is always exact extensions, pattern match extensions, includes, and
; switches. Includes are always processed depth-first. So for example, if you
; would like a switch A to match before context B, simply put switch A in
; an included context C, where C is included in your original context
; before B.
Note that if the pattern match is actually in another include (which it does
not seem to be, given that you didn't imply that it was), then the order of
includes matters - so you would want to include your parkinglots context first.
This is also one way that I believe you could define the behavior you want:
[call_extension]
exten = 100,1,NoOp()
same = n,Verbose(Do stuff to a call)
same = n,Hangup()
include = parkedcalls
include = catch_all
[catch_all]
exten = _X.,1,Noop(22 - Running in ${CONTEXT} at ${EXTEN})
exten = _X.,n,Set(AGISIGHUP=no)
exten = _X.,n,AGI(/var/lib/asterisk/hash3/bin/exten2.py)
exten = _X.,n,HangUp
--
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
--
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Re: [asterisk-users] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 8:38 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: I'm speaking of surface area. Ask any general if he would rather have to defend a 1000 mile front or a 1 mile front. You are right that an open port is an open port, but trying keeping the crowd out of 1 doors is *much* harder than trying to keep them out of 100 doors. Your trite comparison is irrelevant in this context. You are not protecting your 100 ports any more or less than 1000 or 10,000. But do as you will, I'll agree to disagree. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Call Termination Provider Madness
A port is not a door if there is nothing listening on the port. Open ports are not a security issue. Stuff running on open ports are. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Wednesday, October 03, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Termination Provider Madness On Wed, Oct 3, 2012 at 8:38 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: I'm speaking of surface area. Ask any general if he would rather have to defend a 1000 mile front or a 1 mile front. You are right that an open port is an open port, but trying keeping the crowd out of 1 doors is *much* harder than trying to keep them out of 100 doors. Your trite comparison is irrelevant in this context. You are not protecting your 100 ports any more or less than 1000 or 10,000. But do as you will, I'll agree to disagree. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Case-sensitivity of Dialplan variables.
On Tuesday 02 Oct 2012, Mark Michelson wrote: [snip] Some of you might be eager to propose a configuration option to decide which it should be. I'm sick of having hundreds of options in Asterisk to slightly tweak the behavior one way or another. This needs to go one way or the other, not be configurable. All dialplans that I've written so far will work fine in a case- sensitive environment. However, I appreciate that there will be legacy dialplans around that are, for one reason or another, case-inconsistent. To expect them all to switch to the new way of doing things immediately is impractical and unfair. So here's the proposal: make case-insensitivity a configuration option for one or two releases. Document the option (both externally and in the configuration file) with large warnings about how switching it on is DEPRECATED and how it will VANISH IN A FUTURE RELEASE. That will suit the people who do not wish to conform (they will not upgrade), the people who want to conform but need time (will have a few months to fix and test while still being able to use the latest Asterisk features) and the people who are already conformant (don't need to do anything). In short, my vote goes for case-sensitivity with a grace period for switching over. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Call Termination Provider Madness
On 10/03/2012 10:46 AM, Eric Wieling wrote: A port is not a door if there is nothing listening on the port. Open ports are not a security issue. Stuff running on open ports are. Do you have some external software listening on those ports when there isn't an active call? Asterisk isn't listening on them. -- _ -- 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] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 8:46 AM, Eric Wieling ewiel...@nyigc.com wrote: A port is not a door if there is nothing listening on the port. Open ports are not a security issue. Stuff running on open ports are. In other words, a million ports with nothing listening is no worse than one with nothing listening. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Call Termination Provider Madness
On Wed, 3 Oct 2012, Chris Nighswonger wrote: You are right that an open port is an open port, but trying keeping the crowd out of 1 doors is *much* harder than trying to keep them out of 100 doors. Especially since the cost of checking those additional 9,900 doors is so high. An open port is not a security issue if nobody is listening. It's not the size of the port range that's important, it's the robustness of the service that is listening. Limiting the number of potential attackers is much more productive than limiting the size of the port range. Not to skewer anybody's homeland, but if you block China, both Koreas, Iran, Iraq, Kuwait and any other geographic area you don't expect legitimate traffic from, the volume of attacks will decrease by orders of magnitude. -- 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] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 9:06 AM, Steve Edwards asterisk@sedwards.comwrote: Not to skewer anybody's homeland, but if you block China, both Koreas, Iran, Iraq, Kuwait and any other geographic area you don't expect legitimate traffic from, the volume of attacks will decrease by orders of magnitude. It's just simple fact. Around 90% of fraud attempts against our network come from that list. And people, please stop trying to use human security to IP port analogies, they make you look foolish. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Case-sensitivity of Dialplan variables.
- Original Message - From: Raj Mathur (राज माथुर) r...@linux-delhi.org To: asterisk-users@lists.digium.com Sent: Wednesday, October 3, 2012 10:49:30 AM Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables. snip So here's the proposal: make case-insensitivity a configuration option for one or two releases. Document the option (both externally and in the configuration file) with large warnings about how switching it on is DEPRECATED and how it will VANISH IN A FUTURE RELEASE. That will suit the people who do not wish to conform (they will not upgrade), the people who want to conform but need time (will have a few months to fix and test while still being able to use the latest Asterisk features) and the people who are already conformant (don't need to do anything). In short, my vote goes for case-sensitivity with a grace period for switching over. From Mark's original e-mail: Some of you might be eager to propose a configuration option to decide which it should be. I'm sick of having hundreds of options in Asterisk to slightly tweak the behavior one way or another. This needs to go one way or the other, not be configurable. I can't overstate how much I agree with this. A configuration option to 'tweak' the behavior in pbx.c is much more likely to introduce problems than solve them. If a clear consensus cannot be reached, I'd err on the side of doing nothing than put in yet another configuration option. -- 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] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 12:09 PM, Carlos Alvarez car...@televolve.com wrote: And people, please stop trying to use human security to IP port analogies, they make you look foolish. -- Carlos Alvarez TelEvolve 602-889-3003 I stand corrected, Carlos. And thank-you for taking time to tell me how foolish I look. It is mostly true that we tend not to see our own foolishness and need to be told about it occasionally. Kind Regards, Chris -- _ -- 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 extension play sound file then connect caller
I am trying to setup a context to take a inbound call, hold the call, connect to an external number, play a sound file to the external number, then connect the inbound caller to the external number. My thought was to accept the call and place them in a parking lot. Then call the external number, play the sound file and connect the inbound caller to the external number. Is this even possible and if so, is this the best approach? Thank you in advance. -- _ -- 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] call extension play sound file then connect caller
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Gary Carr Sent: Wednesday, October 03, 2012 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call extension play sound file then connect caller I am trying to setup a context to take a inbound call, hold the call, connect to an external number, play a sound file to the external number, then connect the inbound caller to the external number. My thought was to accept the call and place them in a parking lot. Then call the external number, play the sound file and connect the inbound caller to the external number. Is this even possible and if so, is this the best approach? Thank you in advance. You might look into FollowMe, especially if you want the external number to have a choice of whether or not to accept the call. A very high level overview is here: http://answers.oreilly.com/topic/2705-asterisk-how-to-use-followme-to-call-a-series-of-phone-numbers/ (though that gave me enough to get started) Thank you, Noah Engelberth MetaLINK Technologies -- _ -- 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] Case-sensitivity of Dialplan variables.
- Original Message - From: Matthew Jordan mjor...@digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, October 3, 2012 12:17:56 PM Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables. From Mark's original e-mail: Some of you might be eager to propose a configuration option to decide which it should be. I'm sick of having hundreds of options in Asterisk to slightly tweak the behavior one way or another. This needs to go one way or the other, not be configurable. I can't overstate how much I agree with this. A configuration option to 'tweak' the behavior in pbx.c is much more likely to introduce problems than solve them. If a clear consensus cannot be reached, I'd err on the side of doing nothing than put in yet another configuration option. I agree that a configuration option is not the solution. I am not seeing what the big deal is. Software changes between major releases. Someone is not going to, or at least they shouldn't if their livelihood depends on it, upgrade their machines without doing the proper preparation for upgrading. That means reading the UPGRADE.txt file and outlining what needs to be done to upgrade their system if there are features they need in the new version or simply want to be on the latest version. Then they should test those changes as well before putting it into production. We are probably a year away from seeing a release for the version of Asterisk where this change would occur. We are two years away from an LTS version of Asterisk. So, I think there would be plenty of time for evaluation and testing to be performed by those affected. Especially, as in the case of what Raj mentioned at the beginning of his prior email, not too many people may even be affected by this change just like he won't be. Michael L. Young (elguero) PS: If you can't tell, I am really for this change and doing so without any configuration options :) -- _ -- 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 module app_konference
Hi all, I am looking for a complete conferencing solution over asterisk (meetme is not fulfill my needs) . I googled a lot and see a lot of stuff on appkonference. Is anybody using this module? Please suggest me and give me some feedback on it. Thanks!!! -- _ -- 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] Call Termination Provider Madness
Le 03/10/2012 18:09, Carlos Alvarez a écrit : On Wed, Oct 3, 2012 at 9:06 AM, Steve Edwards asterisk@sedwards.com mailto:asterisk@sedwards.com wrote: Not to skewer anybody's homeland, but if you block China, both Koreas, Iran, Iraq, Kuwait and any other geographic area you don't expect legitimate traffic from, the volume of attacks will decrease by orders of magnitude. It's just simple fact. Around 90% of fraud attempts against our network come from that list. Well, here in Europe, in the last past monthes, most of attacks are US based. Remember the Amazon S2 ones 2 and half years ago. -- 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] Case-sensitivity of Dialplan variables.
On 02/10/12 09:02 PM, Vladimir Mikhelson wrote: snip First you need to consider compatibility with currently supported packages which include auto-generated dial plans like AsteriskNow, PIAF, etc. If you plan to break their functionality you need to at least coordinate your move with the maintainers. Then you may want to consider backwards compatibility with packages still widely used but not actively supported any more like Trixbox. Maybe not the best example as their WEB site says, This is the current stable release based on Asterisk 1.6. snip I'm not sure that's really the case. This change would be trunk only, and thus the first time it would show up would be Asterisk 12. Because anyone migrating between major versions should already be looking at CHANGES and UPGRADE.txt, this is just another situation where that would be the case. Deployments already based on a released major version would not be affected. +1 to case-sensitivity. It's the right way!(tm) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Case-sensitivity of Dialplan variables.
On 03/10/12 11:49 AM, Raj Mathur (राज माथुर) wrote: In short, my vote goes for case-sensitivity with a grace period for switching over. I disagree. Migrating between major versions should never be something like installing Asterisk 12 over an existing Asterisk 11 (or earlier) system. It should always be a migration between physical (or virtual) boxes. The time to verify your dialplan works in a major release is during the testing phase, not during the omg I installed over my production system! phase. If someone needs to upgrade to a major version, changes as documented in the UPGRADE.txt and CHANGES file would need to be performed anyways, so testing in a staging environment should catch the issues prior to production deployment. Besides, if it was an option, people would just ignore making the changes until the version where the option was no longer available, and we're basically in the same boat as just changing it in the next major version. Consistency for the win!(tm) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Case-sensitivity of Dialplan variables.
On 03/10/12 03:01 PM, Michael L. Young wrote: We are probably a year away from seeing a release for the version of Asterisk where this change would occur. We are two years away from an LTS version of Asterisk. So, I think there would be plenty of time for evaluation and testing to be performed by those affected. Especially, as in the case of what Raj mentioned at the beginning of his prior email, not too many people may even be affected by this change just like he won't be. Well said. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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 module app_konference
On 03/10/12 03:50 PM, pankaj pandey wrote: I am looking for a complete conferencing solution over asterisk (meetme is not fulfill my needs) . I googled a lot and see a lot of stuff on appkonference. Is anybody using this module? Please suggest me and give me some feedback on it. Perhaps you could give a better idea as to what your needs are, and why MeetMe() doesn't fulfill them? Perhaps ConfBridge() in Asterisk 10 or later would fulfill those needs? -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] I can hear my own voice through the headset
Hi all, Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom PSTN gateway - pstn line to telcoi'm using xlite for windows when I make a phone call (sip - outgoing channel),I can hear my own voice so clear. it's very annoying mewhen talking a little loud... any solution? Thanks, Frangky -- _ -- 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
