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| 03:36.21 | midacts | Is there anyway to have a "free" pbx setup, now that Google Voice no longer provides the ability to use their numbers as trunks |
| 03:38.10 | lvlinux | midacts: you can still use GV |
| 03:38.46 | lvlinux | midacts: and there IPKall too. |
| 03:39.01 | lvlinux | <PROTECTED> |
| 03:40.37 | midacts | So i can run something like taht for free from home? |
| 03:42.10 | lvlinux | yes |
| 03:42.31 | lvlinux | I do |
| 03:42.33 | midacts | GV > IPKall? (IPKall's website kinda sketches me out) |
| 03:42.37 | lvlinux | (in addition to paid services) |
| 03:43.35 | lvlinux | you can do both |
| 03:43.41 | lvlinux | or whatever. |
| 03:43.47 | lvlinux | but IPKall is only incoming |
| 03:44.01 | lvlinux | lots of ways to do it |
| 03:44.54 | lvlinux | My home PBX has several voip lines, IPKall, 2 GV lines, a PSTN line, etc. |
| 03:45.46 | midacts | All for free (or are some of your paid services mandatory) |
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| 04:04.53 | lvlinux | I have a mix of free and paid. For various reasons. |
| 04:05.00 | lvlinux | But my GV trunks work perfectly. |
| 04:05.29 | lvlinux | I actually use Yate for GV and just have Asterisk register to Yate via SIP. works great. |
| 04:07.30 | midacts | did GV stop allowing people to use them for SIP trunks? and now they are allowing it again? |
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| 08:00.27 | utrack | Hi! Is there a way to create automatic triggers or something like that in Stasis? I want to know when call is started or ended and route it, however when the call is put on hold MOH should always be played - is it possible? |
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| 08:01.17 | utrack | (I'm complete neophyte, so please tell me if I'm horribly wrong w/ something :) ) |
| 08:05.04 | utrack | as far as I understand, Stasis() in dialplan holds the lock on the channel until call is finished or something |
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| 08:32.10 | utrack | Also, when I try to use ARI POST /channels/id/redirect?endpoint=PJSIP/100 nothing happens, however internal number 100 with hello-world echo is defined; HTTP resp code is 204 |
| 08:34.02 | utrack | 100 is defined in extensions.conf [from-internal] (as per hello world wiki article), endpoint=PJSIP/100@from-internal doesn't work as well |
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| 08:51.10 | utrack | aha, so redirect creates new call instead of connecting current one. First question about automatic start actions in Stasis stands :) |
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| 11:47.08 | gavimobile | is it a bad idea to block all incomming sip traffic on my pbx excluding my tisp? |
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| 11:52.15 | cw1972_ | gavimobile, as long as all your extensions are internal or connect over VPN there should be no problem |
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| 11:52.36 | cw1972_ | I have all SIP blocked except traffic originating from my provider |
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| 12:47.40 | gavimobile | thanks guys |
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| 13:59.11 | z30 | Hi guys, any one knows how to passtrough fax call in Asterisk extension? i have call on sip and E1 PRI |
| 13:59.49 | [TK]D-Fender | "passthrough" where? |
| 14:00.36 | z30 | passtrough from pri line to sip and sip to pri |
| 14:01.39 | z30 | this is my topology : PBX <-----PRI----->Asterisk<----------Sip-------->Cisco<--------PRI--------->PBX |
| 14:01.44 | [TK]D-Fender | No such thing as "Passthrough". That is either going to be raw audio, or require conversion |
| 14:02.07 | z30 | i must keep it ulaw or alaw i know about that |
| 14:02.18 | [TK]D-Fender | I am not aware of * actually encoding T.38 as a gateway, only terminating itt with a fax app |
| 14:02.33 | [TK]D-Fender | If you're keeping it audio.. then there is nothing to do and it's just a call\ |
| 14:02.55 | z30 | my normal call must be G729 and it's work, all call pass with G729 |
| 14:03.15 | z30 | but when fax detect i must change codec on ulaw or alaw and pass it |
| 14:03.51 | [TK]D-Fender | Not going to happen |
| 14:04.18 | z30 | why? cisco did it |
| 14:05.23 | [TK]D-Fender | So? |
| 14:06.01 | z30 | i use this : exten=>fax,1,set(SIP_CODC=ulaw) |
| 14:06.24 | [TK]D-Fender | That doesn't renegotiate a call |
| 14:07.03 | [TK]D-Fender | so unless you are actually ANSWERING it in * first then that's not even going to matter |
| 14:07.04 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 14:07.24 | [TK]D-Fender | and if you are answering it then * is the side getting the carrier request and that is not the time to try STARTING a call back out |
| 14:07.50 | z30 | so what can i do? what are u suggest? |
| 14:08.34 | [TK]D-Fender | Nothing I can suggest.... |
| 14:08.48 | z30 | do i have any chance with iaxmodem?? |
| 14:08.49 | [TK]D-Fender | as far as * is concerned |
| 14:09.00 | [TK]D-Fender | that isn't passthrough |
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| 14:10.55 | z30 | i just want to route it to cisco. :( |
| 14:11.58 | z30 | if i answer call first |
| 14:12.29 | z30 | and wait for fax detection and then change codec, is there any chance? |
| 14:14.38 | [TK]D-Fender | And you're going to delay ALL calls like this? And get charged (where applicable) immediately? |
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| 14:17.31 | [TK]D-Fender | Also if you answer.. then give silence waiting.. then start ringing again it''ll confuse the caller and they might hang up along the way |
| 14:17.52 | z30 | yes you are right |
| 14:18.27 | z30 | so there is no way to passtrough fax with diffrent codec |
| 14:18.36 | z30 | from the call? |
| 14:18.57 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:19.14 | z30 | in other way: i can't have call in g729 and have fax in g711 for same number |
| 14:19.15 | z30 | ? |
| 14:21.07 | [TK]D-Fender | You don't seem to getting the picture |
| 14:21.20 | [TK]D-Fender | Which is unless * ANSWERES the call it'll never KNOW |
| 14:21.32 | [TK]D-Fender | And stop calling it "passthrough" |
| 14:21.34 | [TK]D-Fender | IT ISN"T |
| 14:21.42 | [TK]D-Fender | If * has to LISTEN then that is ANSWERING THE CALL |
| 14:21.54 | [TK]D-Fender | It'That measn STOPPING the call. |
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| 14:25.26 | z30 | i can't know diffrence between call and fax, unless i Answer it |
| 14:26.22 | z30 | and when asterisk find is incomming fax , it run fax extension |
| 14:27.38 | z30 | is there any problem so far? |
| 14:27.59 | [TK]D-Fender | I asked if you were OK with it screwing regular calls... |
| 14:28.25 | [TK]D-Fender | [09:07][TK]D-Fenderand if you are answering it then * is the side getting the carrier request and that is not the time to try STARTING a call back out |
| 14:38.14 | z30 | no regular call must be untouched |
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| 14:40.57 | z30 | let's forget about my work |
| 14:41.16 | z30 | if u want to do something like this |
| 14:41.25 | z30 | how u split calls and fax |
| 14:42.30 | WIMPy | Numbers |
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| 14:47.28 | z30 | numbers is same |
| 14:48.04 | z30 | diffrence is about income is fax or normal call |
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| 15:03.59 | rjvvliet | Hi all, can anyone give any pointers in why asterisk would report this in debug? Asterisk Schedule Dump (94 in Q, 685289 Total, 103 Cache, 204 high-water) |
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| 15:34.49 | rjvvliet | Hi is there anyone that has any pointers in why asterisk would report this in debug? Asterisk Schedule Dump (94 in Q, 685289 Total, 103 Cache, 204 high-water) |
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| 17:03.29 | *** join/#asterisk nixhr (~nix@sunce2.iprojekt.hr) |
| 17:03.33 | nixhr | hi |
| 17:04.15 | nixhr | I'm getting loads of 'SIP/1.2.3.4-00001edc requested media update control 26, passing it to SIP/trunk1-00001edb' messages in the console |
| 17:05.00 | nixhr | Asterisk 11.13.1~dfsg-2+b1 built by buildd @ brahms on a x86_64 running Linux on 2015-01-05 21:34:10 UTC |
| 17:05.33 | nixhr | What could be causing these? |
| 17:11.17 | *** join/#asterisk Aamit (~Amit@103.54.136.32) |
| 17:13.44 | igcewieling | nixhr: they should be harmless. What verbosity level are you running at? |
| 17:31.55 | lvlinux | midacts: Google only provided SIP trunking on GV for about 2 days and then disabled it. It works through Google's version of the Jingle protocol. You have to access it through that, which both Asterisk and Yate can do. |
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| 17:55.17 | nixhr | igcewieling: verbosity is at 15. when I set it to 8 the messages don't appear |
| 17:56.39 | igcewieling | nixhr: It is seldom useful to set verbosity above 3 in normal production |
| 17:56.56 | nixhr | igcewieling: thanks for your help ;) |
| 17:57.35 | nixhr | igcewieling: do you know what these messages mean? I see that it has something to do with reinvites, however, I have directmedia=no set on both trunks |
| 18:00.40 | nixhr | igcewieling: actually, I see those messages on verbosity set to 3 :/ |
| 18:01.10 | nixhr | on 2 they don't show, it seems |
| 18:04.16 | igcewieling | nixhr: I don't run Asterisk 13 |
| 18:05.44 | nixhr | igcewieling: this is actually asterisk 11 |
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| 18:28.44 | igcewieling | nixhr: I've only seen that message a few times. |
| 18:29.11 | igcewieling | what make/model is the SIP endpoint? |
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| 19:01.44 | nixhr | igcewieling: it's MOR softswitch on inbound and outbound side, I don't know the specific version though |
| 19:13.41 | *** join/#asterisk vandyk (~vandyk@191.187.216.136) |
| 19:16.13 | vandyk | I'm running Asterisk 1.8.21.0 and all customers started complaining about call quality. I started the troubleshooting yesterday but with no lucky yet. |
| 19:16.30 | vandyk | When calls are done from local network the issues are really minor |
| 19:16.39 | vandyk | the call seems to be chopped |
| 19:16.57 | [TK]D-Fender | You are outdated on your branch and the entire branch is EOL |
| 19:18.15 | vandyk | Yeah I know that is outdated [TK]D-Fender, but do you think that this could be the reason of this? I started 2 weeks ago |
| 19:18.33 | vandyk | I mean, the complaints started 2 weeks ago |
| 19:18.47 | [TK]D-Fender | quality = network conditions and server load |
| 19:18.56 | vandyk | I first thought that it was the network link |
| 19:19.10 | vandyk | We changed to different link, much more reliable |
| 19:19.18 | vandyk | and it continues the same |
| 19:19.51 | vandyk | and about the server load, the server is not under pressure |
| 19:20.01 | jeffspeff | I'm getting the following error "Unable to open Asterisk database '/var/lib/asterisk/astdb.sqlite3': unable to open database file" when starting asterisk on a new installation. I have verified that exists in that directory. any suggestions? |
| 19:21.11 | vandyk | jeffspeff: are the permissions of folder and file set to the asterisk user? |
| 19:21.30 | jeffspeff | yes |
| 19:21.34 | jeffspeff | i'm running as root |
| 19:22.15 | vandyk | [TK]D-Fender: take a look at top at this moment |
| 19:22.19 | vandyk | http://pastebin.com/qqmf1Cjm |
| 19:22.58 | vandyk | it's very low consume |
| 19:23.34 | [TK]D-Fender | bandwitdh, PL, jitter, etc... |
| 19:24.29 | vandyk | jitter is off |
| 19:24.56 | vandyk | bandwith I believe that is not the problem, we have 20 MB of link |
| 19:25.14 | vandyk | with no more than 20 calls simultaneous |
| 19:26.13 | vandyk | is there a way to check on asterisk, during the call, what could be causing the bad quality? |
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| 19:32.16 | *** join/#asterisk i0x71 (~vitaly@38.117.92.98) |
| 19:33.26 | i0x71 | Hey, Can someone suggest how i can add a bunch of people to the conference, i have a ConfBridge setup, what i would like to do is have a dial plan, when it calls an extension, if it picks up then its transferred or joined to the bridge and so on |
| 19:33.42 | jeffspeff | i still can't figure this out. the astdb.sqlite3 file is there. i can open it with sqlite3 command. i can start asterisk via 'asterisk -cv' and it start fine. if i start with 'service start asterisk' then it errors on opening the astdb file. |
| 19:34.41 | [TK]D-Fender | i0x71: "core show application dial" <- |
| 19:37.20 | jeffspeff | i even just tried 'chmod -R 777 /var/lib/asterisk' and it still won't open the astdb file |
| 19:41.36 | i0x71 | [TK]D-Fender: from what i understand, this will dial the person and sit on the dial untill i hang up, this doesnt auto trasfer the dialed person to the ConfBridge either |
| 19:42.02 | [TK]D-Fender | "this will " <- what part of this? Based on what options? |
| 19:42.35 | [TK]D-Fender | You said "dial" and weren't clear about what means beyond Dial itself (because you did say "in the dialplan") you were thinking about |
| 19:43.44 | i0x71 | What i am asking is how i would setup the automatically macro/script/dialplan, whatever it may be in the sip.conf, so whenever its triggered it would call extensions and (if answered) transfer them to another extension (ConfBridge) |
| 19:44.52 | [TK]D-Fender | If you're loking for ASTERISK itself to call out then that would be "call files", or "AMI Originate" |
| 19:45.08 | *** join/#asterisk chadxz (Adium@nat/digium/x-ylemcsvsybolxfwq) |
| 19:45.14 | [TK]D-Fender | Both well documented in the book and WIKI |
| 19:46.13 | i0x71 | perhaps originate is what i am looking for |
| 19:48.31 | [TK]D-Fender | Either may do. |
| 19:49.13 | *** join/#asterisk bulkorok (~Adium@92.206.230.45) |
| 20:05.02 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 20:08.05 | i0x71 | Can someone explain exactly how originate is supposed to work, Originate(SIP/5000@local,exten,local,1000,1) Does this mean that asterisk calls ext 1000 from ext 5000 ? |
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| 20:11.30 | igcewieling | i0x71: "core show application Originate" was not helpful? |
| 20:12.23 | [TK]D-Fender | no good |
| 20:12.30 | [TK]D-Fender | And that app... is really bad |
| 20:12.51 | [TK]D-Fender | He wants the callouts based on a trigger, not an already existing channel |
| 20:13.01 | [TK]D-Fender | "automated" |
| 20:13.13 | [TK]D-Fender | i0x71: You don't want that |
| 20:13.16 | [TK]D-Fender | AMI Originate |
| 20:13.19 | [TK]D-Fender | NOT the DIALPLAN Application |
| 20:13.22 | [TK]D-Fender | they are not the same |
| 20:15.31 | i0x71 | This is the originate that i am trying to use http://www.voip-info.org/wiki/view/Asterisk+cmd+Originate |
| 20:16.02 | i0x71 | looking at the core show channels, its sitting at originate in the up state |
| 20:16.16 | i0x71 | however none of the phones ring, so i am not even sure what it is doing |
| 20:16.26 | igcewieling | i0x71: do you need to automate this without placing a call or using the CLO? |
| 20:16.36 | igcewieling | s/CLI/CLI |
| 20:17.10 | i0x71 | igcewieling: yes, i was hoping to make automatic, whether its trigger by a dtmf input or anything else |
| 20:18.41 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 20:19.18 | [TK]D-Fender | You do NOT want that |
| 20:19.35 | igcewieling | i0x71: then your ONLY options are .call files or AMI |
| 20:22.35 | i0x71 | So what does that originate application do |
| 20:22.56 | igcewieling | i0x71: allows you to triger by making a phone call. |
| 20:23.08 | i0x71 | so why wouldnt i want that though ? |
| 20:23.24 | i0x71 | what if i want to trigger it by a call to a specific extension |
| 20:23.51 | i0x71 | as in i call conference, setup confbridge.conf to call extension 1234 when i press *9 |
| 20:23.57 | igcewieling | If you want to trigger the call by dialing an extension then the dialplan originate is for you. |
| 20:24.07 | i0x71 | that extension 1234, runs originate on multiple extension and adds them to the bridge |
| 20:24.16 | i0x71 | thats what i was hoping to do |
| 20:25.23 | i0x71 | i would even be fine using an asterisk cli to do this |
| 20:25.37 | i0x71 | having the perl script run execute the commands |
| 20:25.41 | igcewieling | You keep changing your mind. First you say you want to automate it, now you say you don't need to automate it. Which is it? |
| 20:25.44 | *** join/#asterisk hecatae (~hecatae@host-89-240-2-225.static.as13285.net) |
| 20:25.52 | [TK]D-Fender | Dialplan originate = useless |
| 20:25.57 | [TK]D-Fender | it is a BLOCKING app |
| 20:26.25 | igcewieling | i0x71: using asterisk -rx "a command" is the wrong way. |
| 20:26.27 | [TK]D-Fender | Trigger a script that using any other method |
| 20:26.54 | i0x71 | igcewieling: what i want to automate is having asterisk call each one of the extensions and transferring them to a bridge extension, rather than me manually calling an extension, hitting ## to transfer the call to a bridge |
| 20:26.58 | i0x71 | this part i want to automate |
| 20:27.09 | i0x71 | how that process is triggered, whether its auto or manual does not matter |
| 20:27.37 | igcewieling | i0x71: different ways of triggering are done differently. |
| 20:27.49 | igcewieling | I wish you the best of luck. |
| 20:28.46 | [TK]D-Fender | i0x71: "AMI Originate" or "Call Files". That is it |
| 20:29.11 | i0x71 | [TK]D-Fender: is this what you are talking about: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
| 20:29.19 | [TK]D-Fender | yes |
| 20:31.44 | igcewieling | i0x71: also https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files |
| 20:32.32 | igcewieling | and https://wiki.asterisk.org/wiki/display/AST/AMI+Examples |
| 20:35.36 | i0x71 | Thanks |
| 20:48.11 | i0x71 | hmm, not sure what it was but Originate(SIP/1000,exten,local,4200,1) seems to work now |
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| 21:01.03 | [TK]D-Fender | i0x71: That blocks your calling channel, and there has to be one in the first place. So not what you want for "whether its auto or manual does not matter" |
| 21:09.52 | lvlinux | If I have a host that is type=peer, * should match on only IP address right? or no? |
| 21:10.58 | [TK]D-Fender | It will match on IP ... but AUTH still counts |
| 21:11.36 | lvlinux | hmmm, so if there isn't any from user then I need to treat a call from that peer as unauthenticated like a public SIP call? |
| 21:12.22 | hecatae | question, if you had a server still using Asterisk 1.8, would you upgrade to 11, or go straight to 13? |
| 21:12.52 | lvlinux | 13!!! |
| 21:13.10 | hecatae | hi lvlinux, can I ask why? |
| 21:13.22 | igcewieling | hecatae: We went to Asterisk 11 from Asterisk 1.8. Asterisk 13 has too many basic design changes for me. I'm waiting for Asterisk 15 |
| 21:13.26 | lvlinux | I don't think there are that many config differences between 11 and 13 that would be a problem (if any). |
| 21:13.47 | lvlinux | You can still use chan_sip |
| 21:14.04 | igcewieling | lvlinux: basic design changes were required to support pjsip. |
| 21:14.23 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 21:14.33 | igcewieling | I fully support making Asterisk better. Just don't expect me to use the first release after the changes are made. |
| 21:14.45 | igcewieling | (Asterisk 12 doesn't count) |
| 21:14.53 | lvlinux | igcewieling: but the changes didn't really affect chan_sip did they? |
| 21:14.55 | hecatae | I just noticed that Asterisk 14 is not due for release until October 2016, have not seen a gap like that since 1.4 to 1.6 |
| 21:15.08 | igcewieling | lvlinux: Yopu' |
| 21:15.16 | igcewieling | You'd have to ask on #asterisk-dev |
| 21:16.25 | lvlinux | hecatae: try 13. If you have problems with config do 11. Simple enough :-) Should be pretty quick to figure out with a test server. |
| 21:16.38 | igcewieling | From what I understand of Asterisk (relevant up to 1.4 at least) is that chan_sip and Asterisk's RTP support are so intertwined significant changes would have to be done. |
| 21:17.48 | igcewieling | I am responsible for 70+ Asterisk boxes. |
| 21:18.45 | hecatae | igcewieling: that's about 20 more than I look after |
| 21:18.57 | igcewieling | hecatae: and you still switched to Asterisk 13? |
| 21:19.39 | hecatae | igcewieling: no, still got Asterisk 1.8 on some, Asterisk 11 on others, just asking if people are finding 13 stable now |
| 21:20.02 | igcewieling | hecatae: sorry, I was thinking of lvlinux |
| 21:21.37 | igcewieling | hecatae: to see major changes look at the UPGRADE*.txt files, especially the Asterisk 12 ones (where all the changes I'm referring to were made) |
| 21:21.51 | lvlinux | hecatae: I have 13 running on a couple servers and it hasn't missed a beat. I haven't done anything horribly complex with them but 13 is the stable release :-) |
| 21:22.00 | igcewieling | lvlinux: how many servers? |
| 21:22.10 | lvlinux | 13 on only couple |
| 21:22.19 | igcewieling | more correctly Astersik 13 is an LTS release. |
| 21:22.49 | lvlinux | true |
| 21:23.07 | lvlinux | but in fairness it has been out long enough that there is no reason to assume that 11 is more stable. |
| 21:23.15 | lvlinux | IMHO |
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| 21:39.04 | gofunc | Has anyone seen DIS (Distributed Interactive Simulation) packets in voip traffic before? I have one device sending DIS packets before and after registrations |
| 21:41.36 | *** join/#asterisk talntid (~talntid@mail.llcagent.com) |
| 21:41.47 | talntid | Anyone know where I can find a list of websocket protocols? |
| 21:41.50 | talntid | if I don |
| 21:42.14 | talntid | 't specify a protocol, the asterisk console says i didnt. i have found "echo" (useless) and "sip"... not sure how to find others? |
| 21:43.43 | hecatae | talntid: which version of asterisk? |
| 21:44.01 | talntid | 13.6 |
| 21:48.13 | talntid | unless websockets don't emit AMI events, in which case I don |
| 21:48.17 | talntid | 't care about websockets |
| 21:48.18 | talntid | :P |
| 21:51.25 | hecatae | talntid: http://lists.digium.com/pipermail/asterisk-app-dev/2015-February/000661.html < found this helpful |
| 21:53.55 | talntid | roger that, thanks :) |
| 21:55.00 | talntid | I had seen the ARI applications rest API, but when I did GET /applications, it was always empty |
| 21:55.26 | talntid | so I feel like there's nothing to subscribe to in there? unless I misunderstand how it works |
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| 21:59.31 | file | websockets don't emit AMI events |
| 22:00.55 | talntid | what would be the best method, if I want to get all AMI events in the system? to know current status of an endpoint (on call/off call/ringing) ? |
| 22:04.15 | igcewieling | file: do you think that will ever change? |
| 22:04.46 | file | igcewieling, it could... no idea if someone would do it |
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| 22:06.41 | igcewieling | talntid: see the events in a web browser? |
| 22:07.16 | talntid | that would be ideal, sure. |
| 22:07.18 | [TK]D-Fender | talntid, Best method to get all AMI events is to use AMI |
| 22:07.32 | igcewieling | your best bet might be to write an AMI <> websocket proxy. |
| 22:07.51 | talntid | yeah, I was afraid of that. |
| 22:07.53 | [TK]D-Fender | Brought to you by the Redundent Department of Redundency Department. |
| 22:08.28 | talntid | just figured somehow someone would have already solved it well. |
| 22:09.38 | igcewieling | talntid: my solution to a similar issue while ago was to use something like that (websocket<->AMI gateway.) |
| 22:09.47 | rrittgarn | you can use Node to parse AMI events and push to browsers |
| 22:09.54 | rrittgarn | Dan Jenkins wrote a library iirc |
| 22:10.04 | igcewieling | talntid: are you comfortable with PHP? |
| 22:10.16 | talntid | i am |
| 22:10.21 | talntid | and ruby, and JS |
| 22:10.54 | igcewieling | I'm going to check to see if there is any code I can share which might be examples of at least parts of a proxy setup. |
| 22:10.54 | hecatae | talntid: this is about 5 years old, but there is asterclick see https://code.google.com/p/asterisk-asterclick/ |
| 22:11.36 | talntid | yeah I have looked at asterclick, just figured there was something more recent :) |
| 22:11.47 | talntid | guess there isn't, so I'll just write something |
| 22:12.21 | hecatae | talntid: not many from what i see here http://www.voip-info.org/wiki/view/Asterisk+manager+Examples most inactive |
| 22:12.56 | rrittgarn | http://www.slideshare.net/danjenkins/asterisk-html5-and-nodejs-a-world-of-endless-possibilities-14881614 |
| 22:13.15 | talntid | yeah, I have been looking around... just don't like the... "i have to connect to a telnet socket" approach.. i'd prefer asterisk pushes the events, but, oh well :P |
| 22:13.36 | talntid | thanks, I'll watch this tutorial |
| 22:13.44 | talntid | er, not tutorial... slideshow |
| 22:14.23 | rrittgarn | was from one of Dan's Astricon's i think |
| 22:15.31 | talntid | yeah, looks about like what I would be writing |
| 22:15.55 | talntid | thanks :) |
| 22:16.42 | igcewieling | some hacked together code showing AMI waiting for events. http://pastebin.com/yTt0x4cE |
| 22:16.54 | igcewieling | not complete code, but it shows the concept |
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