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| 04:47.33 | sofltech | Hi all - I'm running an originate action through AMI which calls a number and runs a dialplan. If the call doesn't pick up, the dialplan still executes. Is there a way to prevent the dial plan from not running if the call isn't picked up? |
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| 05:14.47 | [TK]D-Fender | It is picking up. |
| 05:26.37 | sofltech | Hi TK thanks, but I'm not sure what you mean. I'm issuing an Originate request using SIP/# and giving it the relevant context,extension and piroirity info. The # rings and if the call is picked, the dialplan executes accordingly. If the call is not picked up or manually rejected by the recipient (pressing "reject" button on phone), the dialplan still runs. This is the issue? I don't want |
| 05:26.37 | sofltech | the dialplan to run if the call is not picked up or rejected. My understanding is that I don't have dialstatus available to me as I would if I ran Dial from the dialplan itself. |
| 05:27.27 | sofltech | I'm doing something similar to this: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message |
| 05:27.45 | sofltech | which I think would have the same issue if the call wasn't picked up. |
| 05:29.54 | [TK]D-Fender | Showing me something you feelis similar is pointless |
| 05:29.59 | [TK]D-Fender | You are doing something different |
| 05:30.15 | [TK]D-Fender | And if you want to know why it's not working out for you you'll have to actually LOOK at what it's doing |
| 05:30.38 | [TK]D-Fender | And if you can' tell from that... then you should have an idea what comes next.... |
| 05:32.41 | sofltech | Thanks, I'm posting it to pastebin momentarily. |
| 05:38.03 | sofltech | TK: this is the pastebin showing my code: http://pastebin.com/uTi9iTGX |
| 05:38.17 | sofltech | pretty straightforward I think... |
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| 05:40.52 | [TK]D-Fender | You're not looking at the CALL |
| 05:41.07 | [TK]D-Fender | "sip set debug on" <----- |
| 05:41.26 | sofltech | ok, trying that now, brb |
| 05:44.12 | sofltech | ok, got a lot of data, am i looking for anything specific? |
| 05:44.37 | [TK]D-Fender | Let's look at it.... |
| 05:47.58 | sofltech | k, posting in a moment... |
| 05:49.55 | sofltech | TK: http://pastebin.com/dYHPw8vp |
| 05:50.48 | [TK]D-Fender | #170 <---- ANSWERED |
| 05:53.36 | sofltech | TK: are you referring to this line: SIP/2.0 200 OK? |
| 05:54.57 | [TK]D-Fender | yes |
| 05:55.20 | [TK]D-Fender | I figured my pointing out the line number specifically was clear enough |
| 05:55.25 | [TK]D-Fender | THAT is the call getting answered |
| 05:56.43 | sofltech | just wanted to make sure I was looking at the same thing you were in terms of line number, b/c I'm new to this and am still getting used to the lingo what different data patterns mean, thanks. |
| 05:58.57 | sofltech | so am i doing something wrong? obviously if the service is communicating back to asterisk that the call was answered, the dial plan executes, i get that. |
| 05:59.45 | sofltech | but the call wasn't "answered", it was rejected - which I guess can be considered that it was answered, but it also happens if it just times out (meaning, no button is pressed on the phone). |
| 06:00.01 | [TK]D-Fender | It was answered |
| 06:00.10 | [TK]D-Fender | And I'm not seeing you thinking about WHAT answered it. |
| 06:00.24 | [TK]D-Fender | I don't care that the PERSON didn't answer.... |
| 06:00.29 | [TK]D-Fender | SOMETHING did. |
| 06:00.34 | [TK]D-Fender | Now what do you think that would be? |
| 06:02.25 | sofltech | I agree that something answered - the WHAT is obviously the question. is it clear from the sip log file what answered it - is that where I should be looking? |
| 06:03.39 | [TK]D-Fender | No... |
| 06:04.07 | [TK]D-Fender | What do you think HAPPENS when they "reject" the call? |
| 06:05.52 | sofltech | I was thinking about that as well, that clicking the "reject" button accepts the call but than hangs it up, or something like that. |
| 06:07.08 | sofltech | but the thing is, I'm seeing the same behavior even if the call isn't rejected, meaning, if it times out. |
| 06:07.34 | sofltech | but obviously the same thing is happening, meaning its being answered. |
| 06:07.44 | [TK]D-Fender | And you have no idea what could be answering the call like that? |
| 06:07.57 | sofltech | doh |
| 06:07.58 | sofltech | thanks |
| 06:08.06 | [TK]D-Fender | VOICEMAIL <---------------- |
| 06:08.08 | sofltech | it just occurred to me |
| 06:08.09 | sofltech | yep |
| 06:09.04 | sofltech | I can't believe I just spent "so many hours" on this - but that's how you learn. |
| 06:09.14 | sofltech | got to go through the process. |
| 06:11.21 | sofltech | ok, so, this leads me to the next question - is there a way to detect something is voicemail vs. not - and does it matter that the dial plan executes in such cases, perhaps there's no way to avoid it. |
| 06:11.40 | sofltech | just thinking aloud... |
| 06:12.02 | [TK]D-Fender | "core show application amd" |
| 06:12.28 | sofltech | in my particular case, its not an issue - I'm ok with it executing - this is more for learning sake. ok, looking at that now. |
| 06:26.04 | sofltech | So I'm messing around with AMD - cool, so far it only recognized me as a machine :-) even if I pick and speak into it. For my prticular use case, its not going to be an issue - dialplan is short and to the point, will execute and terminate pretty quickly if no confirm from recipient. thanks for your help TK. |
| 06:28.18 | [TK]D-Fender | You're welcome |
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| 06:45.18 | [TK]D-Fender | heads to bed... |
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| 08:34.36 | RaptorJesus | ne1 know for why? Unable to locate aor 1. MWI subscription failed. |
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| 15:36.24 | davlefou | Hi, it seems there are opus in asterisk 13.2 source but not in the list of code in menuselect. |
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| 16:05.52 | sofltech | hi all. using originate via ami / php to initiate a sip call and run dialplan. I'm setting the callerid to a number, but "unknown" is what shows up as the caller id on the recipient phone. I know that the callerid itself is getting set as I can confirm it in the dial plan. not sure what else I should be looking at? |
| 16:07.13 | [TK]D-Fender | Show us..... |
| 16:07.33 | sofltech | <PROTECTED> |
| 16:07.33 | sofltech | <PROTECTED> |
| 16:07.33 | sofltech | <PROTECTED> |
| 16:07.56 | sofltech | NoOp("SIP/rapidvox-0000016a", "CUSTOM: calleridall is "Abc" <1234>") in new stack |
| 16:09.47 | sofltech | the last line is the output from the dialplan showing that the callerid data is received, but I have no use for it in the dial plan - as the call has already been initiated? |
| 16:12.17 | [TK]D-Fender | FULL output |
| 16:12.54 | sofltech | TK: could you be more specific, what does Full OUTPUT mean? sip debug? something else? |
| 16:13.46 | [TK]D-Fender | YES SIP debug, and more complete code |
| 16:16.03 | sofltech | np, thanks, pastebin coming in moment... |
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| 16:21.19 | sofltech | http://pastebin.com/Y9Xt0mbn |
| 16:21.30 | sofltech | TK: see pastebin... let me know your thoughts? |
| 16:22.36 | [TK]D-Fender | You are sending it out. |
| 16:23.04 | [TK]D-Fender | However many providers block you from sending numbers you don't get through them. |
| 16:23.21 | [TK]D-Fender | or numbers that are invalid (like 1234) |
| 16:24.36 | sofltech | ok, thanks, let me try a valid number. |
| 16:26.20 | sofltech | TK: outstanding, thanks, valid number worked! |
| 16:27.08 | sofltech | frustrated that I looked for an answer online regarding this issue... nothing came up... glad to have your help. |
| 16:28.39 | [TK]D-Fender | Never put stupid lloking values in tests. Put things that look right but are just different enough to show if it worked |
| 16:29.13 | [TK]D-Fender | Just because you can set something doesn't mean you can set ANYTHING |
| 16:32.28 | sofltech | yep |
| 16:32.39 | sofltech | making progress... |
| 16:32.41 | sofltech | :-) |
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| 19:03.04 | sofltech | question... my originate script works, dialplan is executed. originate application exits as soon as call is completed while dialplan continues to execute. how can i tell when dialplan is complete so that i can determine next course of action? |
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| 19:12.35 | Penguin | Originate() should be a blocking app. |
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| 19:17.00 | sofltech | Penguin: thanks, that's what I thought as well, this is my test code: http://pastebin.com/Tphr9B0y could you have a look and see if you see anything suspicious? |
| 19:18.38 | MadHatter42 | anyone here has implemented oreka ? |
| 19:20.43 | sofltech | Penguin: nm, I think I figured it out, sip configuration issues, thanks. |
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| 19:28.14 | WIMPy | Originate blocks while the cass is set up. |
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| 20:17.57 | [[thufir]] | what does it mean that I get "Congratulations. You have successfully installed and executed the Asterisk open source PBX. You have also installed a set of sample..."? I tried to google it, not much came back. |
| 20:21.59 | Penguin | [[thufir]]: It means you are using a SAMPLE (read: not to be used for production) dial plan, and your system works properly. |
| 20:22.26 | [[thufir]] | Penguin: ohh, interesting. |
| 20:25.09 | Penguin | Are you using FreePBX? |
| 20:25.41 | [[thufir]] | Penguin: granted, right now I have no outside connectivity, so peers are only on the network. however, other than that, I **think** the dialplan is ok. can you give me some pointers to google on that? just dialplan, sample versus production? I'm using vicidial. I'm looking at extensions.conf now, but it's a huge file. |
| 20:26.28 | Penguin | extensions.conf shouldn't have much stuff in it until you, the admin, put stuff in it. You're seeing the sample file. |
| 20:26.36 | Penguin | ~book |
| 20:26.36 | 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 |
| 20:26.38 | Penguin | ~primer |
| 20:26.38 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 20:27.30 | [[thufir]] | Penguin: yes, I have that book, thank you. it seems very general, but interesting. |
| 20:29.53 | [[thufir]] | I'm taking a look at that burner site, thanks. My dialplan: http://pastebin.com/W2UYifpf |
| 20:31.42 | [[thufir]] | I'm thinking i don't have anything configured for my dialplan to answer. anyhow, thanks again. |
| 20:52.28 | Penguin | [[thufir]]: All those lines in extensions.conf... those are your dialplan. A default asterisk dialplan doesn't have those things, because a default asterisk dialplan is an empty file. |
| 20:55.10 | [[thufir]] | Penguin: in general, I'm with you, I get exactly what you're saying. It's kind of odd, because if I just dial 101 from 102, it works fine. however, when vicidial dials with its web interface, then I get that recording. Maybe it's not dialing 102! If I look in /var/logs/asterisk/messages I should see some information on that, right? I want to see what's happening as it happens. Maybe I'l use the * console instead... |
| 20:56.36 | [[thufir]] | specifically, that extensions.conf is a bit overwhelming for me. I don't mind going through it, but I think the problem lies either with vicidial-extensions.conf or something else. maybe not actually * now that I think about it. |
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| 21:06.02 | Penguin | [[thufir]]: Yeah, you'll need to go to the asterisk CLI and increase the core verbosity level. |
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| 21:17.43 | sofltech | ok, so I posed this question earlier today and though I had it resolved, but now I'm not so sure. Using originate to make a call and run a dialplan. right now, originate is initiating the call and running the dialplan but returning to php while the dialplan is running. |
| 21:18.03 | sofltech | I'd like it to only return once the dialplan is complete. |
| 21:18.09 | sofltech | which is what I thought it was supposed to do. |
| 21:19.50 | sofltech | I'm clearly missing something or not thinking about it correctly. |
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| 22:17.28 | ChannelZ | You're launching this from where? |
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| 22:26.37 | [TK]D-Fender | sofltech, NO. Originate is asynchronous. It is a completely separate call. Also, nothing about AMI is "blocking" at all in the first place. |
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| 22:33.02 | Penguin | If you're doing it from the standard dialplan, processing should not continue until Originate() exits. |
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| 22:48.18 | sofltech | TK: thanks. that's what I'm learning. so what I have to do is make the originate call and listen to events and process accordingly, is that accurate? |
| 22:59.23 | [TK]D-Fender | sofltech, I alreay told you what to do days ago. have your channel send out an AMI UserEvent |
| 23:06.09 | sofltech | TK: thanks, I am doing that, but if they just hang up the call, I don't know what the status is because that event doesn't fire, so I'm left in limbo. I could check for the "hangup" event and assume that if I don't have the userevent, that they hung up, which would work. |
| 23:07.04 | sofltech | right now I have an event listener set up and its working, but I can't seem to be able to determine when the last event has come through - but it may not matter if I can catch the hangup event i guess. |
| 23:07.50 | [TK]D-Fender | sofltech, Use a Local channel as your Channel: to do the actual dialing and have THAT send on failure. |
| 23:08.14 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 23:11.35 | sofltech | TK: thanks, I'll try that route, I wasn't able to get that method to work correctly previously, but have more info now, so perhaps I can make it work. that still leaves me with having to process the events through a listener (which I have) but I don't understand if there's a way to know when the stream of events is done or its just a matter of finding what you're looking for processing it |
| 23:11.35 | sofltech | and logging off - which I think is the way to go. |
| 23:13.25 | [TK]D-Fender | If you expect the script that is running and launched that Originate to wait for the answer then continue to do stuff afterwards then you'll need to pass it that message |
| 23:14.59 | sofltech | TK: I do, b/c if I have to iterate through a few numbers, until someone acks the message. so if the ack userevent isn't passed, i need to continue processing and originate another call. |
| 23:15.27 | [TK]D-Fender | sofltech, You send one on FAILURE as well.... |
| 23:15.42 | [TK]D-Fender | it is a response of "done" or "fail". |
| 23:15.56 | [TK]D-Fender | You're listenening for an a response regardless |
| 23:16.54 | sofltech | TK: yes, it looks like I'll have to send a fail message as well. |
| 23:19.06 | sofltech | TK: could you elaborate a bit more on the benefit of using the local cahnnel to dial? meaning, what is the benefit of the extra step? |
| 23:19.24 | [TK]D-Fender | To be ABLE to send it on fail <- |
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| 23:23.40 | sofltech | TK: I'm sorry, I don't quite follow (I thought I did). where do i inject the fail event in this flow? what is the logic (if you don't mind giving me a few more tidbits). it seems like its an extra step b/c I could pass a fail message in the "h" event of the dialplan? |
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| 23:31.01 | [TK]D-Fender | No. "h" assumes the other side answers and the call dies. |
| 23:31.07 | [TK]D-Fender | You don't need this point |
| 23:31.24 | [TK]D-Fender | If you dial a Local channel to PLACE the outbound in the first place you can put it after your DIAL |
| 23:31.33 | [TK]D-Fender | Because it'll only get there if it fails |
| 23:31.48 | [TK]D-Fender | Then on the OTHER end of your dialplan you have it send on saying SUCCESS |
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| 23:50.58 | sofltech | TK: Thank you, for that expanded explanation, digesting it. |