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| 02:48.07 | bjhaid | hi |
| 02:48.20 | bjhaid | does anyone know why `sip reload` fails on asterisk 13 |
| 02:48.22 | bjhaid | asterisk*CLI> sip reload No such command 'sip reload' (type 'core show help sip reload' for other possible commands) asterisk*CLI> sip show peers No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
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| 04:59.06 | mattwj2002 | hi all |
| 05:02.38 | doop | why hello |
| 05:02.48 | mattwj2002 | hey doop |
| 05:02.50 | mattwj2002 | how are you? |
| 05:03.00 | doop | oh i'm not bad. how's it going |
| 05:03.10 | mattwj2002 | well besides one problem |
| 05:03.17 | mattwj2002 | I can't sleep :( |
| 05:03.49 | doop | well are you here to find a remedy for that problem or |
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| 05:07.32 | mattwj2002 | doop: do you have a barebones asterisk config I could use? |
| 05:07.32 | mattwj2002 | as in an example |
| 05:07.32 | mattwj2002 | :) |
| 05:07.32 | mattwj2002 | I just to make an receive calls off of a sip trunk to start with |
| 05:07.35 | mattwj2002 | actually nevermind doop I google assisted |
| 05:07.36 | mattwj2002 | sure |
| 05:07.46 | mattwj2002 | I have played with asterisk for fun |
| 05:08.00 | doop | ok all your messages just came in simultaneously |
| 05:08.00 | mattwj2002 | but it is nothing I use in my day to day life |
| 05:08.08 | mattwj2002 | O_o |
| 05:08.27 | mattwj2002 | I wonder if Comcast is having issues |
| 05:08.42 | doop | well, if you're a phones person then you'll find a reason to use it always |
| 05:08.48 | mattwj2002 | my coworker/nearby neighbor and friend has reported something similar |
| 05:09.13 | mattwj2002 | I agree doop |
| 05:09.14 | mattwj2002 | :) |
| 05:09.25 | doop | ok |
| 05:09.39 | doop | when are we going to start actually having a conversation |
| 05:09.47 | mattwj2002 | doop: one of my final projects in college back in 2005 was to build an asterisk server! |
| 05:09.49 | mattwj2002 | :) |
| 05:10.02 | mattwj2002 | sure doop |
| 05:10.06 | mattwj2002 | lets talk! |
| 05:11.47 | mattwj2002 | fyi this is what I am looking at http://www.fredshack.com/docs/asterisk.html |
| 05:13.04 | mattwj2002 | doop: may I ask your Internet connection specs? |
| 05:13.12 | doop | why |
| 05:13.18 | doop | what does that have to do with anything |
| 05:13.44 | mattwj2002 | I was just curious about your upload speed and what type of call quality you have |
| 05:13.52 | mattwj2002 | I assume you have asterisk server! |
| 05:13.54 | mattwj2002 | :) |
| 05:14.09 | doop | well then ask directly plz instead of beating about the bush |
| 05:14.26 | mattwj2002 | doop I am sorry |
| 05:14.56 | mattwj2002 | so how is your experience with asterisk sound quality in the past I have had issues |
| 05:14.59 | mattwj2002 | ? |
| 05:15.03 | doop | it's perfect |
| 05:15.06 | mattwj2002 | :D |
| 05:15.13 | mattwj2002 | what codecs do you use? |
| 05:15.18 | doop | ulaw |
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| 05:15.41 | mattwj2002 | what are the names of the wideband ones again? |
| 05:15.48 | doop | i forget |
| 05:15.56 | mattwj2002 | me too |
| 05:15.57 | mattwj2002 | :) |
| 05:16.00 | doop | i can google that for you for only $14.95 |
| 05:16.25 | mattwj2002 | haha |
| 05:16.26 | mattwj2002 | :P |
| 05:16.43 | mattwj2002 | let me guess lmgtfy.com? |
| 05:16.44 | mattwj2002 | ;) |
| 05:18.29 | mattwj2002 | so do you use asterisk as a hobby or in production? |
| 05:19.04 | doop | i've done consulting work |
| 05:19.20 | mattwj2002 | nice |
| 05:21.58 | mattwj2002 | in a different channel |
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| 05:23.05 | mattwj2002 | so one was saying they saw 2,000 active calls going through one server....I guess it had 32 cores that seems impressive...not sure if it was asterisk or something else |
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| 05:27.56 | mattwj2002 | doop: what is up with you? |
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| 05:43.53 | Sean-Der | Does anyone know a way to get the uniqueid used in CEL via 'show channel x' in the asterisk CLI? |
| 05:44.08 | Sean-Der | I have a bunch of hung channel with Rx: BYE |
| 05:44.31 | Sean-Der | But I can't get enough info off the channel, to then debug the dialplan :( and the call volume is way to high to debug |
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| 12:57.43 | sofltech | hi all - my only dahdi channel is showing "in use" and restarting dahdi services doesn't seem to clear it. any suggestions? I know its not in use b/c it can accept incoming calls and answer. |
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| 15:50.23 | sofltech | What is the best way to detect if a channel is available for an outbound call? I only want to be able to make one call at a time on the channel. |
| 15:50.54 | sofltech | I looked at ChanIsAvail but am not sure if that's the right path. |
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| 16:10.38 | mjordan | Channel isn't what you're looking for. A channel is a path of communication between Asterisk and some device (or itself). When a channel is present, you're communicating with something; when not, you aren't. You probably want the device state of the actual device itself, which you can get with the DEVICE_STATE functiono. |
| 16:10.41 | mjordan | function. |
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| 16:22.23 | [TK]D-Fender | sofltech: "core show functions like GROUP" <--- |
| 16:24.19 | mjordan | [TK]D-Fender: I always forget about the GROUP function :-) |
| 16:24.38 | [TK]D-Fender | mjordan: Wishful thinking ;) |
| 16:26.34 | mutilator | so heeey, anyone here use snom870's? |
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| 16:27.40 | mutilator | i've got mine setup to vpn remotely and that works all fine and dandy but i want to get the PC ethernet port to allow access to the network the phone is connected to |
| 16:28.24 | mutilator | does it make more sense to bridge the vpn and dhcp for everything or to NAT that port from the phones VPN IP |
| 16:28.32 | mutilator | if the snom even supports doing NAT... |
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| 16:33.34 | sofltech | thanks guys, I'll check both of those out (device_state and group) |
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| 16:52.18 | Katty | beep. boop. |
| 16:52.21 | Katty | whrrrr. |
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| 16:55.41 | skywalker | hey guys... do you recommend any asterisk "pack" that works as a dialer (call center) or any other daemon/pkg? |
| 17:04.41 | [TK]D-Fender | recommend? No. Outbound war-dialers = evil. |
| 17:04.49 | [TK]D-Fender | However as to what exists... |
| 17:06.03 | skywalker | oh, i see. my boss is running now a call center, either i build this for him or he'll rent it those cloud ones |
| 17:06.16 | skywalker | s/it// |
| 17:06.26 | [TK]D-Fender | The popular "free" one is "vicidial". fugly, semi-(?)maintained |
| 17:06.52 | skywalker | thanks, i'll take a look |
| 17:07.26 | skywalker | i've found this OsDial |
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| 17:26.36 | tuks92 | hello people, can anyone please help me? I want to be able to join two channels together, the first one is created with Zoiper client, and the other one with ARI. The person A wants to talk with person B, so he calls operator with Zoiper or any other client, and when operator answers, he calls B with ARI and joins A and B to be able to talk together. I wanted to add person B to the bridge of A and operator, but I get an error that |
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| 17:52.19 | drmessano | Fire bad, fire bad! |
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| 18:36.48 | Katty | drmessano: FCC |
| 18:37.12 | Katty | drmessano: also hi. |
| 18:37.13 | drmessano | Im neutral on the matter |
| 18:37.16 | drmessano | Hello |
| 18:37.22 | Katty | how're you dear. |
| 18:37.55 | masus | hi, does anyone know how to run an AGI script when agent login is successful, "exten => _123,1,AgentLogin();" |
| 18:38.17 | drmessano | Katty, been better |
| 18:38.26 | drmessano | But taking it day to day |
| 18:38.27 | Katty | drmessano: wanna talk about it? |
| 18:39.06 | tuks92 | hello people, can anyone please help me? I want to be able to join two channels together, the first one is created with Zoiper client, and the other one with ARI. The person A wants to talk with person B, so he calls operator with Zoiper or any other client, and when operator answers, he calls B with ARI and joins A and B to be able to talk together. I wanted to add person B to the bridge of A and operator, but I get an error that |
| 18:39.24 | drmessano | Katty, it's quite boring.. Just changed meds three times in 6 weeks and waiting for the right combo |
| 18:39.39 | Katty | nods |
| 18:39.48 | Katty | it's a shame that it takes so long to figure those chemicals out |
| 18:40.00 | Katty | we just don't know a lot about how those drugs affect brain chemistry yet :/ |
| 18:40.01 | drmessano | So i've been "checked out" for some time now |
| 18:40.37 | drmessano | Pretty much takes all I have to put in my 40 hours and not let cat shit pile to the ceiling at home |
| 18:40.53 | drmessano | So yeah |
| 18:41.22 | file | tuks92, your message was cut off |
| 18:41.29 | drmessano | It's like trying to figure out which Asterisk 1.2 release was best |
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| 18:57.17 | tuks92 | not cut off... it's all... can you help me somehow? can i pm you? |
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| 19:00.55 | mjordan | tuks92: I saw your post on reddit as well. The explanation is a bit lengthy |
| 19:01.25 | mjordan | so, to start, ARI - which is a RESTful API - can control all sorts of things. However, when you try to alter the state of certain things, that thing must be in a state where ARI can safely manipulate it |
| 19:01.40 | mjordan | so, for example, a channel that is floating around in app_voicemail is extremely unsafe to manipulate through ARI |
| 19:01.57 | mjordan | consider what would happen if you initiated a /play operation while something else was playing back media. Hilarity would ensue. |
| 19:02.01 | mjordan | Hilarity => crashes. |
| 19:02.17 | mjordan | As such, the Stasis dialplan application exists to safely hand off channels to ARI to be controlled |
| 19:02.46 | mjordan | when a channel enters into the Stasis dialplan application, that's a sign that Asterisk is saying to the external app - "okay, this is all yours now, you do whatever you want with it" |
| 19:03.28 | mjordan | So, in your case, if O Dials B - through app_dial - you can't just add B to a bridge that A happens to be in inside Stasis |
| 19:03.45 | mjordan | because the B channel - as well as the bridge between O and B - was created by Asterisk, and is not under the control of the external application |
| 19:04.11 | mjordan | if you wanted B to be added to a bridge that A's channel was in, you'd have to have O create an outbound channel to B via a Stasis dialplan application as well |
| 19:04.18 | mjordan | in other words, write your own "Dial" application |
| 19:04.23 | mjordan | as it turns out, that's not super hard to do |
| 19:05.25 | mjordan | A small example in Python - https://github.com/asterisk/ari-py/blob/master/examples/originate_example.py |
| 19:05.54 | mjordan | now, Asterisk/ARI/Stasis is smart enough to handle externally initiated transfers |
| 19:06.08 | mjordan | that is, when a SIP channel driver does a REFER/Replaces |
| 19:06.26 | mjordan | in that case, you'll get notified where the channels end up - either in your Stasis application or outside of it - based on how the transfer occurs |
| 19:06.57 | mjordan | that's typically via an AttendedTransfer/BlindTransfer event, along with StasisEnd/StasisStart events for channels that leave/enter your app due to an externally initiated transfer |
| 19:09.28 | *** join/#asterisk areski (~areski@80.174.128.46.dyn.user.ono.com) |
| 19:14.42 | mjordan | tuks92: does that answer your question? |
| 19:15.36 | tuks92 | i hope, thank you very much for response! i'll try now something |
| 19:15.54 | tuks92 | is there anyway i can contact you if you have will to help me later? |
| 19:16.07 | tuks92 | can call transfer be done with stasis dialplan? |
| 19:16.36 | mjordan | use the app-dev mailing list :-) |
| 19:16.51 | mjordan | http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev |
| 19:17.16 | mjordan | tuks92: call transfers can be initiated externally. To do a DTMF initiated transfer, you have to implement that logic yourself |
| 19:17.21 | mjordan | OTOH, that's actually rather powerful |
| 19:17.37 | mjordan | Say for example you want to have a DTMF key press that automatically pulls a supervisor into a three-way bridge |
| 19:17.48 | mjordan | the notion of a "supervisor" isn't really known in Asterisk features/queues |
| 19:17.55 | mjordan | in your app, however, you can know what that is |
| 19:18.15 | mjordan | and you can implement it to do that - create an outbound channel to the supervisor device, and when it answers, drop it into the bridge with the operator |
| 19:18.21 | mjordan | and it auto switches from 2-party to 3-party :-) |
| 19:18.53 | tuks92 | thank you!! |
| 19:18.57 | tuks92 | i'll try now |
| 19:19.02 | tuks92 | i subscribed to mail list |
| 19:19.42 | mjordan | great! There are plenty of other folks building apps using Asterisk as an engine, so you'll get more feedback there. The archives have some good discussion on this topic as well |
| 19:23.00 | *** join/#asterisk riess82 (~riessma@88-117-84-171.adsl.highway.telekom.at) |
| 19:24.56 | masus | hi, does anyone know how to run an AGI script when agent login is successful, "exten => _123,1,AgentLogin();" |
| 19:25.01 | masus | :( |
| 19:26.20 | [TK]D-Fender | masus: Have some other script monitor AMI or *'s long files |
| 19:27.38 | masus | [TK]D-Fender: Thank You. |
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| 19:55.11 | cyford | i never said anything about virtual routers |
| 19:55.22 | cyford | plus u shouldnt have them in the first place |
| 19:55.36 | cyford | i said asign esxi a private ip |
| 19:55.50 | cyford | not a virtual router |
| 19:56.22 | cyford | lol |
| 19:56.33 | cyford | <PROTECTED> |
| 19:56.51 | cyford | we could of been done a long time ago |
| 19:57.05 | cyford | opps wcc lol |
| 19:58.20 | *** join/#asterisk vomit (~Unknown@unaffiliated/vomit) |
| 20:02.32 | vomit | Hello. We're experiencing problems with some outbound calls and service provider's support responded with the following: http://pastebin.com/G1gwBDxv however we don't know what timeout it may be regarding. Anybody could take a brief look and suggest something? Any input greatly appreciated! |
| 20:04.35 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
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| 20:39.30 | SuperNull | is there a way to pull custom attributes from SIP headers? ex out linksys ATAs have something like P-Station: blahblah=1231;blahblah=1231; |
| 20:45.35 | [TK]D-Fender | SuperNull: "core show functions like SIP" |
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| 21:10.09 | *** join/#asterisk kraut (~kraut@blackhole.netzdeponie.de) |
| 21:10.15 | kraut | hoi |
| 21:10.49 | kraut | what's the preferred language and are questions about the gemeinschaft release welcome? |
| 21:11.24 | robmal | English. Gemeinshaft is great for developing provisioning for openstage. |
| 21:13.28 | kraut | ok. i'm a beginner in asterisk and actually i don't understand how i could handle incoming phone calls via sip trunk to be configured, that it's ringing on sip clients |
| 21:13.53 | kraut | i think i need some points where i need to go into like "calling groups" or something like that, i'm just stuck |
| 21:15.01 | kraut | f.e. it was easy to configure a waiting group and it was reachable from PSTN but how do i configure such phone number i got on the trunk that it will be forwarded to a special sip client? |
| 21:16.52 | *** part/#asterisk madduck (~madduck@debian/developer/madduck) |
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| 21:31.10 | [TK]D-Fender | ~book |
| 21:31.10 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:31.12 | [TK]D-Fender | ^^^^ |
| 21:31.42 | [TK]D-Fender | all call processing = dialplan. Dialplan = extensions.conf. extensions.conf = 90% of what learning Asterisk means. |
| 21:37.55 | *** join/#asterisk CeBe (~CeBe@89.204.130.148) |
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| 21:53.46 | kraut | check check |
| 21:53.59 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:54.00 | kraut | and i just realized that i got a totally outdated installation |
| 21:54.47 | kraut | maybe that's why dialplan is not configurable, there seems to be a bug in the webif |
| 21:55.46 | *** join/#asterisk saint_ (~saint_@gateway/vpn/privateinternetaccess/saint/x-59675695) |
| 21:57.15 | [TK]D-Fender | If you are using a web interface at all this is the wrong place to be for support..... |
| 21:58.31 | *** join/#asterisk cyford (~cyford@c-76-122-73-37.hsd1.ga.comcast.net) |
| 21:59.44 | kraut | i'm fine with both, webif and conf files |
| 21:59.51 | kraut | just needed some hint about the logic |
| 22:00.07 | kraut | that explanation before helped me |
| 22:02.38 | [TK]D-Fender | See how your call comes in. Make an extension to match. Do what you want to do with it. |
| 22:04.15 | *** join/#asterisk prelude2004c (~prelude20@dsl-67-55-28-3.acanac.net) |
| 22:04.19 | prelude2004c | hi everyone good day |
| 22:05.53 | prelude2004c | can anyone help me with parked calls please.. we had asterisk 1.6 and parkedcalls was working fine with i put the include => parkedcalls in the outbound extensions.conf file... now with new asterisk 13 it says " Context 'default' tries to include nonexistent context 'parkedcalls' |
| 22:05.55 | prelude2004c | any ideas? |
| 22:06.30 | rrittgarn | parking got moved out of features in version 12... so its configured differently. do you have the new file set up correctly? |
| 22:06.38 | rrittgarn | i should say features.conf |
| 22:07.05 | prelude2004c | its features.conf ? |
| 22:07.10 | prelude2004c | i will look at a sample |
| 22:07.12 | prelude2004c | wasn't aware of the move |
| 22:07.22 | rrittgarn | https://wiki.asterisk.org/wiki/display/AST/New+in+12 |
| 22:08.37 | prelude2004c | sorry they stopped using features.conf |
| 22:08.43 | prelude2004c | i tried res_parking.conf too |
| 22:08.46 | prelude2004c | looks like similar setup |
| 22:08.51 | prelude2004c | but doesn't seem to like it |
| 22:09.09 | *** join/#asterisk Kunsi (kunsmannf@unaffiliated/kunsi) |
| 22:11.20 | [TK]D-Fender | perhaps you're failing to load the newer modules du to a fixed modules.conf |
| 22:15.47 | kraut | cd |
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| 22:33.53 | Shaan | anyone familiar with gsm gateways? |
| 22:34.12 | Shaan | does anyone know of a manufacturer that makes ones that work with tmobile? |
| 22:41.08 | *** join/#asterisk Elio19 (~elio19@unaffiliated/elio19) |
| 22:45.59 | *** join/#asterisk pjensen00 (~per@ip-64-21-247-189.far.ideaone.net) |
| 22:46.33 | pjensen00 | Hey, is there a limit to the number of web socket connections Asterisk will allow when connecting to listen for ARI events? |
| 22:46.40 | pjensen00 | Asterisk 13 |
| 22:47.58 | *** join/#asterisk Sudravirodhin (~sudraviro@162.218.22.124) |
| 22:48.00 | Sudravirodhin | ~book |
| 22:48.00 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:48.36 | Elio19 | To introduce my self: I saw a nice presentation on Asterisk this past summer given by a guy from Bueno Aires. I'm plan to set up my own (or two) instance of Asterisk, so im sure i will have questions in the coming hours. If anyone was wondering which presentation i saw, it was: http://x.hope.net/schedule.html#areyouread . Hi . |
| 22:48.49 | Sudravirodhin | ~buybook |
| 22:48.49 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
| 22:49.16 | Elio19 | my plan was to just go at it on the console and see what happen |
| 22:50.14 | Elio19 | usually i can force/fight/annoy the computer by tapping on its 'keys' untile it does exactly what i write. |
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| 23:13.19 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 23:40.02 | ChannelZ | ~primer |
| 23:40.02 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
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