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| 00:19.15 | rubio | Hi there.... I have install (from source) asterisk 13 and WebRTC was working on with ICE, now sudendly (or after i don't know what), i have no more audio on WebRTC.... i think this is related to this error: res_rtp_asterisk.c:742 ast_rtp_ice_start: Failed to create ICE session check list: Too many objects of the specified type (PJ_ETOOMANY).... any ideas? thanks! |
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| 01:13.58 | nny | researching a sip based kiosk/entry device to replace a liftmaster setup at a remote gate. Looking for UI customization etc as an option. Algo pops up but looking for other suggestions |
| 01:16.57 | *** join/#asterisk infobot (ibot@69-58-80-29.ut.vivintwireless.net) |
| 01:16.57 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.6.0 (2015/10/09), 11.20.0 (2015/10/09); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 01:17.01 | cmendes0101 | nny: For like contact directory and intercom? |
| 01:20.06 | nny | cmendes0101: yeah, basically have a gate that people go up to and call either a tenant or the office. They can then send dtmf back to activate the gate |
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| 01:21.44 | lvlinux | :q |
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| 01:44.20 | *** join/#asterisk elcontrastador (~textual@c-67-182-43-55.hsd1.ca.comcast.net) |
| 01:50.52 | elcontrastador | What's the best way to pull way, holistically, to pull a snapshot of a given phones status, vmail, calls, etc. from Asterisk? The ARI absolutely perfect, but most things only work for devices within a stasis application. So, should you use ARI/AMI to pull various info on a single device in non-realtime still? |
| 01:51.21 | elcontrastador | oops...didnt mean to send before edit..sorry |
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| 02:47.02 | *** join/#asterisk carrar (~tim@osburn.jp) |
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| 02:50.04 | jpsharp | I'm getting 603 declined when trying to transfer. allowtransfer=yes is set in my sip.conf. Suggestions on whereelse to look? |
| 02:50.31 | ChannelZ | What's the console say? |
| 02:51.34 | jpsharp | Call WQxt0EW45b got a SIP call transfer from caller: (REFER)! |
| 02:51.35 | jpsharp | <--- Transmitting (NAT) to 192.168.128.144:5060 ---> |
| 02:51.35 | jpsharp | SIP/2.0 603 Declined (policy) |
| 02:55.08 | carrar | jpsharp: you reload your sip.conf? |
| 02:55.45 | jpsharp | Yep. |
| 02:55.55 | jpsharp | Let me try a full blown core restart. |
| 02:57.19 | jpsharp | Still same |
| 02:59.20 | jpsharp | Asterisk is sitting behind kamaillo, but I don't think that has anything to do with it. |
| 02:59.46 | carrar | The phone is registered to asterisk not kamaillo right? |
| 03:01.57 | carrar | grep -i allowtransfer sip.conf | grep -i no |
| 03:01.59 | carrar | heh |
| 03:06.36 | jpsharp | the phone is registered against kamaillo. |
| 03:10.50 | carrar | Can you transfer between two phones registered directly to Asterisk and not through kamaillo? |
| 03:12.49 | jpsharp | I'm not sure. But now that I look at things, I don't want clients to be able to transfer calls because then they'll be able to do funny stuff with international calls. |
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| 03:16.54 | carrar | A office without transfer ability is not a very functional office |
| 03:18.03 | ChannelZ | Fences make good neighbors |
| 03:18.27 | carrar | unless you're office is here in Japan |
| 03:18.36 | carrar | where everyone is crammed into the same room |
| 03:18.41 | carrar | and share the same desk |
| 03:18.46 | ChannelZ | And ideally a person shouldn't be able to transfer a call to an extension they couldn't otherwise normally call, so you either trust them or you don't |
| 03:19.44 | carrar | Not much fun if you can do funny stuff |
| 03:19.49 | carrar | can't |
| 03:20.05 | ChannelZ | like Comedian Mail! |
| 03:20.10 | carrar | haha |
| 03:20.23 | carrar | truely funny |
| 03:20.52 | WIMPy | Oh yes. I personally like associated calls. Can be very funny. |
| 03:44.52 | jpsharp | it's not an office environment. It's an a2billing prepaid calling environment. I don't trust the users to not try to screw me. |
| 03:45.39 | WIMPy | And why do you think they wouldn't screw you without transfers? |
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| 03:53.26 | elcontrastador | Has anyone here used Adhearsion as simple way to interact with AMI on Asterisk 13? |
| 03:54.08 | elcontrastador | Simply want to pull all reasonable non-realtime status info on a given handset? |
| 03:54.09 | WIMPy | Oooh. Something I haven't heard about for ages. |
| 03:54.33 | elcontrastador | WIMPy...that's what i'm worried about :-) |
| 03:54.36 | WIMPy | What it non-realtime status? |
| 03:54.41 | WIMPy | is |
| 03:55.07 | elcontrastador | i mean, dont need more info than a snapshot at given time of my web app loading |
| 03:55.23 | elcontrastador | dont need to continuously update |
| 03:56.05 | WIMPy | And what kind of information? |
| 03:57.05 | elcontrastador | registration, line status, vmail qty, etc |
| 03:57.19 | elcontrastador | phone ip... |
| 03:57.37 | elcontrastador | protocol..etc etc |
| 03:57.56 | elcontrastador | whatever i can get basically |
| 03:58.07 | WIMPy | I'm pretty sure it won't be easier if you poll that information. |
| 03:59.14 | WIMPy | Also you lose out on timestamps. |
| 03:59.24 | elcontrastador | i can do most very simply thru syscall to 'asterisk -rx' ... |
| 03:59.49 | elcontrastador | but pretty dirty |
| 04:00.12 | WIMPy | Yes, but is it really easier to parse that information than to listen on AMI and keep track of what's going on in realtime? |
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| 04:02.02 | elcontrastador | no...i'm looking for a good ruby library for AMI...dont want to go off on tangent when i'm waist deep in this webapp |
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| 05:12.17 | JohnnyAsterisk | Can anyone point me in the right direction. I have googled this one to death. Does Asterisk 11 include support for hints on queues? i.e 12001,hint,Queue:12001 Found evidence it was in the beta but doesn't appear to work in the latest release. |
| 05:13.09 | WIMPy | hasn't heard of such a feature, but then I don't hear everything. |
| 05:13.50 | WIMPy | But you can always use cunstom device states. |
| 05:17.25 | JohnnyAsterisk | Issue with custom device state is knowing when last call has left the queue to make it idle again... Easy to trigger it active hard to trigger idle when multiple calls are in queue. If you try to trigger on hangup then you will be changing the state with calls still in queue. Its doable with more logic just easier if it was prepackaged already. |
| 05:18.31 | WIMPy | There are functions to find the number of calls in the queue. So that's just a matter of an ExecIf. |
| 05:19.53 | JohnnyAsterisk | Right thats probably the way to go. If anyone else is aware of anything let me know otherwise custom device state is how I will do it. |
| 05:20.21 | JohnnyAsterisk | Thanks for your help! |
| 05:20.44 | WIMPy | And you know I like AMI... So you could just listen for Queue events and set the devicestate from there. |
| 05:34.20 | igcewieling | looks accusingly at his code. |
| 05:34.58 | WIMPy | What did you tell it to do wrong? |
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| 05:41.03 | igcewieling | WIMPy: I do not know. My code is hiding the bug from me. 8-| |
| 05:42.28 | WIMPy | I have heard of self modifying code. But self hiding code sounds new to me. |
| 05:52.05 | carrar | it's what happens when pixie dust falls on it |
| 05:52.59 | igcewieling | carrar: magical pixie dust is what this script creates. |
| 05:53.06 | carrar | woah |
| 05:54.07 | saratogga | what's a good source for the best prices on IP phones (polycom, yealink, cisco)? Looks like Amazon is pretty much cheaper than all the other "wholesale" sites I find |
| 05:54.46 | carrar | ebay? |
| 05:54.58 | saratogga | yeah but looking for something more consistent |
| 05:55.07 | WIMPy | Now we already have both shops. |
| 05:55.31 | WIMPy | Import directly from China. |
| 05:55.31 | saratogga | both shops? |
| 05:55.35 | carrar | is cisco really a option anymore? |
| 05:55.40 | saratogga | customers ask for it |
| 05:55.54 | saratogga | heck, I got customers who only want Aastra |
| 05:56.03 | saratogga | original Aastras, not mitel stuff |
| 05:56.05 | carrar | Aastra makes a decent phone |
| 05:56.18 | saratogga | yeah |
| 05:56.21 | WIMPy | Amazon and ebey will extinguish all other shops outside of China. |
| 05:57.02 | WIMPy | Unfortunately the also have thoe most unusable sites. |
| 05:57.31 | saratogga | I'm also looking for a good supplier of Cisco SPA122 ATAs |
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| 05:57.38 | saratogga | I need a few hundreds |
| 05:58.19 | JohnnyAsterisk | Just in case anyone else needs this 12000,hint,Queue:12000 |
| 05:58.19 | JohnnyAsterisk | works flawlessly --- The device state identifier is undocumented in the wiki. |
| 05:58.50 | WIMPy | Cool |
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| 06:27.34 | jpsharp | I've got a dozen Cisco 7940s if anyone needs some. :) |
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| 06:33.53 | WIMPy | Not really, no. Or are there interesting hacks for them? Like using them as a game emulator or something... |
| 06:36.34 | carrar | There was a backlight hack out there someplace |
| 06:38.05 | WIMPy | Did they even have backlight? Or was that the hack? |
| 06:38.29 | carrar | never in the 79x0 |
| 06:38.39 | carrar | perhaps the 7970 did |
| 06:38.54 | carrar | but thats a XML based phone and different then the 40/60 model |
| 06:41.33 | elcontrastador | Can someone explain this? https://gist.github.com/elcontrastador/2cd9ee35d22df0d5e175 |
| 06:41.53 | WIMPy | can't see |
| 06:42.15 | elcontrastador | it's public gist...ok...let me check it again |
| 06:42.46 | WIMPy | Yes, but the server requires identification and I won't give it. |
| 06:44.36 | elcontrastador | back to basics...http://pastie.org/10579510 |
| 06:44.44 | elcontrastador | http://pastie.org/10579510 |
| 06:46.14 | WIMPy | I'm somehow missing a context |
| 06:46.54 | elcontrastador | context for extension on line 4... context of problem is on line 1 :-) |
| 06:46.58 | WIMPy | Err, no I'm not. |
| 06:47.38 | elcontrastador | Can't see why it's not finding this extension...must be losing my mind |
| 06:47.55 | WIMPy | But I don't see a hint that could be used. |
| 06:48.27 | WIMPy | So no state defined for that extension. |
| 06:48.29 | elcontrastador | the AMI wiki docs do not describe what the hint is... |
| 06:48.38 | elcontrastador | it's optional too |
| 06:49.04 | elcontrastador | just states if hint present, will use devicestate to check connected device or somethign |
| 06:49.24 | WIMPy | exten => 121,hint,someDivceceState in your dialplan. |
| 06:49.29 | elcontrastador | should, in my mind, at least acknowledge x121's existence |
| 06:50.00 | WIMPy | It doesn't say it doesn't exist, just that the state is unknown. |
| 06:50.03 | elcontrastador | ok, but not sure how to show that as param |
| 06:50.24 | WIMPy | Dialplan. |
| 06:50.49 | elcontrastador | i understand...but as an argument to ExtensionState within AMI |
| 06:51.07 | WIMPy | Extensions don't have a state. But you can define their state in the dialplan as (a combination) of deivestate(s). |
| 06:51.46 | WIMPy | The arguments are extension and context. |
| 06:51.51 | elcontrastador | y |
| 06:52.17 | elcontrastador | got it...that's a response...that was my confusion |
| 06:52.29 | elcontrastador | it's obvious, wasnt thinking straight... |
| 06:52.46 | WIMPy | wonders if you can't query the device states directely. |
| 06:53.45 | elcontrastador | i can query SIPpeerstatus no prob...for that phone at x121 even |
| 06:54.07 | WIMPy | Hmm. Strange. Looks like you have to use the detour through a hint unless it's a sip device. |
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| 06:54.46 | elcontrastador | i'm glad it's strange to u too. |
| 06:54.53 | WIMPy | But strangely tehre seems to be no generic query for devicestates. |
| 06:55.14 | elcontrastador | DeviceStateList is all I see |
| 06:55.39 | WIMPy | Well, as I always say: AMI is telling you everyting you need. If you listen, you don't have to query anything :-) |
| 06:58.03 | WIMPy | (which would be the reason I didn't notice this oddity so far) |
| 06:58.07 | elcontrastador | ok...interesting info |
| 06:58.29 | WIMPy | But there's also the chance that something just isn't documented, but still exists. |
| 06:58.40 | elcontrastador | DeviceStateList does not show the Grandstream phone at x121. Shows the other phones of diff mfr |
| 06:59.12 | elcontrastador | oh...it's there...sorry...i suck tonight...sorry |
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| 06:59.58 | WIMPy | Happens all the time to me if I switch back and forth between IRC and $WHATEVA. |
| 07:01.00 | elcontrastador | ive got debug info and ruby objects dropping into my STDOUT...too much to grok by eyes now |
| 07:07.24 | elcontrastador | WIMPy: Any final thoughts? :-) I'm stumped. I hesitate to call it a bug, as it's my first time with AMI... |
| 07:07.51 | WIMPy | What ewxactely? |
| 07:10.23 | elcontrastador | The pastie...why ExtensionState action is not finding my extension |
| 07:12.44 | WIMPy | Because you have no hint defined in your dialplan. |
| 07:13.58 | WIMPy | Extensions don't have a state by themselves. But you can define their state in the dialplan as (a combination) of deivestate(s) in a hint. |
| 07:14.31 | elcontrastador | i see... |
| 07:14.51 | elcontrastador | i will remedy that now and try again. Thx so much |
| 07:23.31 | elcontrastador | thx WIMPy...it's working... |
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| 09:26.00 | JohnnyAsterisk | Anyone ever run in to a Park Device State always returning INUSE? Even when it most definitely isn't? |
| 09:26.36 | JohnnyAsterisk | exten => _X.,2,Noop(${DEVICE_STATE(park:${EXTEN}@park)}) |
| 09:26.36 | JohnnyAsterisk | exten => _X.,3,Gotoif($["${DEVICE_STATE(park:${EXTEN}@park)}" = "NOT_INUSE"]?4:8) |
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| 10:02.33 | guest9903 | how can I match a variable from the 2nd number to the end? ${VARIABLE}:0:1 is the first character, I suppose ${VARIABLE}:1:0 would give me everything after the first number? |
| 10:02.44 | guest9903 | However, it does not do me well. I have prob done some miss here |
| 10:04.51 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 10:05.37 | guest9903 | Or do I have to use len? |
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| 10:27.08 | rl1 | has anybody have implemented erlang calculator with asterisk |
| 10:27.23 | guest9903 | Another quick one, will _X. match on +3581903002 ? |
| 10:27.31 | rl1 | no |
| 10:27.42 | guest9903 | Is it shame on me to use _.X.? |
| 10:27.43 | rl1 | it will match any digit but not the plus sighn |
| 10:28.08 | rl1 | _+X. this one will match your string |
| 10:28.23 | guest9903 | Will it match both +35800.. and 091904 ? |
| 10:28.27 | rl1 | nope |
| 10:28.39 | rl1 | only ones starting with the plus sign |
| 10:28.49 | guest9903 | Is there a way to match them both? |
| 10:28.57 | rl1 | yeah |
| 10:29.48 | rl1 | _. this one probably will work |
| 10:29.58 | rl1 | but i wouldnt recommend using it |
| 10:30.07 | guest9903 | I will have two seperate matches then |
| 10:30.19 | rl1 | yeah that's better |
| 10:33.09 | rl1 | manager says our client must not exceed 0.5erl load and i dunno how to do that in asterisk.. |
| 10:34.07 | rl1 | how do you even calculate it, there are so many formulas :[ |
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| 11:25.36 | Zogot | MinivmMWI, i should fire that from after MinivmRecord correct? Then on the [100]mailbox=john@example.com and MinivmMWI(john@example.com,0,1,0) should fire it no? |
| 11:32.13 | *** join/#asterisk JanM (5e4fae72@gateway/web/cgi-irc/kiwiirc.com/ip.94.79.174.114) |
| 11:32.58 | JanM | Hey guys. Can you help me solve a problem with enabling ARI in Asterisk? |
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| 11:43.24 | carrar | ~ask |
| 11:43.25 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:47.21 | JanM | I cannot get ARI running. I followed the tutorial on https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI (copying & pasting the configs into the appropriate files). I managed to get the webserver up and running but when trying to connect via wscat it returns: "error: Error: unexpected server response (404)". Asterisk is installed on |
| 11:47.21 | JanM | <PROTECTED> |
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| 11:59.06 | carrar | Are you connecting to the port? |
| 11:59.30 | JanM | I'm connecting to: "ws://192.168.2.19:8088/ari/events?api_key=asterisk:asterisk&app=hello-world" |
| 11:59.50 | carrar | (which is not the example) |
| 12:00.12 | JanM | No, the example is: $ wscat -c "ws://localhost:8088/ari/events?api_key=asterisk:asterisk&app=hello-world" |
| 12:00.12 | JanM | connected (press CTRL+C to quit) |
| 12:00.13 | JanM | > |
| 12:01.06 | carrar | So if you telnet to port 8088 are you getting connected if you use the IP address? |
| 12:02.11 | JanM | Yes |
| 12:02.26 | carrar | So Asterisk is responding to you |
| 12:02.40 | carrar | What do you see in the logs? |
| 12:02.43 | JanM | The connection gets established |
| 12:03.27 | carrar | and for the sake of simplicity did you just go with "asterisk" & "asterisk" as the username and pass? |
| 12:03.59 | carrar | So as not to worry about special characters that might need to be escaped in your URL |
| 12:04.48 | JanM | Yes. After the first customised attempt failed I fell back to simply copying the tutorial (too exclude those kind of problems) |
| 12:05.33 | *** join/#asterisk willianmazzardo (~textual@138.94.78.190) |
| 12:05.37 | carrar | looks simpe enough, I'll try it on my server |
| 12:06.29 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 12:06.29 | willianmazzardo | hi all ... anyone here can tellme how long can be the Dial command? how many characters can be put in Dial string? |
| 12:06.35 | JanM | In the messages log file is a line which looks suspecious to me: "[Nov 25 10:54:38] WARNING[607] loader.c: Error loading module 'res_ari_events.so': /usr/lib/asterisk/modules/res_ari_events.so: undefined symbol: ast_ari_websocket_session_create". I am trying to use the events of ARI, right? |
| 12:10.47 | *** join/#asterisk aness (~aness@2a02:fe0:c310:b3a0:2ca2:99:2a68:a6d8) |
| 12:15.30 | carrar | worked for me ok |
| 12:15.44 | carrar | <PROTECTED> |
| 12:15.47 | carrar | <PROTECTED> |
| 12:16.14 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 12:16.15 | carrar | using wscat --connect "ws://localhost:8088/ari/events?api_key=asterisk:asterisk&app=hello-world" |
| 12:17.02 | carrar | did you get that far? |
| 12:17.50 | JanM | Yes and no. This very line always fails with 404 |
| 12:18.19 | JanM | user@server:/etc/asterisk# wscat --connect "ws://localhost:8088/ari/events?api_key=asterisk:asterisk&app=hello-world" |
| 12:18.19 | JanM | error: Error: unexpected server response (404) |
| 12:19.03 | carrar | and you have the Asterisk web server running on port 8088? |
| 12:19.03 | JanM | The log file: https://www.filepicker.io/api/file/vrOysiSPO7BEvu5yavrQ |
| 12:19.10 | carrar | (http.conf) |
| 12:19.38 | JanM | Yes. If I open it in a browser I get a valid HTTP response (besides the fact that I always get 404s...) |
| 12:20.24 | carrar | [root@tokyo ~]# wscat -c "ws://localhost:8088/ari/events?api_key=asterisk:asterisk&app=hello-world" |
| 12:20.28 | carrar | connected (press CTRL+C to quit) |
| 12:20.30 | carrar | < { "type": "StasisStart", "timestamp": "2015-11-25T04:20:04.351-0800", |
| 12:20.33 | carrar | yeah works great for me |
| 12:20.39 | carrar | folling that guide |
| 12:20.53 | JanM | Which OS are you using? |
| 12:20.58 | carrar | et that far? |
| 12:21.01 | carrar | err |
| 12:21.13 | carrar | Asterisk 13.4.0 on CentOS 7 |
| 12:21.37 | carrar | I should probably upgrade that to 13.6 |
| 12:23.02 | JanM | Okay. So since I'm using Ubuntu (and thus installing Debian packages) it might be a problem there. |
| 12:23.40 | carrar | What I had to do thats not in that doc was in stall wscat on centos |
| 12:23.49 | carrar | yum install nodejs nodejs-options nodejs-commander nodejs-ws |
| 12:25.04 | *** join/#asterisk Parham (~chatzilla@onj3.andrelouis.com) |
| 12:26.01 | Zogot | um im pretty lost on this MinivmMWI, if anyone has any experience with it, id love to have a quick chat |
| 12:26.30 | Parham | Hi all. Does anyone know how I can make the 'hangup' event execute when I use the 'Read' command? |
| 12:28.13 | carrar | JanM, so the simple solution is to just switch your whole system over to CentOS :) |
| 12:28.55 | carrar | ok who has ticket #2 |
| 12:29.49 | JanM | carrar, Yeah, simple me for me to do that, but impossible for my customers. Hopefully their system comes with an Asterisk installation actually working... |
| 12:29.55 | Zogot | carrar: perhaps me :P? im trying to get MinivmMWI working, but im not seeing the notify events come out of it when I reach the application in the dialplan |
| 12:30.28 | carrar | JanM, probably just need to track down whats missing on your system |
| 12:31.27 | carrar | What are you seeing? |
| 12:34.58 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 12:37.55 | JanM | carrar, I'm going to compile Asterisk on my own hoping that only the package is broken. Will report in when done |
| 12:40.01 | *** join/#asterisk membiblio (~membiblio@108.32.57.2) |
| 12:40.35 | *** join/#asterisk acidfoo (18303355@gateway/web/freenode/ip.24.48.51.85) |
| 12:41.31 | JanM | Zogot, I guess "What are you seeing?" was addressed to you |
| 12:49.55 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 12:50.05 | carrar | Parham, MinivmMWI(2100@f20030303,1,1,1) vs MinivmMWI(2100@f20030303,0,0,0) works for me |
| 12:50.20 | carrar | as a example and test |
| 12:50.31 | carrar | MWI light comes on with the 1st one |
| 12:50.38 | carrar | off with the second |
| 12:51.08 | Parham | carrar: Pardon? I didn't get what you meant. |
| 12:51.27 | carrar | Oh sorry |
| 12:51.29 | carrar | that was for Zogot |
| 12:51.44 | Zogot | carrar: hmm where 2100@f2003 is a minivm account? |
| 12:51.52 | carrar | 2100 is my extension |
| 12:52.05 | carrar | f20030303 is the contect in voicemail.conf that extension is in |
| 12:52.08 | carrar | context |
| 12:52.27 | Zogot | hmm, thats even wierder, im trying to get it to work with minivm accounts |
| 12:52.32 | Zogot | so thats normally an email |
| 12:52.49 | Zogot | and your phone is registered at 2100? |
| 12:53.04 | carrar | for voicemail yes, not in sip |
| 12:53.14 | Zogot | mailbox=2100@f2003 |
| 12:53.18 | carrar | yes |
| 12:53.47 | carrar | technically I have |
| 12:53.47 | carrar | vmexten=2100 |
| 12:53.47 | carrar | mailbox=2100@f20030303 |
| 12:53.50 | carrar | but yeah |
| 12:54.10 | Parham | Does anyone know how I can make the 'hangup' event execute when I am waiting for input using the 'Read' command? |
| 12:54.43 | carrar | Parham: use a timeout |
| 12:54.55 | carrar | or hangup |
| 12:56.37 | Parham | carrar: Well, I have a 10 second timeout, but it doesn;t get executed. |
| 12:58.43 | Zogot | carrar: hmm yeh i dont think thats doable for us, we dont use app_voicemail atall |
| 12:59.29 | Parham | carrar: I do get 'User disconnected' in the console, but the hangup event doesn't happen. |
| 12:59.40 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:59.53 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 13:00.16 | carrar | Zogot: worked for me |
| 13:00.20 | carrar | <PROTECTED> |
| 13:00.23 | carrar | <PROTECTED> |
| 13:00.25 | carrar | <PROTECTED> |
| 13:00.28 | carrar | <PROTECTED> |
| 13:00.31 | carrar | read(SOMETHING,,20,5) |
| 13:00.33 | carrar | timeout being 5 |
| 13:00.45 | carrar | 20 digits |
| 13:00.48 | Zogot | carrar: i think i fixed it, thanks :) |
| 13:00.49 | carrar | entered nothing |
| 13:01.21 | carrar | flips to ticket #4 |
| 13:01.50 | carrar | almost bed time |
| 13:01.53 | carrar | Wed Nov 25 22:01:53 JST 2015 |
| 13:01.57 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 13:02.45 | carrar | err woops that was for Parham |
| 13:02.46 | carrar | haha |
| 13:02.50 | carrar | getting people mixed up |
| 13:03.36 | carrar | must be near bedtime when getting names mixed up |
| 13:04.37 | *** join/#asterisk eofster (~eofster@213.61.153.26) |
| 13:04.50 | Parham | carrar: Hehe! |
| 13:05.04 | Parham | carrar: Thanks! Let me try it again... |
| 13:05.16 | Parham | carrar: And if you go to bed meanwhile, have great dreams! |
| 13:08.16 | *** join/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
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| 13:08.51 | *** part/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
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| 13:50.52 | JanM | Zogot: Okay, I compiled Asterisk myself and... still the same problem. Thanks for your help and have a pleasant night :) |
| 13:51.25 | JanM | Oops, *@carrar |
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| 14:30.03 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 15:04.57 | *** join/#asterisk rubio (~rubio@190.19.219.5) |
| 15:05.10 | rubio | Hi there.... I have install (from source) asterisk 13 and WebRTC was working on with ICE, now sudendly (or after i don't know what), i have no more audio on WebRTC.... i think this is related to this error: res_rtp_asterisk.c:742 ast_rtp_ice_start: Failed to create ICE session check list: Too many objects of the specified type (PJ_ETOOMANY).... any ideas? thanks! |
| 15:08.14 | newtonr | What version of Asterisk 13? |
| 15:10.22 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 15:15.56 | newtonr | We had https://issues.asterisk.org/jira/browse/ASTERISK-25516 but we didn't get too far and I'm not sure that is the same issue |
| 15:25.07 | igcewieling | *sigh* Someday I hope to find an e-mail client which does not suck. |
| 15:25.41 | newtonr | I gave up on desktop clients a while back |
| 15:25.56 | igcewieling | newtonr: web based ones are much much worse. |
| 15:26.59 | newtonr | Eh, in my experience about the same, but with less or no crashing. |
| 15:27.14 | newtonr | I've probably lowered my expectations after all these years too |
| 15:28.19 | igcewieling | 10:27 local time. Thunderbird has been downloading messages for about 45 mins now. Don't know why, but it decided to re-download all messages. |
| 15:28.50 | igcewieling | I use Mutt for my personal mail, but at work that isn't very useful. |
| 15:30.18 | newtonr | Yeah I went through outlook, thunderbird, TheBat! and some others. I'm down to just using gmail's web client. |
| 15:30.43 | newtonr | The biggest annoyance is their weird limited filtering. |
| 15:30.51 | igcewieling | We went from outlook to gmail. I hate gmail. now using thundbird with support for gmail. |
| 15:31.56 | igcewieling | I'll re-open the app when I go to bed tonight. It can spend all night working on downloading messages. |
| 15:32.29 | newtonr | Probably want to leave a snack by the computer in case it gets hungry from all that work during the night. |
| 15:49.32 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
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| 15:56.38 | rubio | Hi newtonr, sory for the late responce, asterisk version is: 13.6.0 |
| 15:56.39 | Zogot | igcewieling: im actually a big fan of the gmail ui |
| 15:58.45 | Zogot | igcewieling: i do this, and many standalone mail clients treat the gmail labels as folders |
| 15:58.46 | Zogot | http://i.imgur.com/2UKqdjz.png |
| 15:58.49 | Zogot | which i really dont like |
| 15:59.06 | Zogot | so with this i can open a mail with o, press l, then type 1.1, 9.1 and press enter |
| 15:59.17 | Zogot | press k to go to next mail |
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| 16:29.00 | ks3 | 3 |
| 16:30.06 | ks3 | Ignore me. I apparently keep typing in the wrong window. |
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| 19:22.00 | *** join/#asterisk octo675 (4f9bc289@gateway/web/freenode/ip.79.155.194.137) |
| 19:25.13 | octo675 | hi all |
| 19:25.47 | octo675 | i'm trying to configure pausemonitor, but when press #01 the call hangs up |
| 19:25.53 | octo675 | any help please? |
| 19:26.10 | [TK]D-Fender | Show us the call & configs |
| 19:26.13 | [TK]D-Fender | ~pb |
| 19:26.17 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:26.18 | [TK]D-Fender | ^^^ |
| 19:26.30 | octo675 | ok |
| 19:28.20 | octo675 | http://pastebin.com/4P8Scdk4 |
| 19:29.38 | octo675 | in features.conf i uncomment ;pause.. and ;unpause.. and assign #01 and #02 |
| 19:29.48 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 19:30.34 | octo675 | in console when call hangs tell this: Spawn extension (tecnico, 104, 5) exited non-zero on 'SIP/103-00000002' |
| 19:33.46 | [TK]D-Fender | pastebin the call |
| 19:34.03 | [TK]D-Fender | "sip set debug on" <- we need prrof of which side is killing it |
| 19:34.24 | octo675 | OK |
| 19:39.29 | octo675 | [TK]D-Fender: http://pastebin.com/WgBiiHz2 |
| 19:40.08 | octo675 | i start copy text after pausemonitor key is presed |
| 19:40.21 | *** join/#asterisk salz212 (b6b98be1@gateway/web/freenode/ip.182.185.139.225) |
| 19:46.43 | octo675 | maybe need allowoverlap? |
| 19:53.43 | [TK]D-Fender | show the FULL call |
| 19:54.35 | [TK]D-Fender | You also didn't show your features.conf |
| 19:54.38 | [TK]D-Fender | I asked for configs |
| 19:54.41 | [TK]D-Fender | that's clearly important |
| 19:55.49 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 20:01.37 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 20:02.06 | octo675 | features.conf -> http://pastebin.com/QbRjPCG4 |
| 20:06.34 | octo675 | full call -> http://pastebin.com/US6Y42XE |
| 20:13.06 | salz212 | Hi, does channel.c: Soft-Hanging up channel mean call hanged normally ? |
| 20:18.28 | WIMPy | Unfortunately I'm not sure about that. I think soft hangup means it was requested internally. Otherwise the answer obviousely depends on you definition of "normal". |
| 20:21.19 | [TK]D-Fender | octo675: How long did you talk? |
| 20:21.55 | octo675 | nothing just answer and 2 secons press #001 and call ends |
| 20:23.00 | salz212 | WIMPy: I am trying to find a case of short duration calls hangup issue with DEBUG and pcaps ON. I do not see any re-trans or asterisk err for such calls all I see is this soft hang. I am trying to figure out a reason. |
| 20:24.39 | salz212 | Also regarding ID inside DEBUG[] or Verbose[] for at least one leg it remain same, right? What about rtp packet logger id's should they comply with VERBOSE ID ? |
| 20:24.53 | *** join/#asterisk K0HAX (~K0HAX@shellhost.home.englehorn.com) |
| 20:25.53 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 20:26.04 | WIMPy | Might be easier to listen on AMI to find out what's going on. Have you cranked up verbose? |
| 20:33.04 | salz212 | WIMPy: I am looking at messages and tcpdump. tcpdump sip trace is straight forwards invite 200 and then BYE. But I do not see reason in messages. Here are some filtered logs: http://pastebin.com/S2TDy8Q4 |
| 20:34.16 | WIMPy | Who sent the BYE? |
| 20:34.52 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:36.17 | salz212 | WIMPy Asterisk B# side. |
| 20:41.44 | *** join/#asterisk saratogga (~saratogga@166.170.47.229) |
| 20:42.33 | salz212 | actually asterisk send bye |
| 20:43.22 | octo675 | any know why h extension exten => h,1,Dial(SIP/100) only calls for 1 second and ends automatically? |
| 20:44.14 | igcewieling | octo675: you cannot dial from inside H |
| 20:44.18 | *** join/#asterisk carrotcake (~carrotcak@unaffiliated/carrotcake) |
| 20:44.59 | octo675 | igcewieling: ok thanks, im trying to redirect an incorrect password from Authenticate . try to dial if its wrong to admin extension |
| 20:45.14 | igcewieling | octo675: h is called after the call is hung up. |
| 20:45.22 | igcewieling | you need to handle it in the normal dialplan. |
| 20:46.01 | octo675 | cant use authenticate for that purpose? |
| 20:46.09 | octo675 | if wrong password -> make something? |
| 20:46.57 | igcewieling | octo675: from looking at core show application authenticate you cannot do that. You try READ or something like that and handle it manually. |
| 20:47.10 | octo675 | ok thanks ;) |
| 20:53.56 | *** join/#asterisk w9sh (~sph@c-73-54-187-250.hsd1.ga.comcast.net) |
| 20:59.45 | carrotcake | kinda of newbie here, succefully installed asterisk, I digit 600 on the phone, but nothing happens. what could I check for ? |
| 21:19.45 | *** part/#asterisk carrotcake (~carrotcak@unaffiliated/carrotcake) |
| 21:20.51 | *** join/#asterisk areski (~areski@80.174.128.10.dyn.user.ono.com) |
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| 22:13.36 | f0ner00t | Can someone help me.. I'm using the Lenny Script and getting |
| 22:13.46 | *** part/#asterisk kharwell (kharwell@nat/digium/x-kxnosycxkqwcrgvf) |
| 22:14.25 | f0ner00t | [2015-11-25 14:14:04] WARNING[32563][C-000011a7]: file.c:701 ast_openstream_full: File lenny/Lenny01 does not exist in any format |
| 22:14.28 | f0ner00t | [2015-11-25 14:14:04] WARNING[32563][C-000011a7]: file.c:1017 ast_streamfile: Unable to open lenny/Lenny01 (format (ulaw)): No such file or directory |
| 22:14.32 | f0ner00t | [2015-11-25 14:14:04] WARNING[32563][C-000011a7]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/from-trunk-0000018f for lenny/Lenny01 |
| 22:14.50 | f0ner00t | What is causing this? |
| 22:15.35 | robmal | lenny/Lenny01 |
| 22:18.10 | f0ner00t | What should it be? |
| 22:18.25 | robmal | An existing file. |
| 22:18.50 | f0ner00t | /var/www/html/admin/modules/itslenny/sounds/lenny/Lenny05.ulaw |
| 22:19.09 | f0ner00t | What directory is it looking in? |
| 22:19.44 | robmal | I have no idea, but strace -p `pidof asterisk` might help. |
| 22:21.03 | *** join/#asterisk roler (~textual@unaffiliated/roler) |
| 22:26.23 | f0ner00t | Just got : |
| 22:26.24 | f0ner00t | poll([{fd=-1}, {fd=0, events=POLLIN}], 2, -1) = ? ERESTART_RESTARTBLOCK (To be restarted) |
| 22:26.28 | f0ner00t | --- SIGURG (Urgent I/O condition) @ 0 (0) --- |
| 22:26.30 | f0ner00t | rt_sigreturn(0x17) = -1 EINTR (Interrupted system call) |
| 22:26.33 | f0ner00t | poll([{fd=-1}, {fd=0, events=POLLIN}], 2, -1 |
| 22:26.33 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 22:28.31 | robmal | It seems to be urgent. |
| 22:28.47 | robmal | Still, not the part you were looking for. |
| 22:28.51 | robmal | And don't paste here. |
| 22:28.55 | f0ner00t | OKay |
| 22:28.56 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 22:28.58 | f0ner00t | I'll use pastebin |
| 22:29.03 | f0ner00t | but thats all it showed |
| 22:29.34 | robmal | Because the process is idle. You need to repeat the call you made to make it do anything :-) |
| 22:29.50 | f0ner00t | I did |
| 22:29.51 | f0ner00t | lol |
| 22:29.56 | f0ner00t | thats how I got that output |
| 22:29.59 | robmal | Impossiburu. |
| 22:30.01 | f0ner00t | but I'll try again |
| 22:30.50 | f0ner00t | Same stuff |
| 22:31.50 | robmal | So i might have gone lazy on that `pidof asterisk` part, strace the pid of asterisk that's serving the call. |
| 22:32.01 | robmal | It'll show you where it's looking for Lenny01 |
| 22:37.56 | f0ner00t | The only thing i see serving the asterisk pid is 1862 |
| 22:38.56 | robmal | So, the strace should show everything it does to make the call. |
| 22:41.06 | f0ner00t | http://pastebin.com/fpB7F9vH |
| 22:41.11 | f0ner00t | that is all my asterisk pid |
| 22:41.26 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 22:42.31 | robmal | So strace -p 1862 should show everything your asterisk does to call. |
| 22:43.15 | *** join/#asterisk saratogga (~saratogga@wsip-98-174-232-77.ph.ph.cox.net) |
| 22:43.24 | robmal | Don't listen to the guys at #freepbx, they're giving you the easy way. You need strace and coredumps! |
| 22:44.31 | f0ner00t | lol |
| 22:44.38 | f0ner00t | I saw you on that channel too |
| 22:45.00 | f0ner00t | I showed you all my PIDS for asterisk but it doesn't seem like their is any others except for 1862 |
| 22:45.16 | f0ner00t | I already new it was a directory / file issue |
| 22:45.20 | f0ner00t | that was the simple part |
| 22:45.29 | f0ner00t | I wanted to know why its not calling it |
| 22:45.29 | f0ner00t | lol |
| 22:45.44 | robmal | A strace on asterisk pid has to show you every function it runs to call. |
| 22:47.22 | f0ner00t | .window 3 |
| 22:47.33 | f0ner00t | I'm getting the same thing each time what I pasted above. |
| 22:52.35 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 23:00.05 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 23:08.28 | f0ner00t | Thank you for your help robmal |
| 23:08.36 | MrTAP | f0ner00t: grep astvarlibdir /etc/asterisk/asterisk.conf |
| 23:08.43 | f0ner00t | I still don't know why strace would not help me |
| 23:08.53 | f0ner00t | MrTAP: Its been resolve synlink |
| 23:09.33 | MrTAP | okie dokie |
| 23:10.22 | *** join/#asterisk dorphalsig (be9d89b9@gateway/web/freenode/ip.190.157.137.185) |
| 23:10.41 | dorphalsig | Hey all. |
| 23:11.46 | dorphalsig | Why if I have two sip clients in the same subnet as my asterisk box will I get an Address Family Mismatch notice and no audio? |
| 23:13.25 | dorphalsig | anybody? |
| 23:16.46 | [TK]D-Fender | PROTOCOL |
| 23:17.11 | [TK]D-Fender | And like everything else... you shouldn't be asking us without SHOWING us the call |
| 23:17.19 | dorphalsig | ok |
| 23:19.50 | dorphalsig | http://pastebin.com/bS0Eh03Q |
| 23:19.57 | dorphalsig | There's the call |
| 23:21.25 | dorphalsig | asterisk 192.168.0.2, phone 192.168.0.13, webphone: 192.168.0.11 |
| 23:24.20 | dorphalsig | I keep seeing rtp sent via ice |
| 23:24.42 | dorphalsig | but, why? I mean, I'm in the same subnet |
| 23:25.09 | dorphalsig | and nat = no is set in the two parties involved |
| 23:27.59 | dorphalsig | [TK]D-Fender: Maybe you could show me where I messed up? |
| 23:29.40 | dorphalsig | here are the confs: http://pastebin.com/5EZQpXQi |
| 23:32.14 | [TK]D-Fender | no SIP DEBUG? |
| 23:32.19 | [TK]D-Fender | .... |
| 23:35.19 | *** join/#asterisk theron (~theron@2620:10d:c091:200::a:c026) |
| 23:40.01 | *** join/#asterisk cyford (~support@c-73-137-1-6.hsd1.ga.comcast.net) |
| 23:42.53 | dorphalsig | [TK]D-Fender: http://termbin.com/mmyf |
| 23:42.54 | dorphalsig | the whole thing |
| 23:46.49 | *** join/#asterisk saratogga (~saratogga@wsip-98-174-232-77.ph.ph.cox.net) |
| 23:46.57 | igcewieling | I thought ICE was required for WebRTC? |
| 23:48.54 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 23:49.26 | dorphalsig | AFAIK it is, but I'm in the same subnet and ice seems to be sending some data to the wrong address... |
