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| 08:22.24 | FuriousGeorge | hey all |
| 08:25.43 | FuriousGeorge | is there any protocol out there challenging sip? not that i have any problems with it, but iax wouldn't exist if it was the pinnacle of voip, i assume |
| 08:29.55 | FuriousGeorge | i remember ten years ago wondering if there were some way I could get presence working between sip clients on different servers, and determining it would be a very long and difficult project. i doubt that's changed |
| 08:31.57 | FuriousGeorge | i've been trying to get sip video working lately between a mobile device and pc, and i was eventually able to do it (not using asterisk, though i have a ticket pending with counterpath as to that), but it seemed way more complicated than it should have been |
| 08:32.52 | FuriousGeorge | i'm not blaming *. it was complicated with the other server, and i got it working in that case because there was a fast "instance rollout". it may have worked with * on a compute engine slice too |
| 08:38.01 | FuriousGeorge | and i'd hate to try and get that working with between two clients on two different servers, even on the same platform. then there is SIMPLE, which was never really picked up, probably for good reason... |
| 08:38.17 | FuriousGeorge | isn't the problem the protocol itself? |
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| 09:20.58 | Samot | Presense? |
| 09:21.00 | Samot | Presence? |
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| 09:25.48 | wasanzy | hi |
| 09:26.12 | wasanzy | how possible is to build an intelligent IVR with asterisk? |
| 09:26.24 | wasanzy | can a normal dialplan do that? |
| 09:26.36 | Samot | What do you mean by "intelligent"? |
| 09:28.40 | Maliuta_ | handles phone calls better than Trump? ;) |
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| 09:59.19 | Samot | So what happened with this intelligent IVR question? |
| 10:16.54 | wasanzy | Samot: I was attending to something else |
| 10:17.23 | wasanzy | something like this: http://www.genesys.com/resource-thank-you/genesys-omnichannel-engagement-center-solution |
| 10:17.50 | wasanzy | IVR for CRM... |
| 10:18.03 | Samot | Sure. |
| 10:18.05 | Samot | AGI |
| 10:18.07 | Samot | dialplan |
| 10:18.12 | Samot | Any of those options. |
| 10:18.22 | wasanzy | I see |
| 10:18.23 | wasanzy | thanks |
| 10:18.52 | Samot | They press DTMF, you run AGI or some script via diaplan that can send that response to the CRM |
| 10:18.56 | Samot | Caller ID, etc. |
| 10:19.13 | Samot | I do this for someone now |
| 10:19.45 | Samot | Call comes in, they enter their ticket number and it will send all the details up to that point to the CRM/agent screen... |
| 10:19.48 | wasanzy | I was thinking ael |
| 10:19.54 | Samot | I guess. |
| 10:19.57 | Samot | You could. |
| 10:20.25 | wasanzy | that is great |
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| 11:18.00 | Aljone | Hey all im trying to connect my asterisk to notify my webserver when there is incomming call, i wonder how do i do it ? |
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| 11:41.21 | dadrc | Dialplan hook via AGI, AMI listener or ARI listener |
| 11:49.55 | Samot | Or straight dialplan. If it's just a one way notice then you can do a CURL() |
| 11:50.06 | Samot | Or some other function, depending on the needs |
| 11:54.27 | dadrc | That, too. |
| 11:55.06 | dadrc | so basically however you want :> |
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| 13:03.03 | nibbier | Dial(SIP/123,20,${Parm}) <-- does this basically work, defining the parameters via some variable? |
| 13:04.02 | [TK]D-Fender | of course |
| 13:04.57 | nibbier | ok, great :) |
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| 14:23.19 | scgm11_ | Hi, anyone has any idea why I often see this warning on console although Im not using IAX |
| 14:23.20 | scgm11_ | [Jul 20 10:13:49] WARNING[7305][C-00001e11] chan_iax2.c: Resyncing the jb. last_delay 0, this delay -377017856, threshold 1000, new offset 377017856 |
| 14:23.32 | scgm11_ | Im using cahn_sip asterisk 13.17 webrtc |
| 14:24.25 | [TK]D-Fender | webrts should have nothng to do with IAX |
| 14:25.13 | [TK]D-Fender | If you're not using IAX then go disableit |
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| 14:37.09 | igcewieling | I hate G729 |
| 14:44.12 | Samot | Well it is 2017 |
| 14:44.33 | igcewieling | Congratulations, you know the year |
| 14:44.44 | Samot | I'm good like that. |
| 14:45.10 | Samot | My point, there is no need for g729 these days |
| 14:46.17 | igcewieling | awesome! I'll call Digium and get some free keys. |
| 14:46.36 | Samot | Honestly, why do you need g729 |
| 14:46.45 | igcewieling | 'cause you know, "the licenses expired". |
| 14:49.08 | Samot | You just stated that you hate g729, which I can see why it sucks, so why are you still using it in this day and age? |
| 14:49.25 | Samot | What valid reason is there still to use it? |
| 14:49.48 | igcewieling | Customer needs to push more calls over a T-1 than will work with G726 |
| 14:49.59 | Samot | g711 is just fine. |
| 14:50.02 | Samot | But OK. |
| 14:50.07 | [TK]D-Fender | GSM baby! |
| 14:50.20 | igcewieling | No, it isn't. |
| 14:50.25 | Samot | OK |
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| 14:50.31 | Samot | I believe you. |
| 14:50.52 | igcewieling | With IP UDP overhead you'll not get more than about 20 calls on a data T-1. |
| 14:50.59 | igcewieling | using ulaw or alaw |
| 14:53.36 | Samot | So crappy audio to get what, 2 maybe 3 extra calls out of it? |
| 14:53.45 | Samot | More calls! |
| 14:54.23 | [TK]D-Fender | don't forget to bump to 30ms PR |
| 14:54.39 | [TK]D-Fender | if your jitter/PL can survive it |
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| 14:56.55 | Samot | Well it's a data T-1 so there is also the factor of them using it for that |
| 14:59.14 | igcewieling | [TK]D-Fender: good point. |
| 14:59.47 | igcewieling | Only SIP and RTP go over the T-1. |
| 15:02.45 | igcewieling | If it was my decision they would be running on 5Mpbs+ EOC and G722, but it isn't up to me. |
| 15:03.30 | igcewieling | [TK]D-Fender: the circuit has QoS end to end so jitter is rarely more than 2ms. |
| 15:04.40 | [TK]D-Fender | So sacrifice a little latency and you'll get a lot more payload out of larger packets |
| 15:06.06 | Samot | Why even g722? |
| 15:06.17 | Samot | I mean, if you're transcoding sure. |
| 15:07.07 | [TK]D-Fender | igcewieling, These are calls to independent devices that only do SIP, right? |
| 15:09.11 | igcewieling | [TK]D-Fender: Asterisk to Asterisk. I'm still in therapy from my last experience with IAX2. I just finished installing the extra license. |
| 15:09.25 | [TK]D-Fender | Dude... |
| 15:09.39 | Samot | It's only g722 betewen you and the client |
| 15:09.41 | [TK]D-Fender | Deal with it. Seriously. You're wasting that overhead straight-up |
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| 16:08.35 | igcewieling | [TK]D-Fender: I'll reconsider. 8-| |
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| 16:50.03 | qakhan | [TK]D-Fender my yesterday issue with T-mobile network, i cannot asnwer the call on the device softphone. as you saw i was not getting 200 OK in sip debug |
| 16:50.40 | qakhan | what could be the issue ? |
| 16:54.31 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 16:58.30 | [TK]D-Fender | Them blocking you |
| 16:58.41 | [TK]D-Fender | you advertising the wrong address to them |
| 16:59.05 | [TK]D-Fender | random packet loss (but this would have to be random, not "always" |
| 16:59.12 | qakhan | i fixed that. remember. |
| 16:59.19 | [TK]D-Fender | no, I dont |
| 16:59.28 | [TK]D-Fender | Not sure I care either |
| 16:59.37 | [TK]D-Fender | prove your IP's again |
| 17:00.11 | salviadud | < [TK]D-Fender> Not sure I care either <---- lol |
| 17:00.42 | [TK]D-Fender | salviadud, Stories are what people without actual evidene provide if anything... |
| 17:00.49 | [TK]D-Fender | is too old for story-time |
| 17:00.53 | qakhan | https://pastebin.com/vyY9VxP7 |
| 17:01.48 | *** join/#asterisk hdon (~hdon@68.110.137.138) |
| 17:02.28 | salviadud | [TK]D-Fender, you are correct, when in lack of evidence, stories spring into action. |
| 17:02.48 | [TK]D-Fender | qakhan, What router are you using? |
| 17:02.58 | [TK]D-Fender | We need to prove it isn't screwing with things either |
| 17:03.21 | qakhan | Cisco 3845 |
| 17:03.24 | hdon | hi all :) what are groups and categories for? i have Asterisk The Definitive Guide 4th edition and i don't see them mentioned in the index |
| 17:03.46 | [TK]D-Fender | what "groups"? |
| 17:03.51 | [TK]D-Fender | there are LOTS of differnt groups |
| 17:03.56 | hdon | oh, right :c |
| 17:04.00 | [TK]D-Fender | read the SAMPLE configs <- |
| 17:04.00 | hdon | channel groups |
| 17:04.03 | hdon | ok |
| 17:04.08 | [TK]D-Fender | what KIND of channel groups? |
| 17:04.37 | hdon | what kinds are there? |
| 17:04.52 | [TK]D-Fender | thre is a dialplan function with the word |
| 17:04.58 | [TK]D-Fender | there is a setting in DAHDI configs. |
| 17:05.01 | [TK]D-Fender | completely different thing |
| 17:05.05 | [TK]D-Fender | what are YOU talking about? |
| 17:05.05 | hdon | the first one |
| 17:05.12 | [TK]D-Fender | YOU'RE the one bringing it up |
| 17:05.58 | [TK]D-Fender | You're not going to get a diaplan app getting an entry in the index of a book. |
| 17:06.07 | [TK]D-Fender | There are DOZENS of apps& functions |
| 17:06.18 | [TK]D-Fender | So your methodology is crazy |
| 17:06.26 | hdon | :3 |
| 17:06.30 | [TK]D-Fender | If you're asking about the function, then go read its instructions |
| 17:08.49 | [TK]D-Fender | There are also "pickupgroup", and a few other things |
| 17:10.31 | hdon | i actually don't see the group function beings used in ./configs/samples |
| 17:15.16 | hdon | so, what is Group() function used for? does using Group() have any intrinsic effects, or is it just another mechanism for the programmer to do with as he wants? |
| 17:16.04 | salviadud | checks the wiki... |
| 17:17.17 | *** join/#asterisk miralin (~Thunderbi@91.237.94.1) |
| 17:17.36 | salviadud | I've never used function group to be honest. |
| 17:23.14 | [TK]D-Fender | its only for you and your call counting |
| 17:23.19 | Samot | ^^^^ |
| 17:23.38 | [TK]D-Fender | "core show function GROUP" ,_ not sure what part of "read the instructions" was unclear.... |
| 17:23.57 | [TK]D-Fender | <hdon> i actually don't see the group function beings used in ./configs/samples <- that was for the OTHER ones that are SETTINGS in config files |
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| 17:39.42 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:40.38 | hdon | i want to make an outbound call using call files, and i want to execute some dialplan code when the call is answered. is there an option for this? |
| 17:40.50 | hdon | (this is for a notification program, not an auto dialer :) |
| 17:41.09 | Samot | hdon: It's how call files work. |
| 17:41.17 | [TK]D-Fender | call file IS the option |
| 17:41.19 | [TK]D-Fender | that's what it does |
| 17:41.23 | Samot | Because they do nothing but Originate. |
| 17:41.27 | hdon | hmm |
| 17:41.28 | [TK]D-Fender | Calls the Channel: and sends it INTO the dialplan |
| 17:41.53 | [TK]D-Fender | You don't seem to have read about the very thing you're asking about |
| 17:42.26 | [TK]D-Fender | Call comes IN to your server * tries to auth it if required and then leads to dialplan. |
| 17:42.43 | [TK]D-Fender | Call File / Originate = tells * to call OUT and then dump them into the dialplan |
| 17:42.51 | [TK]D-Fender | Same end result, different beginning |
| 17:48.43 | qakhan | [TK]D-Fender any though on my quest |
| 17:48.48 | qakhan | question* |
| 17:49.15 | [TK]D-Fender | You didn't ASK one |
| 17:52.26 | qakhan | [TK]D-Fender my yesterday issue with T-mobile network, i cannot asnwer the call on the device softphone. as you saw i was not getting 200 OK in sip debug |
| 17:52.32 | qakhan | what could be the issue ? |
| 17:52.44 | [TK]D-Fender | I answered that already |
| 17:53.08 | [TK]D-Fender | hours ago |
| 17:53.27 | [TK]D-Fender | hor* |
| 17:53.30 | [TK]D-Fender | hour* |
| 17:54.24 | qakhan | you asked me what router i am using |
| 17:54.35 | qakhan | i said cisco 3845 |
| 17:54.53 | [TK]D-Fender | <qakhan> what could be the issue ? |
| 17:54.53 | [TK]D-Fender | * salviadud (~ralfalfa@189-211-190-134.static.axtel.net) has joined |
| 17:54.53 | [TK]D-Fender | <[TK]D-Fender> Them blocking you |
| 17:54.53 | [TK]D-Fender | <[TK]D-Fender> you advertising the wrong address to them |
| 17:54.53 | [TK]D-Fender | <[TK]D-Fender> random packet loss (but this would have to be random, not "always" |
| 17:55.03 | [TK]D-Fender | those were th 5 lines from the point you asked |
| 17:55.08 | [TK]D-Fender | IMMEDIATELY after |
| 17:55.23 | [TK]D-Fender | If you're not going to read the answers don't ask the question |
| 17:55.49 | qakhan | i said i fixed the issue yesterday, remember |
| 17:55.56 | lorsungcu | [TK]D-Fender: spicy today! |
| 17:56.02 | qakhan | https://pastebin.com/vyY9VxP7 |
| 17:56.06 | [TK]D-Fender | 3 things <- |
| 17:56.13 | [TK]D-Fender | that was my answer |
| 17:56.15 | [TK]D-Fender | the end |
| 17:56.20 | [TK]D-Fender | that was what I had to offer |
| 17:56.40 | qakhan | ok |
| 17:56.48 | [TK]D-Fender | <[TK]D-Fender> <[TK]D-Fender> you advertising the wrong address to them <- this was ONE thing |
| 17:56.57 | [TK]D-Fender | I said 1 thing BEFORE tha, and 1 thing AFTER that |
| 17:57.20 | qakhan | ok |
| 17:57.30 | qakhan | thank you |
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| 18:39.12 | hdon | hi all :) do call files in the outgoing queue support calling multiple parties using ampersand as delimiter in the Channel: line? |
| 18:39.35 | Samot | No. |
| 18:39.57 | Samot | Channel: Tech/Exten@Context |
| 18:40.18 | Samot | Once you are there, you can do what you want. |
| 18:40.27 | hdon | thanks Samot |
| 18:40.37 | Samot | It sends them to exten,context,priority |
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| 18:41.46 | Samot | It just defaults to 1 as the priority in the Channel: setting |
| 18:49.40 | *** join/#asterisk scgm11_ (~scgm11@r186-52-124-7.dialup.adsl.anteldata.net.uy) |
| 18:50.34 | [TK]D-Fender | <hdon> hi all :) do call files in the outgoing queue support calling multiple parties using ampersand as delimiter in the Channel: line? <- dial a Local channel to do the actual dial dirty work |
| 19:15.29 | *** join/#asterisk Typhon (~Typhon@ipservice-092-211-209-241.092.211.pools.vodafone-ip.de) |
| 19:24.14 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 19:29.27 | tuxd00d | Is there another way to clear a hung call? (âAutodestruct on dialog âXXXX@192.168.1.192' with owner SIP/AAAAA-XXXXX in place (Method: BYE). Rescheduling destruction for 10000 msâ - likely due to a full hard drive for a few seconds). âchannel request hangup SIP/AAAAA-XXXXXâ does not clear it. |
| 19:40.33 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |
| 19:56.06 | [TK]D-Fender | AMI Redirect off a cliff |
| 19:56.50 | tuxd00d | Iâm a bit rusty on my AMI⦠let me look into that. |
| 20:09.41 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 20:13.16 | darkunderlord | dammit. corosync has to be on the same IP network. Guess I"ll be sharing state/presence via Jabber |
| 20:15.03 | lorsungcu | darkunderlord: what? |
| 20:16.06 | darkunderlord | I have geographically dispersed (close in speed tho) servers I need to share state between |
| 20:16.22 | darkunderlord | I thought I could use corosync, but that' only meant for machine on the same ip subnet |
| 20:16.50 | darkunderlord | no I guess now my only choice to share geographically and between 11 and 13 servers is XMPP? |
| 20:21.17 | darkunderlord | all I want is to share BLF between two or three asterisk servers. |
| 20:22.09 | Samot | You can. |
| 20:22.19 | Samot | You just have to update the state between them. |
| 20:36.21 | darkunderlord | then how? do I have to use xmpp? |
| 20:36.33 | darkunderlord | and they aren't all on 13, some are 11 |
| 20:37.08 | Samot | Asterisk uses hints for this |
| 20:37.24 | Samot | So take what you know about Presence/Subscriptions and put it aside. |
| 20:37.40 | darkunderlord | that's what I actually want to share, is hints |
| 20:37.56 | darkunderlord | I had no issue from doing one phone on one server to another phone on another server in Asterisk 11 |
| 20:38.05 | Samot | You will need to use AMI or ARI to monitor those hints |
| 20:38.18 | Samot | Then you can use the same to update the other servers. |
| 20:38.21 | darkunderlord | well ARI isn't in 11, so I guess AMI? |
| 20:38.27 | Samot | Yes |
| 20:39.26 | file | darkunderlord: was the phone configured on both servers? |
| 20:39.48 | file | that is: was there a sip friend/peer/user on both servers that the phone asking for hint information could authenticate as? |
| 20:39.53 | *** join/#asterisk juvenal (~juvenal@189-18-34-155.dsl.telesp.net.br) |
| 20:39.55 | darkunderlord | file: no |
| 20:40.04 | file | then how were you sharing information in 11? |
| 20:40.08 | darkunderlord | I have realtime on both |
| 20:40.16 | file | ah |
| 20:40.19 | darkunderlord | it just worked, Josh said he dind't know how it did either |
| 20:40.34 | darkunderlord | but the realtime is in it's own database on each |
| 20:40.43 | darkunderlord | so they are separate, but somehow it worked |
| 20:41.04 | file | if you can determine how it actually worked then you may be able to recreate it |
| 20:41.16 | darkunderlord | a server at my remote location coudl have a button assigned to someone at my current site. I don't need the callgroup/pickup, just the light for the states |
| 20:41.21 | file | otherwise you have to use the XMPP mechanism, or do as Samot was saying and create your own layer |
| 20:41.42 | darkunderlord | lol you're Josh :D |
| 20:41.42 | Samot | Are you using FQDNs with RealTime |
| 20:41.49 | file | I am. |
| 20:42.14 | darkunderlord | I think the allow_subscribe might have let me with chan_sip, but not sure |
| 20:42.28 | Samot | darkunderlord: Are you using domains? |
| 20:42.37 | darkunderlord | but now I'm on pjsip with 13 for a coupel sites |
| 20:43.05 | darkunderlord | domains? We're all on the same domain, connected via fast speeds. |
| 20:43.11 | darkunderlord | not sure what you mean by domains |
| 20:44.03 | darkunderlord | i'm using realtime with MySql from each server to another db server for each |
| 20:44.47 | darkunderlord | I could probably create an account in realtime on the other one too, but I wouldn' thave the phone register to both, At least I don't think i would right? |
| 20:45.48 | darkunderlord | I have them all connected via IAX2 and trunked together. But that's just for call routing |
| 20:47.39 | Samot | Well in order to subscribe you have to auth |
| 20:48.23 | Samot | I'm just thinking of ways it would have authed you on the other servers for those subscriptions |
| 20:48.36 | darkunderlord | I wonder how I didn't have to auth to get the status on 11, hmm |
| 20:48.50 | tuxd00d | I have an âAutodestruct on dialogâ that wonât clear. Iâve tried âchannel redirectâ and âchannel request hangupâ but neither have cleared it. AMI is not enabled on this server. Do you have any other suggestions? InterWeb searches donât offer any solution other than a core restart. |
| 20:49.14 | darkunderlord | how can I check what servers are authed? |
| 20:49.32 | file | tuxd00d: pretty much have to do a core restart, something is making it so the channel won't go away no matter what and chan_sip may be the culprit itself |
| 20:50.04 | tuxd00d | file: Thanks buddy :) |
| 20:50.31 | file | darkunderlord: you can only see what something was matched against and authed as when something occurs, like a subscription or call |
| 20:51.11 | darkunderlord | file: so watch my logs on the current servers with 11 that seem to work? |
| 20:51.26 | file | you'd need to have sip set debug on |
| 20:51.39 | file | then it'll tell you what user/peer it matched and authed as |
| 20:52.08 | darkunderlord | will it be specifically a sub packet or something? |
| 20:52.24 | file | if the phone is subscribing to that server... yes. |
| 20:52.41 | file | heck, you could turn it on, unplug/plug back in a phone, and see |
| 20:55.50 | darkunderlord | oh I know that my Indy phones aren't logging into the Lafayette server. |
| 21:01.14 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:09.52 | darkunderlord | file: check here https://pastebin.com/zJmR4LCT |
| 21:10.09 | darkunderlord | this is the notify to 212, about 612 and they are on different servers. |
| 21:10.38 | file | what is 10.1.40.158? |
| 21:11.03 | darkunderlord | 212 |
| 21:11.19 | file | then that phone has directly contacted that server |
| 21:11.20 | darkunderlord | so it sends it directly to the phone that asked for a hint |
| 21:11.32 | file | that doesn't tell you how it authenticated, though |
| 21:11.34 | darkunderlord | yeah, but when I do that from PJSIP, it denies me |
| 21:11.37 | darkunderlord | ok |
| 21:11.41 | file | just that it did. |
| 21:11.59 | darkunderlord | so reboot 212, and watch sip debug for that? |
| 21:12.05 | file | sure |
| 21:13.16 | darkunderlord | is there an easy way to send the debug to a file? |
| 21:13.35 | file | I don't know off the top of my head. |
| 21:14.14 | lorsungcu | darkunderlord: open asterisk, enable debug, reboot phone, wait for notify, exit asterisk |
| 21:14.28 | file | you want a subscribe |
| 21:14.29 | lorsungcu | darkunderlord: copy everything from opening asterisk onwatrd |
| 21:15.31 | darkunderlord | like this? https://pastebin.com/vNrptEaC |
| 21:15.42 | file | that's a NOTIFY |
| 21:15.44 | file | not a subscribe |
| 21:15.47 | darkunderlord | ok |
| 21:16.52 | darkunderlord | https://pastebin.com/7a4W3ph9 |
| 21:18.29 | file | you have anonymous access enabled, and stuff is accessible from the default |
| 21:18.37 | file | er from the default context |
| 21:18.46 | darkunderlord | I'm ok with taht for now. Everything is behind firewalls no sip trunks outside |
| 21:19.05 | darkunderlord | I'd like to secure it but not as much as I'd like to upgrade :) |
| 21:19.20 | file | then in PJSIP you would enable the anonymous endpoint identifier in the Resources section of "make menuselect" and create an endpoint named "anonymous" pointed to wherever |
| 21:20.01 | file | you can also do an identify section based on a subnet mask so all traffic from a specific location will be identified as a specific endpoint |
| 21:20.02 | darkunderlord | what is wherever? Per phone or would they all share that one |
| 21:20.13 | darkunderlord | ah that would be good |
| 21:20.53 | darkunderlord | sucks I didn't do this right from the beginning /shrug |
| 21:21.26 | file | https://www.irccloud.com/pastebin/MXr9CmFI/ |
| 21:21.43 | file | so, you weren't sharing state at all |
| 21:21.55 | file | the phones were just directly contacting other servers and because of anonymous it was working |
| 21:22.13 | file | wanders off to cobble together sustenance |
| 21:23.08 | darkunderlord | file: thanks |
| 21:24.59 | Aljone | Hey all im trying to connect my asterisk to notify my webserver when there is incomming call, i wonder how do i do it ? |
| 21:31.04 | [TK]D-Fender | Aljone, This was already answered for you 10 hours ago |
| 21:31.09 | [TK]D-Fender | And the answer isn't changing |
| 21:33.13 | [TK]D-Fender | 7:18 am |
| 21:33.14 | [TK]D-Fender | Aljone |
| 21:33.14 | [TK]D-Fender | Hey all im trying to connect my asterisk to notify my webserver when there is incomming call, i wonder how do i do it ? |
| 21:33.14 | [TK]D-Fender | 7:41 am |
| 21:33.14 | [TK]D-Fender | dadrc |
| 21:33.15 | [TK]D-Fender | Dialplan hook via AGI, AMI listener or ARI listener |
| 21:33.17 | [TK]D-Fender | 7:49 am |
| 21:33.19 | [TK]D-Fender | Samot |
| 21:33.21 | [TK]D-Fender | Or straight dialplan. If it's just a one way notice then you can do a CURL() |
| 21:33.23 | [TK]D-Fender | 7:50 am |
| 21:33.25 | [TK]D-Fender | Or some other function, depending on the needs |
| 21:33.27 | [TK]D-Fender | 7:54 am |
| 21:33.29 | [TK]D-Fender | dadrc |
| 21:33.31 | [TK]D-Fender | That, too. |
| 21:33.33 | [TK]D-Fender | 7:55 am |
| 21:33.35 | [TK]D-Fender | so basically however you want :> |
| 21:33.37 | [TK]D-Fender | oops |
| 21:35.16 | lorsungcu | [TK]D-Fender: i want a BUTTON. |
| 21:35.29 | lorsungcu | conniptions |
| 21:35.32 | Aljone | [TK]D-Fender: i didnt seee |
| 21:35.55 | Aljone | curl is perfect |
| 21:36.19 | [TK]D-Fender | You were answered immediately by multiple people |
| 21:36.47 | Aljone | i missed it |
| 21:37.13 | Aljone | anyhow so curl, how do i actually tell the asterisk to run curl? |
| 21:40.29 | lorsungcu | Aljone: https://wiki.asterisk.org/wiki/display/AST/Home |
| 21:42.54 | Aljone | for example if someone call i want to send a request using curl wit hthe number |
| 21:43.01 | Aljone | https://wiki.asterisk.org/wiki/display/AST/cURL < i saw that |
| 21:43.02 | Aljone | BUT |
| 21:43.14 | Aljone | i didnt really see where it sending the number |
| 21:43.44 | [TK]D-Fender | wrong page |
| 21:44.30 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_CURL |
| 21:44.40 | [TK]D-Fender | that other page was about DB integration |
| 21:44.52 | [TK]D-Fender | not "just place a request to a URL" |
| 21:45.02 | [TK]D-Fender | [TK]D-Fender> Or straight dialplan. If it's just a one way notice then you can do a CURL() |
| 21:45.44 | Aljone | hmm le me see |
| 21:46.13 | Aljone | but i dont want to send to the asterisk |
| 21:46.21 | Aljone | i want the asterisk send to the webserver |
| 21:46.42 | [TK]D-Fender | <[TK]D-Fender> [TK]D-Fender> Or straight dialplan. If it's just a one way notice then you can do a CURL() |
| 21:47.01 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_CURL |
| 21:47.04 | [TK]D-Fender | read the instructions |
| 21:47.17 | Aljone | the first sentence "Retrieve content from a remote web or ftp server" |
| 21:47.24 | Aljone | retrieve < |
| 21:47.51 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |
| 21:47.54 | Aljone | and there are no instructions just the syntax |
| 21:47.55 | [TK]D-Fender | post-data - If specified, an HTTP POST will be performed with the content of post-data, instead of an HTTP GET (default). |
| 21:47.58 | [TK]D-Fender | POST <----------------- |
| 21:48.20 | Aljone | ok how do i put the number there? |
| 21:48.33 | [TK]D-Fender | in the encoded URL |
| 21:48.47 | Aljone | im sooo confused right now. |
| 21:48.53 | lorsungcu | Aljone: what is preventing you from trying this |
| 21:49.09 | [TK]D-Fender | Know what a URL looks like? |
| 21:49.09 | Aljone | lorsungcu : my understanding of how its spose to work |
| 21:49.32 | lorsungcu | Aljone: you call the function, include a URL and some post-data. |
| 21:49.44 | lorsungcu | Aljone: check your http logs, seee what was sent |
| 21:49.46 | Aljone | i want that everytime someone call it will send the number that called to http://domain.com/AsteriskNumber |
| 21:49.48 | lorsungcu | if it matches what you need, you're done |
| 21:50.00 | [TK]D-Fender | "http://webserver.tld/myscript.php?shit=123456" |
| 21:50.10 | Aljone | but where the ?shit=123456 comes from |
| 21:50.26 | [TK]D-Fender | And it will call out to that page and pass those vars and that script on your webserver will RUN with that and you can do whatever you want |
| 21:50.40 | Aljone | the 123456 is dymanic , where do i set it to take the number that called |
| 21:50.43 | [TK]D-Fender | YOU put that on the URL you give CURL() |
| 21:50.56 | Aljone | but how do i generate it AUTOMATICLY |
| 21:50.58 | [TK]D-Fender | So make that part of your call to CURL |
| 21:51.03 | [TK]D-Fender | it's a VARIABLE |
| 21:51.10 | [TK]D-Fender | use it in your dialplan |
| 21:51.27 | Aljone | ok im not asterisk person, what is dialplan |
| 21:51.36 | [TK]D-Fender | NOW you're screwed |
| 21:51.37 | Aljone | how do i set it up? |
| 21:51.39 | lorsungcu | lol |
| 21:51.40 | [TK]D-Fender | You're expected to know * |
| 21:51.50 | Aljone | i just need to somehow do that configuration |
| 21:51.59 | lorsungcu | Aljone: are you using freepbx |
| 21:52.02 | [TK]D-Fender | somehow = make your dialplan accordingly |
| 21:52.15 | [TK]D-Fender | along with everything else required to get it to accept those calls in the first place |
| 21:52.35 | [TK]D-Fender | Aljone> how do i set it up? <- You don't know asterisk |
| 21:52.40 | [TK]D-Fender | You need to actually learn |
| 21:52.41 | Aljone | we had a asterisk guy , all system is running |
| 21:52.44 | [TK]D-Fender | ~book |
| 21:52.44 | 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 |
| 21:52.50 | Aljone | hes gone, and we just ned to solve it somehow |
| 21:52.53 | [TK]D-Fender | Not YOU have to become that guy |
| 21:52.58 | [TK]D-Fender | Hor hire a consultant |
| 21:53.02 | [TK]D-Fender | or* |
| 21:53.06 | [TK]D-Fender | now* |
| 21:53.10 | [TK]D-Fender | can't type tonight |
| 21:53.23 | Aljone | :) |
| 21:53.59 | Aljone | ok i will check all u wrote |
| 21:54.44 | [TK]D-Fender | This time for SURE! |
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| 22:08.48 | *** mode/#asterisk [+o newtonr] by ChanServ |
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