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| 14:18.58 | *** join/#asterisk n8n8 (~nate@204.9.88.4) |
| 14:23.24 | n8n8 | Hey there, I'm having an issue where my asterisk 13.23.1 is occasionally not sending out the re-invite for some forwarded calls. I've worked with the upstream and they've pinpointed that they do not see the re-invite from my asterisk server. I'm trying to figure out what the difference is between working calls and non-working ones. I've got about 150 pjsip endpoints off this instance and, this seems to have just reached a point where I'm now hearing about it fro |
| 14:23.25 | n8n8 | m my customers |
| 14:24.24 | Samot | 13 is no longer supported |
| 14:25.16 | Samot | But we would need to see debugs to tell you anything. |
| 14:26.34 | n8n8 | I thought 13 is LTS... (https://www.asterisk.org/downloads/asterisk/all-asterisk-versions/) I'm working on upgrading, too, but that will take a bit. I'll post a link to logs in a sec |
| 14:26.54 | Samot | It was |
| 14:27.03 | Samot | Its EOL |
| 14:27.28 | file | security fix only, technically |
| 14:27.42 | Samot | 16 and 18 are the two current versions. Both LTS. |
| 14:28.04 | n8n8 | gotcha. I'll move that up my list. It's been rock solid for me for the past few years, and I tend to follow the "if it ain't broke" too often |
| 14:28.25 | Samot | But without seeing data of the calls...we cant say what the problem is. |
| 14:30.20 | igcewieling | n8n8: generally if you are using a old version of Asterisk it is mostly pointless asking about it. |
| 14:30.37 | igcewieling | I use a lot of older Asterisk versions, I know. |
| 14:31.42 | n8n8 | I've been hesitant to upgrade as the link I posted above still lists 13.... In this case, I'm hoping to buy myself some time to do my upgrade correctly, rather than rush the deployment out |
| 14:31.53 | n8n8 | But, yeah, I get it |
| 14:32.08 | file | it's listed as it is still security supported, it goes full EOL end of the yearish |
| 14:32.19 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions dates are here |
| 14:32.26 | file | October 24th. |
| 14:32.53 | file | can update the page though to make it clearer |
| 14:34.15 | n8n8 | Thanks! I'll watch there instead. |
| 14:34.16 | file | done |
| 14:34.22 | Samot | So what that means is if this is a bug, it is not being touched. |
| 14:34.34 | Samot | Unless this is security related |
| 14:34.51 | n8n8 | Fully understood. I'm hoping I can figure out if this is an issue on my end, or, if upgrading will make the issue go away. |
| 14:35.03 | Samot | Ive asked for debugs twice |
| 14:35.09 | n8n8 | Here are my relevant full log messages: https://pastebin.com/Nem5CPJg |
| 14:35.24 | n8n8 | Sorry, I had some grepping to do |
| 14:35.30 | Samot | No. Real SIP debugs. |
| 14:35.45 | Samot | If this is PJSIP then pjsip set logger on |
| 14:35.47 | n8n8 | I'm working on the console messages with debug set to 99 , they don't look all that different |
| 14:35.52 | Samot | No |
| 14:35.56 | Samot | Sip debugs |
| 14:36.03 | n8n8 | Ah, gotcha. Pjsips in a moment |
| 14:37.00 | Samot | Need to see a call that isnt working right |
| 14:37.06 | Samot | Start to finish |
| 14:37.21 | *** join/#asterisk cyberfab007 (~djfab007@c-73-1-175-240.hsd1.fl.comcast.net) |
| 14:37.40 | cyberfab007 | Yo yo yo , must be a decade since I was last in this room lol, How has life been |
| 14:40.07 | seanbright | thanks for asking... i guess we should start from the beginning |
| 14:40.09 | igcewieling | cyberfab007: it as not changed at all. |
| 14:40.25 | seanbright | let me take you back to 2011... |
| 14:41.23 | Samot | Narrator: Actually is was 2010. |
| 14:41.25 | seanbright | cyberfab007: you still in florida? |
| 14:41.34 | cyberfab007 | hehe |
| 14:41.39 | cyberfab007 | How you been man ? |
| 14:42.07 | seanbright | great, great |
| 14:42.23 | cyberfab007 | Acutally Left lived in Bali, then moved to NYC the came back to florida when the pandemic started |
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| 14:42.44 | seanbright | wow, nice |
| 14:42.45 | cyberfab007 | So yes I am still in Florida sorta lol, |
| 14:43.03 | seanbright | well welcome back |
| 14:43.10 | cyberfab007 | Hey listen I have been out of the game for a while and need a vicidial install done and interconnect with twillio , willing to pay |
| 14:43.45 | Samot | Vicidial? |
| 14:43.46 | Samot | No. |
| 14:43.54 | cyberfab007 | Its for my video game company , cheeck it out ! we are being funded! |
| 14:44.05 | seanbright | doesn't vicidial have a hosted version? |
| 14:44.22 | cyberfab007 | connect.gameprofits.co/paid-to-play |
| 14:44.36 | cyberfab007 | Well ya if wanna pay 5X the cost to hose vicidial lol |
| 14:45.04 | cyberfab007 | That shit works on a 40/month VPS , so I would rather throw one of y'all a few hundred bucks for setup |
| 14:45.14 | seanbright | roger that |
| 14:45.28 | cyberfab007 | I used to use the goautodial |
| 14:46.02 | seanbright | yeah i've never dealt with vicidial |
| 14:46.09 | seanbright | i only hear good things though |
| 14:46.20 | cyberfab007 | but they really killed that project with the shitty updated version and the old version which was bad ass is broken now when you try to use it cause the dependecy resolutions are all out of whack |
| 14:46.20 | Samot | We must hang out in different circles. |
| 14:47.15 | cyberfab007 | I remember 10 years ago trying to sell doctors on the telehelath with the video phones , and they told me I was nuts ! |
| 14:48.17 | cyberfab007 | Vicidial is a beast, I remember first learning about it and doing that Training when I was in Toronto , I got to meet Kevin Flemming and Marc at that small demo they did back in I think it was 07-08 |
| 14:48.39 | cyberfab007 | vicidal has always been a beast |
| 14:49.13 | Samot | Vicidial 99.99% of the time indicates outbound dialing campaigns. |
| 14:49.32 | Samot | The same campaigns that my customers say "Can you block this number" |
| 14:49.51 | seanbright | OB dialing is so hot right now though |
| 14:50.22 | seanbright | i have this software that will take a state and just randomly generate a 10 digit caller id that appears to be from that state |
| 14:50.22 | Samot | I mean anything that starts with a V and some numbers as the caller name, blocked. |
| 14:50.32 | seanbright | so easy, the code runs so fast |
| 14:53.17 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-olkqmgngapewavso) |
| 14:53.18 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:54.52 | igcewieling | Telemarketers are evil and must be destroyed. |
| 14:59.06 | seanbright | i would just like to say that since i said "i only hear good things though" everything has been a joke |
| 14:59.32 | seanbright | exact for that |
| 14:59.35 | seanbright | and that |
| 14:59.39 | seanbright | ... and that |
| 14:59.57 | seanbright | grrrr, except* |
| 15:00.47 | file | stares at seanbright |
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| 15:18.25 | n8n8 | Sorry this took so long, I was trying to remove all the other activity from the file: https://pastebin.com/CjBQenwS |
| 15:28.24 | n8n8 | While I'm thinking about digging up and filtering my PJSIP logs, are there some best practices / tools / tips around debug logging and data retention? I mean I rarely have issues, but when I do, it seems like there's a lot of noise to filter through to even be able to isolate the call in question.... |
| 15:28.49 | Samot | n8n8: OK so this isn't call forwarding. |
| 15:29.05 | Samot | n8n8: What re-invite should Flowroute be seeing? |
| 15:30.21 | n8n8 | Good question. They told me they went through the working and non-working calls on their side, and for the non-working ones, (where non-working is no audio, but connected call), they did not see the re-invite from me. |
| 15:30.52 | Samot | Wait, the none working is no outbound audio? |
| 15:30.52 | n8n8 | I don't know exactly which message is missing as I do not have good tooling to compare the working vs non-working messages |
| 15:31.00 | n8n8 | Correct. |
| 15:31.08 | Samot | And this one didn't have audio? |
| 15:31.10 | n8n8 | Well, neither direction |
| 15:31.13 | igcewieling | n8n8: the most important thing is to make NO changes on the debug output you paste, EXCEPT for actual passwords. |
| 15:31.29 | n8n8 | Yes, this call had no audio. |
| 15:31.49 | n8n8 | I have the full output as well, it's just bigger and noisier. |
| 15:31.51 | Samot | OK so first off... |
| 15:31.57 | Samot | PJSIP/anonymous-00002b96 <-- That's already a bad start. |
| 15:32.31 | Samot | That means your trunk with Flowroute is *not* setup correctly and you actively setup an anonymous endpoint for that fact. |
| 15:32.41 | Samot | Since there is no anonymous endpoint by default for PJSIP. |
| 15:33.34 | Kobaz | seanbright: in theory, STIR/SHAKEN will start to put a damper on that auto-generated callerid crap |
| 15:34.04 | n8n8 | Gotcha, but, I'm processing thousands of calls a month like this, so I'd like to set this aside unless it is a likely culprit for this no-audio issue. |
| 15:34.13 | igcewieling | Kobaz: eventually. |
| 15:34.25 | n8n8 | I have no doubt that I've got some warts on my config |
| 15:35.44 | Samot | n8n8: Do you have direct_media=no on the trunks and endpoints? |
| 15:35.53 | Kobaz | anyone have any feedback on using google-fi ? |
| 15:35.59 | n8n8 | One sec, I'll verify |
| 15:38.55 | n8n8 | For endpoints, yes, direct_media = no, For trunks it seems it is the default, as I cannot find it specified |
| 15:39.30 | igcewieling | Kobaz: It is google. they will discontinue it, change it to make it not compatible, or abandon it. Why bother even thinking about it. |
| 15:39.37 | Kobaz | ahahaha yeah |
| 15:41.54 | igcewieling | 2 weeks after I moved some services to Google Compute Services, there was talk about Google getting out of that industry. Thankfully, it was an unfounded rumor, but I was worried. |
| 15:42.11 | Kobaz | hah, aw |
| 15:43.54 | igcewieling | I still really dislike Gmail, but since we moved away from Exchange, I no longer have to deal with our "exchange admin" so I'm happy. |
| 15:46.41 | n8n8 | Samot: you mentioned what I am doing is not forwarding. Way back when when I set it up I was following : https://www.voip-info.org/asterisk-call-forwarding/ which is why I called it forwarding. Is there a better way to "redirect" inbound calls to an external number? |
| 15:47.46 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 15:48.03 | Samot | Asterisk is a b2bua |
| 15:48.50 | Samot | Unless you are using proper redirects then using Dial() to open a new channel is just another call leg from Asterisk. |
| 15:50.31 | n8n8 | Up until a few days ago, I didn't have a problem with the dial approach. Is there a different dialplan application I could be using to trigger the redirects? I'm completely unfamiliar with them |
| 15:51.08 | Samot | Go add direct_media=no on the trunks |
| 15:51.22 | Samot | Because by default it is yes |
| 15:51.51 | Samot | So the call is trying to have Flowroute have the media directly flow between itself |
| 15:52.12 | Samot | And flowroute doesnt have their own media |
| 15:52.24 | Samot | They use their upstreams |
| 15:52.33 | n8n8 | Would the default direct_media=yes on the trunks result in sometimes audio working, and sometimes not? |
| 15:52.44 | Samot | It could |
| 15:53.04 | n8n8 | Dependent on which of flowroute's upstreams we're talking to? |
| 15:53.10 | Samot | Yes |
| 15:53.37 | n8n8 | I'll test this right now. Should only take a moment |
| 16:01.33 | n8n8 | Seems like that did the trick. I guess I was just getting lucky all this time that it worked at all. Awesome.... Thanks for working me through that. |
| 16:02.20 | Samot | Yup |
| 16:02.33 | n8n8 | When you mentioned redirects, could that be a drop in replacement in the dial plan for Dial? Or, is that a whole other bag of cats? |
| 16:02.51 | Samot | Wouldnt see if from endpoint to trunk and vice versa because the endpoint config stopped it |
| 16:03.41 | Samot | But since its going in and out the same trunk that allows it. It was always having the RTP flow directly between the two sources. |
| 16:04.13 | Samot | And sometimes the two sides couldnt flow between each other. |
| 16:04.30 | Samot | Now, it flows through Asterisk. |
| 16:04.49 | n8n8 | Gotcha. Regarding the RTP flow. Makes sense. |
| 16:05.05 | Samot | And to do proper forwarding you need return proper SIP replies like a 302 redirect. |
| 16:05.09 | igcewieling | Does ANYONE ever successfully use directmedia? |
| 16:05.31 | Samot | Telling Flowroute that Asterisk isnt taking the call, the destination is. |
| 16:05.53 | Samot | igcewieling: In an internal setup |
| 16:05.54 | igcewieling | IIRC the Transfer application sends 302s |
| 16:05.54 | n8n8 | Well, it seems I was on my trunk endpoint for the past several years, until a few days ago ... no idea what changed where |
| 16:06.06 | Samot | n8n8: the carriers |
| 16:06.16 | Samot | n8n8: perhaps a route was changed |
| 16:06.28 | n8n8 | Yeah. It would seem. I'm just amazed I made it this long |
| 16:06.34 | Samot | Outside of your control |
| 16:06.56 | n8n8 | This config has been in place since 13.23.1 was new |
| 16:06.57 | Samot | n8n8: Now fix your trunk |
| 16:07.16 | n8n8 | You mean the anonymous bit? |
| 16:07.34 | Samot | n8n8: You should not be accepting calls over an anonymous endpoint |
| 16:07.49 | Samot | It means you dont know the source |
| 16:08.06 | Samot | And are allowing unknown sources |
| 16:08.10 | n8n8 | I firewalled it off by IP ranges outside asterisk. But, I'd be happy to do it correctly |
| 16:09.16 | n8n8 | IIRC I was struggling to get PJSIP going back then coming from the chan_sip configs, and landed on this config that seemed to consistently work... |
| 16:10.10 | Samot | OK, well voip-info.org is random people posting things to help |
| 16:10.29 | Samot | There's not really quality assurance on said help and some of it is ancient. |
| 16:10.50 | Samot | That means there could be cases a config sample is a "I cobbled this together to make it work as basic as possible" |
| 16:11.32 | n8n8 | That quote probably accurately describes how I got this system up and running. |
| 16:11.38 | igcewieling | voip-info.org should be the LAST place you check and, as Samot said, it is full of outdated and flat out wrong information |
| 16:11.51 | n8n8 | I'd like to think I've been improving it bit by bit over the past few years. |
| 16:12.08 | n8n8 | I'll be sure to avoid void-info.org in the future |
| 16:12.25 | n8n8 | Google is not always my friend it would seem :) |
| 16:12.39 | igcewieling | Once Digium provided good online documentation voip-info.org became nearly irrelevant. |
| 16:12.41 | n8n8 | (I know they never are) |
| 16:13.26 | igcewieling | i.e. https://wiki.asterisk.org/wiki/display/AST/Home |
| 16:13.35 | n8n8 | I've definitely appreciated Digium's docs. |
| 16:13.47 | n8n8 | I've come across the wiki periodically as well. |
| 16:16.25 | n8n8 | I'm partly in the position of having a system that has appeared to be stable and functional for a few years. I'm leery of poking around and changing things... I think my next big move is to do a bit of a re-architecting to enhance my failover capabilities. Which might have me re-working the rest of the config significantly. |
| 16:17.35 | n8n8 | In any case, I really appreciate the help I've been able to find here over the years. I'm not here often, but when I am, my issues tend to get sorted out. And that's a great thing |
| 16:22.56 | Samot | Plus it's owned by 3CX now. |
| 16:24.47 | Samot | And they are literally paying for people to be active. They give away cash to the top 5 active users every quarter. |
| 16:25.54 | Samot | I mean, I got a $100 of the deal just for being active. |
| 16:26.45 | n8n8 | That's nice of them. I can see how that helps them from a marketing perspective too. They seem to be betting a lot on asterisk |
| 16:26.49 | Samot | Well |
| 16:27.07 | Samot | That activity was due to me actively mocking a vendor and a project that was being worked on. |
| 16:27.16 | Samot | So it's all in perspective. |
| 16:27.31 | n8n8 | :) |
| 16:30.20 | igcewieling | Looks like I might be able to get rid of my Sangoma transcoding cards soon. |
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| 18:24.10 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.2, 16.16.2 (2021/03/04) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); 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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| 18:34.33 | *** topic/#asterisk by kharwell -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.3.0, 16.17.0 (2021/03/25) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); 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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| 18:36.41 | igcewieling | Thank you for fixing chan_local: Multistream support breaks T.38 faxing (Reported by Matthias Hensler) !!!!!!! |
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