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| 00:17.18 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 03:08.31 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 05:10.38 | *** join/#asterisk masked (~masked@hpavc/masked) |
| 05:10.43 | masked | hi |
| 05:10.48 | masked | is anyone good with chan_dongle? |
| 05:11.33 | masked | i have loaded the module, the dongle initialises, dongle show device state shows everything is correct, even reports the RSSI |
| 05:11.48 | masked | but then it tries to send AT+CSQ to the modem and times out on an OK |
| 05:12.16 | masked | if i try AT+CSQ\n in minicom it returns the rssi and and OK just fine |
| 05:13.01 | masked | <PROTECTED> |
| 05:13.04 | masked | thats what i get |
| 05:14.08 | masked | https://pastebin.com/EpAHH6wP |
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| 05:26.13 | masked | brb |
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| 14:47.06 | *** join/#asterisk EmleyMoor (42b789682f@firthpark.tinsleyviaduct.com) |
| 14:48.46 | EmleyMoor | How would I put in my dialplan something to process the status field from this: http://www.telepest.co.uk/01639896465?f=json ? Basically I want to get the status as a variable in the dialplan to decide how to handle the incoming call |
| 14:50.04 | [TK]D-Fender | "core show function CURL" |
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| 14:52.16 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 14:55.23 | Samot | Oh dear god, it's an AOL spam system for telephony. |
| 14:58.51 | Samot | Oh BTW, new FCC rules for 911 calls. |
| 15:03.01 | EmleyMoor | [TK]D-Fender: Thanks, I'll start there |
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| 15:29.03 | *** join/#asterisk effprime (~effprime@unaffiliated/effprime) |
| 15:34.47 | effprime | not sure how active this channel is but I'll try |
| 15:35.10 | effprime | does anyone use the SimpleAction function in the asterisk python library? |
| 15:35.30 | effprime | I am very new to PBX and trying to get the different variables |
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| 16:38.38 | *** join/#asterisk infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net) |
| 16:38.38 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 16:43.17 | effprime | [TK]D-Fender: Unfortunately I can't but I understand your point |
| 16:43.29 | effprime | I am not allowed to share anything related to my job's code base |
| 16:46.11 | Samot | I'm not sure out a debug output is sharing code base. |
| 16:46.41 | [TK]D-Fender | Nobody can tell what youève screwed up without the evidence |
| 16:46.44 | Samot | Either via the verbose logs or a pcap. |
| 16:46.54 | [TK]D-Fender | So you should understand that asking it is largely a waste of time |
| 16:46.56 | Samot | None of that would expose code base. |
| 16:47.01 | effprime | I just said I understand your point |
| 16:47.10 | effprime | [TK]D-Fender |
| 16:47.23 | Samot | My point is, debug details aren't going to expose IP. |
| 16:47.39 | [TK]D-Fender | And as Samot said seeing the dial command and what follows isnèt Ãyour code baseà |
| 16:48.09 | effprime | I can see why you think that, but it would stem into how we configure things |
| 16:48.17 | effprime | I will see if I can get permission |
| 16:48.21 | [TK]D-Fender | 1 dial command? |
| 16:48.32 | Samot | Timeout is a timeout. |
| 16:48.33 | [TK]D-Fender | That's ridiculous. |
| 16:48.41 | effprime | wow this guy is a prick |
| 16:48.44 | Samot | 32 seconds after the call connects means RTP issues. |
| 16:48.45 | [TK]D-Fender | It's networking one way or another |
| 16:48.52 | Samot | That's the best we can give you. |
| 16:48.57 | effprime | thanks for trying |
| 16:49.02 | *** part/#asterisk effprime (~effprime@unaffiliated/effprime) |
| 16:49.11 | [TK]D-Fender | Either in what you're sending, what's ggetting mangled in between, or what's happening on the recieving end |
| 16:49.18 | Samot | They left. |
| 16:49.21 | [TK]D-Fender | yup |
| 17:06.20 | *** join/#asterisk jasonwert (~w3rt@198.167.245.16) |
| 17:56.11 | *** join/#asterisk effprime (~effprime@unaffiliated/effprime) |
| 17:56.41 | effprime | hi, i just wanted to apologize for the name calling earlier, I was just frustrated over my issue and trying to follow my company's policy (we do govt work so it's more than just IP) as a new employee |
| 17:57.01 | effprime | I am going to see what else I can figure out on my own, but felt the need to apologize. thanks |
| 17:57.28 | effprime | specifically this is directed at [TK]D-Fender |
| 17:58.02 | [TK]D-Fender | All good. |
| 17:59.01 | [TK]D-Fender | Companies come up will all sorts of policies that completely impractical and ineffective for goals they don't help achieve |
| 17:59.09 | [TK]D-Fender | that are* |
| 17:59.52 | [TK]D-Fender | Seems they're telling you to fix the car but you're not permitted to open the hood |
| 18:00.26 | [TK]D-Fender | Which is ridiculous ... but it's not your fault |
| 18:01.04 | effprime | I agree with you very much |
| 18:01.11 | effprime | nobody here knows VOIP at all |
| 18:01.40 | effprime | fortunately we are just testing VOIP traffic, so it's not a full setup |
| 18:01.58 | effprime | but they configured their own setup before I was hired and I have no idea what it does or if it should be changed |
| 18:09.27 | [TK]D-Fender | You need to validate that the IP's being offered are valid, if everything matches on both ends, checking firewalls, etc |
| 18:09.41 | [TK]D-Fender | Which means you need to see it break in real-time... |
| 18:15.36 | effprime | I am doing it all via 0.0.0.0 or localhost since it's just a local testing setup |
| 18:17.32 | [TK]D-Fender | You have another UA on that same host? |
| 18:18.03 | effprime | I'm sorry, what is a UA? |
| 18:18.16 | effprime | Just using Asterisk and a SIP client |
| 18:19.08 | [TK]D-Fender | that other client |
| 18:19.30 | effprime | I'll clarify some more |
| 18:19.30 | [TK]D-Fender | it's also on that server directly? |
| 18:19.34 | effprime | yes |
| 18:19.38 | effprime | its all on my machine |
| 18:20.00 | effprime | I am sending an Action through logging into my AMI via Python's asterisk library |
| 18:20.14 | effprime | our codebase is Python so it's essential I can do it this way |
| 18:20.38 | effprime | logging in works fine, I receive the call on the client |
| 18:20.45 | effprime | but when I answer, it fails after 32 seconds |
| 18:21.04 | [TK]D-Fender | "an action" should be clarified |
| 18:21.25 | effprime | its an originate action |
| 18:21.32 | [TK]D-Fender | better |
| 18:21.37 | [TK]D-Fender | And was what I was suspecting |
| 18:21.38 | effprime | channel="SIP/localhost/extensionname" |
| 18:21.50 | [TK]D-Fender | that sounds circular |
| 18:21.58 | effprime | context = "send" which is predefined in our extensions.conf |
| 18:22.15 | [TK]D-Fender | I don't understand the point |
| 18:22.35 | [TK]D-Fender | Why are you doign a SIP call from your server to itself? |
| 18:23.00 | file | oh, hi |
| 18:23.01 | [TK]D-Fender | that entire layer adds no value that I can imagine yet |
| 18:23.15 | effprime | just for testing, in the real world it will be to another IP |
| 18:24.56 | [TK]D-Fender | so far I don't see what that "tests" |
| 18:25.16 | [TK]D-Fender | You're left with what is functionally 2 local channels |
| 18:25.54 | [TK]D-Fender | But shoved SIP in the way ... which is actually the part that is failing and for which your "test" is the least meaningful |
| 18:26.10 | [TK]D-Fender | that looks like all-pain, no gain |
| 18:26.17 | *** join/#asterisk somepoortech (~somepoort@72.12.70.165) |
| 18:26.59 | effprime | [TK]D-Fender: our project is for generating VOIP traffic |
| 18:27.07 | effprime | so I am testing that generation by sending a test call |
| 18:27.14 | effprime | there's very little I can tell you with regards to why |
| 18:27.51 | [TK]D-Fender | Well basically all it does is load-test your server itself... and use comms it's supposed to be able to handle |
| 18:28.10 | [TK]D-Fender | go verify your ports are properly open, that it IS using the IP's you think it is, etc |
| 18:29.00 | effprime | I'm going to make a paste of just our sip.conf if you could help me understand what its doing |
| 18:31.57 | effprime | [TK]D-Fender: https://pastebin.com/w1V18PuS |
| 18:32.16 | effprime | then for [send] we have |
| 18:32.26 | [TK]D-Fender | Doesn't work that way. We need to verify how it's going throught he networking stack |
| 18:32.45 | effprime | could you help me with getting that info then? |
| 18:33.14 | [TK]D-Fender | You also have no HOST specified which means they need to be registered to |
| 18:33.34 | [TK]D-Fender | which you're not doing here. |
| 18:33.44 | effprime | registered to? |
| 18:33.50 | [TK]D-Fender | none of those should end up effectively allowing a connection from localhost. |
| 18:34.27 | [TK]D-Fender | this is all broken |
| 18:34.41 | effprime | it was written by someone else |
| 18:35.03 | effprime | I only need this asterisk setup to work locally, just for referene |
| 18:35.05 | [TK]D-Fender | your SIP call out is not using a peer entry at all, and those cannot match it wither. There is no way your system should accept a call like you've described at all |
| 18:35.42 | effprime | what does accept mean here? my empathy client receives a call |
| 18:35.52 | effprime | also I have two contexts defined |
| 18:36.38 | [TK]D-Fender | you server is calling itself |
| 18:36.48 | [TK]D-Fender | And from what you've shown it should REFUSE it at best |
| 18:37.13 | [TK]D-Fender | This is a dead end |
| 18:37.17 | effprime | I am logging into the AMI as user1 |
| 18:37.21 | [TK]D-Fender | no |
| 18:37.31 | [TK]D-Fender | AMI != SIP |
| 18:37.32 | Samot | 2:20:46 PM <effprime> but when I answer, it fails after 32 seconds |
| 18:37.35 | effprime | can you explain why I am seeing the call on the client then? |
| 18:37.37 | [TK]D-Fender | so that sip config means nothing to AMI |
| 18:37.48 | Samot | 12:48:45 PM <Samot> 32 seconds after the call connects means RTP issues. |
| 18:37.50 | *** join/#asterisk wonderworld (~wonderwor@unaffiliated/wonderworld) |
| 18:37.51 | [TK]D-Fender | what you Originate also has nothing to do with a SIP dlient |
| 18:37.52 | effprime | please understand I don't get this pipeline at all |
| 18:38.04 | wonderworld | hey all |
| 18:38.05 | Samot | effprime: Follow what I said. |
| 18:38.11 | [TK]D-Fender | That originate has NOTHING to do with your SIP client |
| 18:38.21 | effprime | Samot: can you be more specific how I would even follow that? |
| 18:38.28 | Samot | effprime: If the call is connecting but dying after 32 seconds, that's an RTP stream issue. The default is 32 seconds on the time out\ |
| 18:38.37 | effprime | the originate sends a command to the asterisk pbx right? |
| 18:38.47 | [TK]D-Fender | yes |
| 18:38.51 | Samot | It means that Asterisk isn't receiving an RTP stream from the other side. |
| 18:38.54 | [TK]D-Fender | and tells * to dial out. |
| 18:39.09 | [TK]D-Fender | Which in the sample you gave was "go talk to yourself" |
| 18:39.22 | [TK]D-Fender | which it should refuse |
| 18:39.31 | Samot | effprime: No audio, no call. So if there is no audio detected in 32 seconds, dead call. |
| 18:39.47 | effprime | so I need to send audio back via the client? |
| 18:39.52 | [TK]D-Fender | <effprime> channel="SIP/localhost/extensionname" <-- dead end here |
| 18:40.31 | effprime | if you want me to understand I have to understand the pipeline after asterisk receives the Originate command |
| 18:40.37 | effprime | so can we start there |
| 18:40.49 | Samot | There's no audio |
| 18:40.54 | Samot | Go prove what is going on with that |
| 18:41.03 | Samot | a full SIP debug/packet capture is needed. |
| 18:41.06 | Samot | Is the SDP right? |
| 18:41.19 | effprime | this test should play "hello-world" after picking up the phone |
| 18:41.21 | effprime | that is what I was told |
| 18:41.30 | Samot | OK. |
| 18:41.32 | effprime | I haven't shown you my extensions |
| 18:41.38 | Samot | It doesn't matter. |
| 18:41.39 | Samot | Stop. |
| 18:41.50 | Samot | Call goes out, call connects, call dies after 32 seconds. |
| 18:42.06 | Samot | That all points to an issue with the RTP/audio stream with the call. |
| 18:42.12 | Samot | And Asterisk is killing the call. |
| 18:42.20 | [TK]D-Fender | It shouldn't even connect <- |
| 18:42.24 | Samot | Stop. |
| 18:42.32 | [TK]D-Fender | He's looping it back with no peer that should ever match |
| 18:42.33 | effprime | yikes |
| 18:42.41 | Samot | Let's be clear. |
| 18:42.47 | [TK]D-Fender | no codec defined either yet |
| 18:42.51 | Samot | He's not doing anything but asking questions on IRC. |
| 18:42.52 | [TK]D-Fender | https://pastebin.com/w1V18PuS |
| 18:42.54 | igcewieling | Guy hot a custom license plate NULL. "First, Droogie got a parking ticket, incurred for an actual parking infraction. Then, once a particular database of outstanding tickets had associated the license plate NULL with his address, it sent him every other ticket that lacked a real plate. The total came to $12,049 worth of tickets. " |
| 18:42.56 | [TK]D-Fender | There's no "there" there |
| 18:42.57 | Samot | He didn't do that. |
| 18:43.03 | *** join/#asterisk miralin1 (~Thunderbi@host-46-45-217-207.stavropol.ru) |
| 18:43.24 | [TK]D-Fender | <[TK]D-Fender> <effprime> channel="SIP/localhost/extensionname" <-- dead end here <- no peer used to dial out, LOOPING to localhost (server itself) |
| 18:43.42 | effprime | I guess I need to read how this works, but I will consider the RTP issue first |
| 18:44.06 | [TK]D-Fender | effprime, No. You don't understand SIp peers on * itself at all yet |
| 18:44.20 | effprime | well I'm being told two different things here |
| 18:44.24 | effprime | can you see this from my POV? |
| 18:44.29 | Samot | OK... |
| 18:44.31 | [TK]D-Fender | Dno't worry about pot-holes when you don't even have the key int he ignition yet |
| 18:44.34 | Samot | What TK is valid. |
| 18:44.53 | Samot | The fact _YOU_ are claiming the call _DOES_ connect but 32 seconds it dies... |
| 18:45.02 | [TK]D-Fender | If what you showed was actual then it's broken |
| 18:45.03 | Samot | Leads to what I was saying. |
| 18:45.13 | *** part/#asterisk igcewieling (~ewieling@speedy-02.nyigc.net) |
| 18:45.14 | Samot | But since we can't see anything, we can't figure anything out. |
| 18:45.29 | [TK]D-Fender | if that assumption can be trusted there's also no codecs defined... what ar you going to have for RTP? |
| 18:45.47 | effprime | I have no idea how to answer that |
| 18:45.50 | effprime | hence why I need to read |
| 18:45.51 | Samot | These are poorly configured sip peers. |
| 18:45.58 | [TK]D-Fender | You need to learn the basics of * first |
| 18:46.06 | Samot | This Asterisk box is poorly configured overall I would guess. |
| 18:46.15 | effprime | of course it is |
| 18:46.18 | effprime | that's why I'm here |
| 18:46.55 | Samot | Well this is a company project to generate VOIP traffic... |
| 18:47.09 | Samot | The company should put some resources in hiring an Asterisk consultant. |
| 18:47.56 | effprime | welp, off to read |
| 18:47.58 | effprime | thanks for your time |
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| 22:00.07 | Kobaz | gah |
| 22:00.11 | Kobaz | the benefits of using old asterisk |
| 22:00.14 | Kobaz | CUT is broken |
| 22:00.59 | Kobaz | queue_members=SIP/c30030-161|0|0|SIP/c30030-161|SIP/c30030-161 agent = ${CUT(queue_members,|,1)}; |
| 22:01.05 | Kobaz | so this is -161... haha |
| 22:01.18 | Kobaz | it's ignoring the cut delimiter and using - anyway, but in a broken kind of way |
| 22:02.08 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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| 23:03.35 | *** join/#asterisk khronos (~khronos@072-238-027-220.res.spectrum.com) |
| 23:04.12 | khronos | Hi. |
| 23:04.55 | khronos | I've got someone asking me what's out there these days for a call rating gateway. |
| 23:05.38 | khronos | In my day job we use a high end solution and I know it will be out of this guy's budget. |
| 23:07.44 | khronos | From what he told me he's got a corp location and he wants to be able to make each branch location responsible for their own international charges. |
| 23:08.39 | khronos | He has told me that he has 10 branch office locations or so so small time in my opinion. |
| 23:24.12 | khronos | Right now he's got Windstream at his head end. |
| 23:25.01 | khronos | They have told him that they only send out a master bill and because he controls the branches any billing split he wants to do is on him. |