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| 00:19.59 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 01:59.01 | Forbidd3n | Hey everyone. I am calling a SIP and passing extra custom variables in the url for example -- sip:5.5.5.5;param1=value1;param2=value2 -- I have an inbound route that handles the request to an AGI(script.php) file. I need to pass param1 and param2 to the PHP file script.php but figure out how to do this. Anyone here that can help me with this, please? I have tried -- AGI(script.php,${param1},${param2} -- but no luck |
| 02:03.41 | rwb | File Permissions for asterisk user question: How do I maintain file permissions to executable for the asterisk user? My shell scripts get reset to rw on reboot. |
| 02:16.41 | Forbidd3n | Anyone that can help shed light on my question on how I can achieve it? |
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| 03:35.31 | [TK]D-Fender | Forbidd3n, Those don't magically become * channel variables |
| 03:36.18 | [TK]D-Fender | And the only headers you have access to in the dialplan are actual standard headers, not the INVITE line itself, or most of the later parts of the initial invite |
| 03:36.55 | [TK]D-Fender | rwb, Noting in * changes permissions |
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| 03:49.19 | Forbidd3n | [TK]D-Fender: So is there not a way to get query vars from the sip link to the extension dialplan AGI file? |
| 03:49.46 | [TK]D-Fender | no |
| 03:50.08 | [TK]D-Fender | if you're lucky you might have a way using PJSIP and either ARI or a dedicated dialplan app |
| 03:50.22 | [TK]D-Fender | but nothing with chan_sip, and AGI |
| 03:56.44 | Forbidd3n | Can I pass additional headers through the sip request and get it on the extension or AGI? |
| 04:00.04 | Forbidd3n | [TK]D-Fender: ^ |
| 04:03.14 | Samot | Forbidd3n: They are passed like any command line script variable. |
| 04:03.36 | Samot | So if it's .php then it's $argv[1], $argv[2], etc etc. |
| 04:04.27 | Samot | AGI(script.php,${ARG1},${ARG2}) turns in script.php arg1 arg2 |
| 04:11.18 | Forbidd3n | Samot: I need to get a variable passed through the sip:5.5.5.5 url to the AGI and to the extension. |
| 04:11.57 | Forbidd3n | I know I can do $argv[...], but how do I get it from the sip url to the AGI reference |
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| 04:34.24 | Forbidd3n | Samot: you still around? ^ |
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| 11:15.19 | igeni | with ARI how would i be able to see wich peers are active |
| 11:20.04 | igeni | ow ARI is somethign dirant then Asterisk Manager API |
| 11:20.05 | igeni | i gues |
| 11:20.49 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+ARI |
| 11:21.37 | igeni | yeah ARI doesnt have stuff like sippeers |
| 11:33.56 | Samot | Yeah.. |
| 11:34.00 | Samot | I wouldn't image it would |
| 11:34.09 | Samot | Since ARI was introduced in 12 |
| 11:34.41 | Samot | And by the time 12 was done and 13 was released, Chan_SIP become deprecated by Digium and they stopped working on it. |
| 11:35.02 | Samot | Chan_SIP has been community developed/supported since like Aug 2014. |
| 11:36.15 | igeni | so everyone uses the manager now? |
| 11:36.50 | igeni | i gues they would, it has way more options |
| 11:38.18 | Samot | ARI is not a replacement for AMI |
| 11:38.23 | Samot | Not at the moment. |
| 11:57.54 | file | indeed, they serve two different purposes |
| 11:58.25 | file | ARI can be a replacement for AMI+AGI usage for some people, if they were using them combined to do what ARI does |
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| 13:02.23 | Forbidd3n | Samot: you around? |
| 13:02.40 | Samot | Yes. |
| 13:02.59 | Forbidd3n | In regards to my question earlier. |
| 13:03.11 | Forbidd3n | I need to get a variable passed through the sip:5.5.5.5 url to the AGI and to the extension. |
| 13:03.11 | Forbidd3n | I know I can do $argv[...], but how do I get it from the sip url to the AGI reference |
| 13:04.16 | Samot | First, what are you doing? |
| 13:04.34 | Samot | Where does this SIP URI come from? |
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| 13:13.17 | Forbidd3n | I am using Twilio SDK to call a SIP url to collect PCI data on our FreePBX box. My intentions is when it get to a specific point within a call, then hit the PBX server, collect the DTMF tones, store it in an encrypted file and send back a response to the Twilio SDK a token. Then when I am ready to process the payment I would request the PCI data through API call with the token. |
| 13:14.11 | Forbidd3n | So the flow would be call comes into Twilio number, I ask a series of questions, when I get to the credit card info question, I ask and then send request to SIP server, capture data, send back token. |
| 13:14.15 | Forbidd3n | Samot: ^ |
| 13:14.28 | Samot | And their SDK requires you to send this information to a SIP URI? |
| 13:15.50 | Forbidd3n | The SIP url is our internal PBX box. |
| 13:16.16 | Samot | OK. |
| 13:16.21 | Samot | So, when you're in dialplan.. |
| 13:16.28 | Forbidd3n | Twilio SDK isn't requiring anything. Twilio isn't PCI compliant so the way I am doing it will make it PCI compliant. |
| 13:16.33 | Samot | And you call AGI(script.php) |
| 13:16.42 | Forbidd3n | ok... |
| 13:16.47 | Samot | You're calling the script via the command line of the system |
| 13:16.50 | Samot | That's what is does. |
| 13:16.55 | Forbidd3n | correct |
| 13:16.58 | Samot | OK |
| 13:17.14 | Samot | So whatever you send at that point as arguments is up to you. |
| 13:17.30 | Samot | You could do AGI(script.php,Test) |
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| 13:17.53 | Samot | That would basically be /script.php test making test $argv[1] |
| 13:17.56 | Samot | In the script. |
| 13:18.20 | Samot | If you are trying to send data from the call, you need to set those variables to pass. |
| 13:18.26 | Samot | Like any other call. |
| 13:18.50 | Samot | If you pass ${ARG1}, the you need to set ${ARG1} via the dialplan with a Set() command. |
| 13:18.58 | Forbidd3n | Ok, is it possible to return text as a response to the SIP call? |
| 13:19.02 | sibiria | hmmm, will reloading the dial plan send HUP to processes spawned via AGI? |
| 13:19.17 | Samot | Response to the SIP call? |
| 13:19.20 | Samot | No. |
| 13:19.37 | Samot | Unless I'm misunderstanding what you mean by that. |
| 13:19.41 | [TK]D-Fender | Forbidd3n |
| 13:19.41 | [TK]D-Fender | Can I pass additional headers through the sip request and get it on the extension or AGI? <- no |
| 13:19.55 | Samot | Yes. |
| 13:19.59 | Samot | Yes he can. |
| 13:20.06 | Samot | As long as they are passed on the INVITE |
| 13:20.18 | Samot | You can get the SIP Headers being sent, including custom. |
| 13:20.46 | [TK]D-Fender | I was reForbidd3n |
| 13:20.46 | [TK]D-Fender | Can I pass additional headers through the sip request and get it on the extension or AGI?If they are headers. His previous request was for the INVITE string itsel |
| 13:20.47 | Forbidd3n | Samot: Ok, so how would I retrieve and send the custom SIP headers to the AGI script? |
| 13:20.58 | [TK]D-Fender | not a "Fred: Value" header |
| 13:21.18 | Samot | n,Set(__FROMUSER=${CUT(CUT(SIP_HEADER(From),@,1),:,2)}) |
| 13:21.20 | Samot | ^^ Yes. |
| 13:21.34 | Samot | SIP_HEADER(HEADERNAME) |
| 13:21.40 | Forbidd3n | [TK]D-Fender: this would work for me as well. I basically need to get data from my system to PBX box |
| 13:22.06 | [TK]D-Fender | and what mechanisms do you have for those value to get passed BACK after collecting DTMF? |
| 13:23.05 | Samot | That's what the AGI is for. |
| 13:23.17 | Samot | Call comes in from Twilio with custom SIP headers |
| 13:23.23 | Forbidd3n | [TK]D-Fender: I will pass the sessId through the header, store it on the server and when I need it do a API request to the server to retrieve it |
| 13:23.38 | Samot | Takes those headers, with the DTMF collected and passes it back to Twilio via API through the AGI call |
| 13:23.39 | Forbidd3n | my session id, in the header, would be my token to request |
| 13:23.52 | Samot | Am I getting that right? |
| 13:23.56 | Samot | In a nutshell? |
| 13:24.04 | Forbidd3n | Samot: correct |
| 13:24.08 | Samot | OK |
| 13:24.10 | [TK]D-Fender | Forbidd3n, what API request? |
| 13:24.13 | Samot | exten => *98,n,Set(__VMUSR=${SIP_HEADER(P-CVN-VMU)}) |
| 13:24.13 | Samot | exten => *98,n,Set(__VMCTXT=${SIP_HEADER(P-CVN-VMC)}) |
| 13:24.21 | Forbidd3n | [TK]D-Fender: I am going to create the API request |
| 13:24.25 | Samot | That's how you would grab your headers, for example. |
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| 13:24.31 | [TK]D-Fender | what KIND? How? |
| 13:24.51 | Samot | In that example, I pass P-CVN-VMU as a custom sip header |
| 13:24.58 | Forbidd3n | Samot: ok, I will give this a shot. Where can I find documentation on the items in the comma-delimited string? I have seen -- n,s,w, and numbers; but not sure what they all are or mean. |
| 13:25.07 | Samot | SIP_HEADER(HEADERNAME) will get the value of that header. |
| 13:25.10 | Forbidd3n | Samot: Yeah, I understand. Just new to the syntax |
| 13:25.42 | Samot | Well you're going to need to learn some dialplan syntax. |
| 13:25.43 | Samot | 9:21:18 AM <Samot> n,Set(__FROMUSER=${CUT(CUT(SIP_HEADER(From),@,1),:,2)}) |
| 13:25.50 | Samot | ^^ That gets the FROM header value |
| 13:26.04 | Samot | The rips it apart at the @ character |
| 13:26.15 | Samot | The trims off more data at the ; from the header |
| 13:26.20 | Samot | To give me the value I want. |
| 13:26.53 | Samot | That basically takes user@sip.domain.com:5060;var=val |
| 13:26.59 | Samot | And returns just user |
| 13:27.25 | Samot | So you are going to have to deal with the data you are sending in your SIP headers. |
| 13:27.32 | Samot | Set the variables you need based on that. |
| 13:27.51 | Forbidd3n | Samot: gotcha, is ;var=val a header var? |
| 13:28.21 | [TK]D-Fender | no |
| 13:28.35 | Samot | No. |
| 13:28.53 | Forbidd3n | I figured. It isn't the header. Let me figure out how to pass custom headers through Twilio SDK SIP method |
| 13:29.01 | Samot | I JUST TOLD YOU |
| 13:29.05 | [TK]D-Fender | A line like "From: "300@somedomain.com;blah" in the invite is a header |
| 13:29.11 | Samot | 9:24:13 AM <Samot> exten => *98,n,Set(__VMUSR=${SIP_HEADER(P-CVN-VMU)}) |
| 13:29.18 | Samot | 9:24:52 AM <Samot> In that example, I pass P-CVN-VMU as a custom sip header |
| 13:29.42 | Samot | I send P-CVN-VMU: 2110 |
| 13:29.59 | Samot | That piece of dialplan sets VMUSR to 2110 |
| 13:30.05 | Samot | Because that's the VALUE of the header. |
| 13:30.07 | Forbidd3n | Samot: I understand. This is the Twilio SDK SIP method that I am using - https://www.twilio.com/docs/api/twiml/sip |
| 13:30.42 | Samot | JFC. |
| 13:30.43 | Samot | $dial->sip('sip:jack@example.com?mycustomheader=foo&myotherheader=bar'); |
| 13:30.46 | Samot | Take that |
| 13:30.51 | Samot | And it would be |
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| 13:31.11 | Forbidd3n | Yeah, I just didn't know how to receive it on the other end with the proper syntax |
| 13:31.13 | Samot | $dial->sip('sip:jack@example.com?P-CVN-VMU=2110'); |
| 13:31.40 | Samot | exten => _X.,n,Set(__VMUSR=${SIP_HEADER(P-CVN-VMU)}) |
| 13:31.52 | Samot | I can only show this so many times in so many ways. |
| 13:32.03 | Samot | SIP_HEADER() will get the damn sip headers. |
| 13:32.05 | Forbidd3n | Samot: I understand now, thanks |
| 13:32.59 | Samot | Then it would be AGI(script.php,${VMUSR}) |
| 13:34.05 | Forbidd3n | Samot: another question, if I do a SIP call with that example would it collect the DTMF tones by the user store them in my AGI script before it continues through the rest of the local script asynchronous ? |
| 13:34.21 | Samot | You need to write dialplan for this |
| 13:34.21 | [TK]D-Fender | AGi is not storage. |
| 13:34.38 | Samot | I mean you can use AGI commands to collect the DTMF |
| 13:34.52 | Forbidd3n | [TK]D-Fender: I know. |
| 13:35.16 | Samot | You're focusing on the two bookends |
| 13:35.23 | Samot | And not putting books between them. |
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| 13:35.58 | Forbidd3n | would it be asynchronous, meaning I do the sip call, would that finish before it continues with the local script that made the SIP call? |
| 13:36.22 | Forbidd3n | Thank you very much for your help with understand the flow and syntax. It is much appreciated! |
| 13:36.47 | [TK]D-Fender | no. |
| 13:37.20 | [TK]D-Fender | you send a call to *. * gets the call with whatever headers you send. You would ANSWER the call to start colelcting DTMF. The call is now IN PROGRESS |
| 13:37.52 | [TK]D-Fender | now that call is ROLLING... when do you expect to END the call and how do you expect to RETURN the data? |
| 13:38.21 | Forbidd3n | [TK]D-Fender: that is what I am trying to understand. |
| 13:38.51 | Forbidd3n | I want to end the call when the user puts in the requested data and hits the pound key |
| 13:39.18 | [TK]D-Fender | and you expect * the terminate that SIP call it was sent? |
| 13:39.41 | [TK]D-Fender | Sending on that data? |
| 13:40.09 | Samot | So again, that's all dialplan. |
| 13:40.26 | [TK]D-Fender | should be |
| 13:40.29 | Forbidd3n | My objective is to do a SIP call, collect DTMF tones, and put the caller back to the original call and continue the original workflow |
| 13:40.36 | Samot | You have to collect the DTMF, and when they hit pound do what you need to do and hangup the call when you send your data. |
| 13:40.45 | Samot | Wait. |
| 13:40.49 | [TK]D-Fender | put back how? to * that is just a new call. |
| 13:40.55 | Samot | The "original call" |
| 13:41.01 | Forbidd3n | Samot: yes. |
| 13:41.09 | Samot | You don't have an original call |
| 13:41.13 | Samot | You dialed to Asterisk |
| 13:41.15 | Samot | That's the call |
| 13:41.16 | [TK]D-Fender | There is no "give back", because there is no concept of original state to * |
| 13:41.20 | Samot | The call is now on Asterisk. |
| 13:41.32 | Samot | If you want to send it back to Twilio, you have to Dial() out via Asterisk to Twilio. |
| 13:41.35 | Forbidd3n | Ok, so now I have to call back to the original Twilio call? |
| 13:41.38 | Forbidd3n | from Asterisk |
| 13:41.45 | Samot | There is NO ORIGINAL CALL |
| 13:41.56 | Forbidd3n | Samot: ok the initial call |
| 13:42.02 | Samot | 9:31:13 AM <Samot> $dial->sip('sip:jack@example.com?P-CVN-VMU=2110'); |
| 13:42.13 | Samot | ^^^ Once you do that you are sending the call OFF Twilio. |
| 13:42.19 | Samot | They are sending it to Asterisk. |
| 13:42.20 | Forbidd3n | correct |
| 13:42.22 | [TK]D-Fender | You could Transfer() the call to toss it off your server which IIRC looks like a 302 redirect |
| 13:42.33 | Samot | As far as Twilio is concerned that call is done. |
| 13:42.38 | Forbidd3n | Samot: correct |
| 13:42.48 | Forbidd3n | so I need to call twilio back from asterisk |
| 13:42.51 | Samot | So you're not sending back to the "original" or "initial" call. |
| 13:43.00 | Samot | You are sending a new call back to Twilio. |
| 13:43.01 | [TK]D-Fender | Not sure if that would be identifiable coming back |
| 13:43.13 | [TK]D-Fender | Samot, Transfer() <- might do it |
| 13:43.17 | Forbidd3n | as long as I have that session id I can reload it, figure out the step where they were and redirect them to the correct spot in the workflow |
| 13:43.36 | Forbidd3n | I think this would work |
| 13:43.52 | [TK]D-Fender | Forbidd3n, Go make a simple dialplan to Answer, Playback some audio, and Transfer() back to your server and see what it looks like coming back |
| 13:44.12 | [TK]D-Fender | Forbidd3n, Go test that before wasting any extra time. |
| 13:44.15 | Forbidd3n | [TK]D-Fender: Great recommendation. I think I will do that to understand the complete workflow |
| 13:44.30 | Forbidd3n | Thank you both for your help with this. |
| 13:44.57 | [TK]D-Fender | Forbidd3n, SPECIAL NOTE. Yuo can't send the answer BACK in a header by any other means that using DIAL() to call twilio which forces * to be in the loop from that point forward. |
| 13:45.14 | Samot | Yup |
| 13:45.23 | [TK]D-Fender | Which why I'm tryying to get a solid ansqwer on HOW you were expecting to get the results back |
| 13:45.29 | igeni | withn the manager api of asterisk how would i track if a peer or extension starts a call |
| 13:45.34 | Samot | That call is still Twilio -> asterisk -> Twilio from that point forward. |
| 13:45.46 | [TK]D-Fender | And your sessid is DEAD |
| 13:45.52 | [TK]D-Fender | that's a NEW CALL * is sending to twilio |
| 13:46.01 | [TK]D-Fender | which probably kills that idea |
| 13:46.05 | Samot | Well.. |
| 13:46.13 | Samot | Twilio is a 100% programmatic. |
| 13:46.24 | Forbidd3n | Ok so there is no way to get the session id from asterisk that I original got from Twilio headers back to Twilio? |
| 13:46.24 | [TK]D-Fender | but * is what's sending the call |
| 13:46.26 | Samot | So it is possible they could bridge that call back to the original and drop Asterisk. |
| 13:46.40 | Samot | JFC |
| 13:46.41 | Samot | YES |
| 13:46.46 | Samot | Send it as a SIP HEADER |
| 13:46.50 | Samot | Collect it. |
| 13:47.06 | Forbidd3n | correct the same way I sent it to asterisk |
| 13:47.13 | Samot | Use SIPAddHeader() to add it as a SIP header to send it back |
| 13:47.22 | Samot | You need to go and learn some dialplan my friend. |
| 13:47.23 | [TK]D-Fender | When you Dial() You're going to get another one generated. What are you going to do, send a duplicate along with it? |
| 13:47.24 | Samot | That's it. |
| 13:47.31 | [TK]D-Fender | Sounds like suicide |
| 13:47.34 | Samot | We can't sit here and give you pieces. |
| 13:47.46 | Forbidd3n | [TK]D-Fender: why does it sound like suicide? |
| 13:47.49 | Samot | HE can send it back as a custom SIP header. |
| 13:47.56 | Samot | There are ways to do this |
| 13:48.08 | Samot | It's all going to depend on what Twilio's side can do. |
| 13:48.15 | *** join/#asterisk nighty- (~nighty@s229123.ppp.asahi-net.or.jp) |
| 13:48.17 | Forbidd3n | Would you mind explaining the bridge concept on a high level? |
| 13:48.28 | Samot | Asterisk uses channels.. |
| 13:48.30 | Forbidd3n | I think this is what I want, but couldn't think of that term |
| 13:48.41 | Samot | When Channel A calls Channel B and Channel B answers, they are bridged.. |
| 13:48.43 | Samot | So audio can flow |
| 13:48.47 | [TK]D-Fender | if twilio has some way of matching this new call's custom header to the call still in progress |
| 13:48.49 | Samot | and any other media |
| 13:49.48 | Forbidd3n | Gotcha. I will do research on bridging. Thanks again for all your help with understanding this. |
| 13:50.02 | Forbidd3n | I will let you know how it goes. |
| 13:50.11 | Samot | Twilio's SDK and Progammed SIP is pretty extensive. |
| 13:50.34 | Samot | This is actually the type of use case it is designed for. |
| 13:50.41 | Forbidd3n | Did you see documentation somewhere. All I seen was that URL I shared with you |
| 13:50.42 | Samot | His issue is the Asterisk part in the middle. |
| 13:50.58 | Forbidd3n | Samot: correct |
| 13:50.58 | Samot | You just have to scour through their stuff. |
| 13:51.08 | Forbidd3n | Thanks. I will. |
| 13:51.09 | Samot | So figure out Asterisk stuff |
| 13:51.19 | Samot | You need to learn how to program Asterisk for this as well. |
| 13:51.36 | Samot | And the amount of options that could/would help you is impossible to go through. |
| 13:52.04 | Forbidd3n | I understand. Thanks again. Off to a meeting |
| 13:53.40 | [TK]D-Fender | Samot, only thing I can think of for sending that call back with the same info it came in with (sessid, etc)so it would been as the original call would be Transfer() but there is no mechanism for adding headers in that response. |
| 13:53.48 | Samot | I'm not a 100% but I'm sure if I spent time in the Twilio dev docs I'm pretty sure I would see he's doing it wrong. |
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| 13:53.59 | Samot | So.. |
| 13:54.03 | Samot | INVITE comes in. |
| 13:54.14 | Samot | SIP_HEADER() to grab the headers you want, including the session id |
| 13:54.25 | Samot | Which Twilio will probably send as a custom P header. |
| 13:54.30 | Samot | You store that |
| 13:54.43 | Samot | Add it back to the DIAL via SIPAddHeader() as a custom header. |
| 13:54.44 | [TK]D-Fender | figured it was the SIP session ID |
| 13:54.48 | [TK]D-Fender | not a custom one. |
| 13:54.52 | Samot | Right |
| 13:55.01 | Samot | But if he wants the original SEssionID |
| 13:55.09 | Samot | HE can still get it and pass it back as a custom header |
| 13:55.29 | Samot | P-OrigSessID: <sessionid> |
| 13:55.32 | [TK]D-Fender | Sounds like he wanted the original call sent back so it still has that unique callid header. |
| 13:55.32 | Samot | Not that hard. |
| 13:55.39 | [TK]D-Fender | Can twilio match that? |
| 13:55.40 | Samot | Yes. |
| 13:55.43 | Samot | But it's Twilio. |
| 13:55.49 | [TK]D-Fender | and the only way means going MITM via Dial() |
| 13:55.53 | Samot | They might have a method to drop asterisk. |
| 13:56.04 | [TK]D-Fender | and having to maintain 2 legs through * now |
| 13:56.13 | Samot | Again... |
| 13:56.24 | Samot | Twilio might be able to bridge back to the original call and drop the Asterisk leg. |
| 13:56.41 | [TK]D-Fender | and that call from * would have the * generated callid ... and a 2nd header of the sme name if its the standard one |
| 13:56.47 | Samot | OK. |
| 13:56.52 | [TK]D-Fender | since it is a new call |
| 13:56.56 | Samot | Right |
| 13:57.07 | Samot | And the old calls data can be passed back in custom headers |
| 13:57.12 | [TK]D-Fender | * isn't a proxy... so How would Twilio know they were related and how would * step out of the way? |
| 13:57.13 | Samot | Which Twilio can parse. |
| 13:57.15 | Samot | And do shit with |
| 13:57.17 | [TK]D-Fender | I don't see that part happening |
| 13:57.34 | Samot | How much have you dived into Twilio and what they can do? |
| 13:57.59 | Samot | Like I said.. |
| 13:58.07 | [TK]D-Fender | I haven't. its just the separate legs side on * I don't know how it can hand off |
| 13:58.15 | Samot | If I dove more into their docs I would probably find a way to do this that didn't even require Asterisk. |
| 13:58.24 | [TK]D-Fender | Transfer() would throw the original back ... but no extra headers |
| 13:58.31 | Samot | I think some of this is lack of knowledge. |
| 13:58.41 | Samot | On both the Twilio side and the Asterisk side. |
| 13:58.45 | Samot | Asterisk might not even be needed. |
| 13:59.08 | [TK]D-Fender | There's always other software... |
| 13:59.24 | [TK]D-Fender | and things they may offer. I don't know Twilio.. or others for that matter |
| 13:59.44 | [TK]D-Fender | just looking at the * side for his being here. |
| 14:02.01 | Samot | Well the Twilio and even places like Bandwidth have SIP API platforms... |
| 14:02.25 | Samot | Allowing you to use their voice network to handle calls and control them programmatically. |
| 14:02.41 | Samot | In some cases you don't even need a voice server on your end just a server that can do commands. |
| 14:02.47 | [TK]D-Fender | And here we are with * not being a SIP proxy and making "having fun" a PITA |
| 14:03.38 | [TK]D-Fender | Where it can just accept the call collect more data, add headers to the original and bounce it back "intact" as it was to resume its journey |
| 14:04.02 | Samot | The SIP proxy would have to do media. |
| 14:04.12 | Samot | Since the DTMF would be in the media stream. |
| 14:04.24 | [TK]D-Fender | it's that hand-off and "still keep the rest of the original header" type stuff that trips this up |
| 14:04.28 | Samot | Things like Kamailio and OpenSIPS can't grab DTM |
| 14:04.30 | Samot | Things like Kamailio and OpenSIPS can't grab DTMF |
| 14:04.53 | [TK]D-Fender | My knowledge does drop right off after this point |
| 14:05.48 | Samot | Well this is the part where Asterisk is used as a voice toolkit and not a PBX |
| 14:06.37 | Samot | But again, it is totally possible this is something that doesn't even require Asterisk but that would require looking at Twilio's docs and figuring out how to do it myself. Which, I don't need to. |
| 14:08.31 | [TK]D-Fender | #notourjob |
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| 15:56.41 | radicaldev | [TK]D-Fender: I picked up an SPA112 ATA which functionally has everything I need, but I've got a weird problem where everything that's heard through the phone speaker is sent in the outgoing audio stream like it came through the microphone, even when the mic is muted. Any idea about that? |
| 15:58.26 | [TK]D-Fender | check your gain levels, AEC, etc |
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| 16:05.00 | radicaldev | dang, sorry I missed that thread with forbidden. I think he's going about all of that hackily. |
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| 16:28.59 | jamesaxl | Hello |
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| 16:37.39 | jamesaxl | I am trying to beat freepbx :) |
| 16:37.57 | pclerie | Hello! I am looking for some help troubleshooting a one-way audio problem. |
| 16:38.20 | pclerie | First my set up: asterisk 13.14.1 on Debian |
| 16:38.43 | pclerie | external calls through GSM gateway GoIP4 |
| 16:39.05 | pclerie | GSM gateway has 4 SIMS in hunt group mode |
| 16:39.21 | pclerie | Outgoing call use SIMs in round-robin |
| 16:39.33 | pclerie | phones are Polycom 331 and 650 |
| 16:39.45 | pclerie | single codec configured : ulaw |
| 16:39.57 | pclerie | no trunks so not NAT traversal. |
| 16:40.20 | pclerie | No problems with audio on internal VoIP |
| 16:40.51 | pclerie | No problems with audio on incoming calls through GSM gateway |
| 16:41.24 | pclerie | But on outgoing call, there's no outgoing audio, but incoming audio is fine. |
| 16:41.41 | pclerie | And the problem appears to be intermittent! |
| 16:42.00 | pclerie | End of problem description. |
| 16:42.27 | pclerie | Suggestions? |
| 16:42.50 | file | break down the call when it happens and isolate. |
| 16:43.21 | file | use packet capture or rtp set debug on to check the media flow, and in the case of a packet capture you can examine the audio itself |
| 16:45.17 | pclerie | file: ok. I can try that. |
| 16:45.49 | pclerie | Still trying to figure out how to read SIP packets. |
| 16:46.51 | pclerie | Any clues as to what could cause that problem? |
| 16:47.32 | file | GSM gateway or Asterisk isn't getting the media |
| 16:47.38 | file | Asterisk itself just acts a packet forwarder in that case |
| 16:47.41 | file | it gets audio, it forward sit |
| 16:47.44 | file | rinse and repeat |
| 16:50.15 | pclerie | I see. Ok. Thanks. |
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| 16:58.05 | hursjohn | Hey all, I have a E&M Wink trunk. I'm trying to answer a channel and send another wink in order for my switch to send me more information on the trunk. I'm hoping that there is a way to do this in my extension config. |
| 17:02.20 | [TK]D-Fender | I don't know of any way with DAHDI to trigger a Wink "just because" |
| 17:03.21 | pclerie | file: Just tried a call with rtp debug on. On a bad call I only get Sent RTP packet to lines. And it's sending packets to the phone and to the GSM gateway, but neither seem to be responding. |
| 17:04.35 | [TK]D-Fender | so far that doesn't prove what was negotiated |
| 17:05.02 | [TK]D-Fender | Without proofe of where it was told to go I have no reason to believe it was told to you to the right place at all |
| 17:05.37 | hursjohn | I've been working with my switch vendor, Genband. They have a call come in on a trunk group with a spacific extension. Once the call is answered, the switch is expecing a wink back to send the next set of digits. Googling has so far yeilded no results, so this is kinda my last resort. |
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| 17:06.26 | [TK]D-Fender | hursjohn, if you explicitly answer the channel you don't get the wink? |
| 17:06.56 | hursjohn | There is the answer() wink, but I'm trying to get another one. |
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| 17:53.21 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 18:34.08 | ntwrk | Hi everyone. Does anybody have a good understanding of Asterisk RealTime for loading sip friends? We are loading our agents from the database now, which works for registration but Asterisk is not tracking state for those agents who were loaded dynamically. If loaded from the local sip.conf things work as expected but when registering as a user from the database Asterisk thinks the agent is always in Ready status even when they are on a |
| 18:34.08 | ntwrk | <PROTECTED> |
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| 18:55.52 | ntwrk | it seems like even though the queues the realtime members can join my locally configured queues, the eventmemberstatus=yes parameter is not applying to them or the QueueMemberStatus events are not being emitted/not associating with the realtime members |
| 19:06.54 | robmal | You've got state_interface set incorrectly. |
| 19:20.29 | ntwrk | now that I am logging QueueMemberStatus events, I can see that it sets 5 (Unavailable) correctly when I close my softphone but on ringing/busy/etc it just emits more 1 (available) events |
| 19:21.16 | robmal | You've got state_interface set incorrectly. |
| 19:21.24 | ntwrk | state_interface is set on AddQueueMember if I remember correctly |
| 19:21.35 | robmal | Indeed. |
| 19:22.33 | ntwrk | the thing is, it is setting state_interface the same way for local vs realtime users |
| 19:22.56 | ntwrk | ${chan_proto}/${ext} so SIP/whatever |
| 19:23.17 | ntwrk | does it need to be set differently for realtime agents? |
| 19:25.14 | ntwrk | QueueMemberStatus can have a 'membership' of dynamic/realtime/static, when doing AddQueueMember does it need to know they are a realtime user? |
| 19:25.17 | robmal | Nope, but it works fine for me on 13.18.2 |
| 19:25.24 | robmal | With members like so: |
| 19:25.28 | robmal | <PROTECTED> |
| 19:25.53 | robmal | Where 8623 is agents internal number and sip/asf is the device he's at. |
| 19:26.58 | robmal | They can join a queue both by dialplan and via AMI |
| 19:27.52 | robmal | I don't remember what that /n does but i recall it being important ;-) |
| 19:30.16 | file | it stops optimization and ensures the Local channel will continue to exist, even if it could go away and each side could talk directly |
| 19:31.16 | robmal | Thanks. So indeed, useful ;-) |
| 19:34.30 | ntwrk | it seems like it must be setting state_interface correctly because they do go unavailable if the phone/softphone is turned off and they are still in queue |
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| 19:35.19 | ntwrk | but it doesn't recognize when a peer registered through realtime goes into ringing/busy/hold and just emits an available event |
| 19:37.39 | robmal | What about pauses? |
| 19:39.03 | ntwrk | QueuePause does work |
| 19:40.43 | robmal | ringinuse=yes maybe? |
| 19:46.59 | ntwrk | ringinuse wasn't set for realtime users, i set it to no but it still just emits status 1 events when ringing or busy |
| 19:47.31 | ntwrk | i'm also setting callevents, callcounter both to yes |
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| 19:49.48 | robmal | When the user is busy - what does core show hints say? |
| 19:50.23 | ntwrk | no registered hints |
| 19:51.57 | ntwrk | but it says the same thing when I try with a local user and it does emit a 3 (busy) for the local user |
| 20:00.32 | robmal | Try rtcachefriends=yes |
| 20:00.39 | robmal | I'm running out of ideas ;-) |
| 20:01.11 | zigggggy | are you guys familiar with any solutions that provide an android\ios app for visual voicemail ? |
| 20:01.31 | ntwrk | no worries man, I appreciate the help |
| 20:01.57 | ntwrk | if we can't get it with any of your suggestions I may install a copy of 13 and try it there |
| 20:02.25 | ntwrk | we are running 11 and don't really have time to convert our dialplans and custom software for 13 but we need realtime for the new autoprovisioning we built :P |
| 20:02.52 | igcewieling | ntwrk: do you use CEL? |
| 20:02.56 | ntwrk | is rtcachefriends a sip friend parameter? |
| 20:02.59 | ntwrk | yeah we do |
| 20:03.04 | robmal | Yup. |
| 20:03.11 | igcewieling | the format of them changed |
| 20:03.44 | ntwrk | from 11->13? |
| 20:05.22 | igcewieling | ntwrk: It might be 14 or 15. look in the UPGRADE*.txt files on Asterisk 13. If the change is in 13, it will be mentioned in the 12 or 13 UPGRADE* files. |
| 20:07.17 | ntwrk | rtcachefriends didn't do it :( |
| 20:07.51 | ntwrk | I'm logging all events from asterisk manager and It looks like the AgentConnectEvent, NewstateEvent, BridgeEvent are all normal |
| 20:08.14 | robmal | I stand corrected, rtcachefreinds should be in [global] |
| 20:08.18 | ntwrk | it's just QueueMemberStatusEvent that is wrong and says 1 when it shouldn't be |
| 20:09.00 | robmal | Enable queue_log if you haven't already and check if there's a connect event. |
| 20:09.00 | ntwrk | when configuring from the database, can you just enter a row with the name global to set global params? |
| 20:09.37 | ntwrk | yeah, the connects do exist in queue_log |
| 20:10.14 | ntwrk | and an agent from realtime can even take calls and speak to a customer normally, but since it thinks they are in 1 it will continually ring all calls in the queue to them |
| 20:11.37 | robmal | You can (and maybe should?) have a normal sip.conf file with [global] despite having all your peers in the database. |
| 20:11.46 | igcewieling | I found with realtime, in order to pick up new peers on static IPs (i.e. no registration) I had to run a "sip reload" anyway so what is the point of realtime in that setup. |
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| 20:12.24 | ntwrk | oh cool, it can load general params from the flat file for db users? |
| 20:12.30 | robmal | Sure. |
| 20:12.33 | robmal | Wait. |
| 20:12.53 | robmal | igcewieling: I had to do an ugly hack of my own and this might be the case. |
| 20:12.58 | ntwrk | for us, our agents are dynamic and we are authing/provisioning from our crm so it makes sense to use realtime |
| 20:13.03 | robmal | ntwrk: Do queue show every time you can. |
| 20:13.17 | ntwrk | queue show will show the agent in queue correctly |
| 20:13.28 | ntwrk | but they will always be (not in use) whether they are on the phone or not |
| 20:13.43 | robmal | So this is not the case :-/ |
| 20:14.58 | ntwrk | in extconfig I am loading the table as sipfriends, would it need to be sipusers or sippeers instead? |
| 20:17.58 | robmal | sippeers => odbc,xxxxxx,sippeers |
| 20:18.01 | robmal | Is all you need. |
| 20:18.34 | ntwrk | rob |
| 20:18.35 | ntwrk | you |
| 20:18.36 | ntwrk | are |
| 20:18.38 | ntwrk | the |
| 20:18.40 | ntwrk | MAN |
| 20:18.52 | robmal | Next time read the docs? ;-) |
| 20:18.59 | ntwrk | rtcachefriends in the general section of sip.conf fixed it |
| 20:19.04 | robmal | Oh. |
| 20:19.05 | robmal | ;-) |
| 20:19.06 | robmal | Well. |
| 20:19.20 | robmal | GL&HF |
| 20:19.24 | ntwrk | haha I did read those other docs but I missed rtcachefriends somehow |
| 20:20.15 | robmal | Now you can complain to file the docs for queues weren't updated since 1.6 ;-) |
| 20:20.26 | ntwrk | XD |
| 20:22.13 | robmal | file: https://wiki.asterisk.org/wiki/dosearchsite.action?where=AST&spaceSearch=true&queryString=rtcachefriends Maybe worth mentioning somewhere that this option still exists? |
| 20:23.07 | file | chan_sip is legacy code, it doesn't use recent stuff which enforces documentation - its documentation is in the sample sip.co |
| 20:23.14 | file | er sip.conf |
| 20:23.29 | robmal | Understood. |
| 20:23.36 | robmal | ntwrk: It's your fault. |
| 20:24.32 | ntwrk | should rtcachefriends=yes maybe be the default behavior? |
| 20:24.43 | ntwrk | lol |
| 20:24.49 | ntwrk | I accept full responsibility |
| 20:24.55 | robmal | It's legacy code, don't use it if you want to live long and prosper. |
| 20:25.17 | file | chan_sip is community supported so such a change would have to be contributed by someone, and changing behavior that has existed for probably 10 years on people is... not a road that one goes down easily |
| 20:25.54 | ntwrk | fair enough, maybe we should make the jump to pjsip |
| 20:26.05 | ntwrk | I do want to live long and prosper |
| 20:29.52 | ntwrk | well, so long and thanks for all the fish.. err help |
| 20:34.33 | robmal | Remember to take your towel with you and in case you need anything else you know where to look ;-) |
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| 22:15.34 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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