| 00:06.02 | *** part/#asterisk kharwell (kharwell@nat/digium/x-lthenbaclmtxujrs) |
| 00:23.08 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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| 01:20.09 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:20.09 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); 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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| 02:58.35 | snadge | does nat=yes in sip.conf general section require a restart or is reload enough? |
| 02:59.03 | [TK]D-Fender | reload should do |
| 03:01.05 | snadge | OK, I'm trying to figure out why asterisk is sending rtp to the address in the sdp |
| 03:01.34 | snadge | Even though nat=yes for both the general case and the registered extension |
| 03:02.15 | snadge | ie its honouring it, even though its unroutable |
| 03:02.46 | snadge | so it should be sending to the source address instead |
| 03:11.13 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 03:15.13 | snadge | I know why now :) |
| 03:15.29 | snadge | the documentation lies |
| 03:15.58 | snadge | Nat=yes does not mean it won't honour the sdp |
| 03:16.42 | snadge | it means it does until it receives an rtp packet, then it will correct IRS behaviour.. So I've learned about the nat on each side rtp deadlock behaviour |
| 03:17.05 | snadge | IRS, its.. Stupid phone |
| 03:18.10 | snadge | this is what happens when you try to do retarded stuff like set your externip to an internal ip |
| 03:18.37 | snadge | I suspect we need to define local networks to fix the nat behaviour |
| 03:23.31 | snadge | I am sorry to bother you with trivial crap like this TKD.. You've been around for ages :) |
| 03:38.12 | *** join/#asterisk infobot (ibot@rikers.org) |
| 03:38.12 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); 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 |
| 03:42.20 | Samot | Your SDP should not contain an RFC1819 address when going over NAT. |
| 03:57.36 | Samot | Also nat=yes has been deprecated since Asterisk 10 |
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| 04:41.51 | snadge | <PROTECTED> |
| 04:42.45 | snadge | It gets worse, I'm using proprietary shite.. Pbxware from bicom |
| 04:43.35 | snadge | I started a new job, unfortunately more VoIP related than the last |
| 04:44.19 | snadge | I think I'd rather chop my dick off than have anything to do with pbxs or voice |
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| 05:04.13 | Samot | Deprecated means it might now work properly down the road. |
| 05:04.55 | Samot | If you are using v13/v15 you shouldn't use v10 configurations |
| 05:05.10 | drmessano | To be fair |
| 05:05.13 | Samot | Sure they will probably work for the most part but things change. |
| 05:05.16 | drmessano | It's really a 1.0 config |
| 05:06.04 | Samot | But RTP goes to the IP:Port presented in the SDP body. |
| 05:10.34 | drmessano | 23:41:52 <snadge> Haha, because deprecated means nobody uses it anymore.. If only ;) |
| 05:10.41 | drmessano | ^ It literally doesn't |
| 05:11.11 | drmessano | Deprecation can come in many forms. In this case, as in many cases with Asterisk, settings have changed and been improved |
| 05:11.22 | Samot | Right. |
| 05:11.29 | drmessano | Feel free to Google for the non-15 year old options for nat= |
| 05:11.31 | Samot | Asterisk rarely removes old setings. |
| 05:11.35 | Samot | Asterisk rarely removes old settings. |
| 05:12.02 | Samot | Just marks them deprecated with a note that says "Use X instead" |
| 05:12.07 | drmessano | or just keep upgrading and come back and complain when it doesnt work in 16 or so |
| 05:12.19 | Samot | canreinvite still works |
| 05:12.21 | drmessano | But also |
| 05:12.27 | drmessano | If you ever looked at a log |
| 05:12.31 | drmessano | Its printed there |
| 05:12.32 | Samot | But directmedia has replaced it. |
| 05:12.46 | Samot | None of the current sample configs have a reference to it |
| 05:12.56 | Samot | Just like nat=yes is no longer in the sample config. |
| 05:13.17 | drmessano | I believe both of those settings print log warnings on startup/reload |
| 05:13.24 | drmessano | Descriptive ones |
| 05:13.36 | drmessano | <Insert Kermit Meme> |
| 05:13.58 | Samot | Well, there was a guy in here the other week... |
| 05:14.18 | Samot | Trying to use the sample configs to troubleshoot settings in Asterisk 1.8 |
| 05:14.26 | drmessano | ha |
| 05:14.51 | Samot | I forgot that in that version you couldn't set multiple settings separated by a comma |
| 05:15.09 | snadge | I got that particular problem sussed, asterisk allows multiple values for localnet |
| 05:15.10 | Samot | I can't remember the exact setting but it still required a line for each. |
| 05:15.17 | Samot | Yes, it does |
| 05:15.27 | snadge | so you can define as many networks as you need to, to exclude from the nat setting |
| 05:15.33 | Samot | Yes. |
| 05:15.45 | Samot | localnet is important when dealing with NAT |
| 05:15.50 | snadge | I think that will fix the problem but have to wait until after hours to teat |
| 05:16.19 | drmessano | Also, now is a good time to use the current options for the nat= setting |
| 05:16.22 | snadge | err test.. Theres a fortigate in the mix which complicates things |
| 05:16.32 | snadge | its asterisk 1.8 |
| 05:16.46 | snadge | so no need for current settings ;) |
| 05:17.10 | snadge | I'm aware of comedia and rport |
| 05:18.03 | drmessano | But not the localnet setting? |
| 05:18.08 | drmessano | Thats Bizarre |
| 05:18.41 | snadge | the complication was the software I'm using only allows one entry for localnet |
| 05:18.50 | drmessano | Which software? |
| 05:19.01 | snadge | but a google death revealed there's an extra sip settings option |
| 05:19.06 | Samot | You understand that neither of those change the IP where Asterisk sends RTP, right? |
| 05:19.10 | drmessano | Wait |
| 05:19.14 | snadge | death, search.. Using a phone |
| 05:19.15 | drmessano | "Extra SIP Settings" |
| 05:19.18 | drmessano | You mean FreePBX? |
| 05:19.25 | snadge | Its pbxware, bicom |
| 05:19.29 | drmessano | Okay |
| 05:19.44 | drmessano | I guess they use the same term |
| 05:19.48 | Samot | So in other words, "The GUI didn't have a spot for it" |
| 05:19.56 | drmessano | Which = everything |
| 05:20.05 | snadge | In extra sip settings you can simply add localnet=blah |
| 05:20.10 | Samot | Right |
| 05:20.11 | snadge | yeah |
| 05:20.21 | Samot | Because its letting you add sip settings to the sip.conf |
| 05:20.23 | Samot | Like anyone would. |
| 05:20.32 | drmessano | Bizarre |
| 05:20.44 | snadge | I wish it would allow you to rip off the config files from the gui |
| 05:21.00 | Samot | You can't SSH into it? |
| 05:21.03 | snadge | its a 3rd party PBX and we want to host it ourselves :P |
| 05:21.08 | snadge | Unfortunately not |
| 05:21.24 | snadge | They won't give us the configs but we have web access |
| 05:21.59 | drmessano | Are the SIP ports exposed to the public internet? |
| 05:22.25 | snadge | I believe so |
| 05:22.32 | drmessano | With 1.8 on it? |
| 05:22.37 | drmessano | Do me a favor |
| 05:22.44 | drmessano | Send me the info for your HR person |
| 05:22.53 | snadge | Their might be an SBC in front of it |
| 05:22.59 | snadge | Err there |
| 05:23.00 | drmessano | I wanna apply for the position of the InfoSec guy that needs to be fired for allowing that |
| 05:23.30 | snadge | If we can get root access via 1.8 that would be useful ;) |
| 05:23.43 | drmessano | 1.8 shouldn't even be used |
| 05:24.35 | drmessano | 1.8 went security-fix in 2014 and EOL in 2015 |
| 05:24.39 | drmessano | It's beyond dead |
| 05:27.09 | snadge | This ones unfortunately a third party.. I'm only just starting to learn about pbxware |
| 05:27.19 | snadge | It uses gentoo.. lol |
| 05:27.39 | drmessano | Sure but isn't your company using it? |
| 05:32.01 | snadge | Right, it's my job to try and fix this stuff, its day #3 |
| 05:32.29 | snadge | Migrating away from this PBX is challenge |
| 05:32.57 | snadge | obviously easiest option is to use the same software except on our own server, I feel thats step one |
| 05:33.23 | snadge | but with only GUI access to the configs, thats bound to end badly |
| 05:33.51 | Samot | I don't understand. |
| 05:33.54 | Samot | It's Asterisk. |
| 05:34.07 | Samot | At this point go get FreePBX, make a VM, installed FreePBX. |
| 05:34.16 | Samot | You know your extensions |
| 05:34.22 | Samot | You know your IVRs and call flow |
| 05:34.29 | snadge | OK sure but apparently theres crap everywhere |
| 05:34.39 | Samot | What does that mean? |
| 05:34.42 | snadge | bound to miss stuff and get it wrong |
| 05:34.57 | Samot | You can't export from that system to another system. |
| 05:34.58 | snadge | Heaps of options to drill down into |
| 05:35.07 | drmessano | wait |
| 05:35.10 | drmessano | 00:34:43 <snadge> bound to miss stuff and get it wrong |
| 05:35.13 | drmessano | ^ Uhm |
| 05:35.23 | drmessano | Isnt that part of doing ones job? |
| 05:35.38 | drmessano | Figuring it out.. |
| 05:35.52 | Samot | You can't take an PBXWare system, export the configs and then upload that into a FreePBX system. |
| 05:35.54 | drmessano | Looking at the "Stuff everywhere" and documenting it |
| 05:35.59 | Samot | It's just not going to work. |
| 05:35.59 | snadge | sure :) this PBX is one part of it |
| 05:36.01 | drmessano | So you DONT get it all wrong |
| 05:36.17 | drmessano | So how is that a viable excuse for anything |
| 05:36.29 | snadge | Freepbx can approximate their current setup sure |
| 05:36.33 | drmessano | "Well, its going to be hard and we may get stuff wrong, so lets not do it" |
| 05:36.41 | snadge | The transition is going to be bumpy |
| 05:36.45 | Samot | No kidding. |
| 05:36.47 | snadge | exactly |
| 05:36.49 | drmessano | No kidding |
| 05:36.49 | Samot | That's the breaks. |
| 05:36.57 | drmessano | Part of being a tech |
| 05:37.01 | Samot | That's what you are there for. |
| 05:37.02 | drmessano | Doing actual work |
| 05:37.08 | snadge | Yeah I wanted to try and two part it |
| 05:37.14 | Samot | To realize it is going to be bumpy and how to make it as smooth as possible. |
| 05:37.28 | snadge | Ie some how rip off the exact setup but be in control of it 100% |
| 05:37.37 | Samot | You CAN'T |
| 05:37.40 | snadge | then we have a solid reference to work with |
| 05:37.42 | Samot | For so many reasons. |
| 05:37.48 | drmessano | If you migrate it to another of the same box, you wont replace it |
| 05:37.52 | Samot | 1) It's Asterisk 1.8 |
| 05:38.00 | drmessano | I can already tell from this convo |
| 05:38.03 | Samot | 2) It's a closed system using Asterisk... |
| 05:38.09 | drmessano | Because once you solve one problem you'll drop it |
| 05:38.11 | Samot | So it's not going to be setup the same way any place else. |
| 05:38.22 | drmessano | Grow a pair and get a new PBX in place |
| 05:38.29 | drmessano | Document, plan, migrate |
| 05:38.30 | drmessano | JFC |
| 05:38.38 | Samot | You're going to actually have to do this the hard way. |
| 05:38.52 | snadge | I'm allergic to effort |
| 05:38.55 | Samot | Because whoever was before you there decided to do it the lazy way. |
| 05:38.58 | Samot | Dear god. |
| 05:39.05 | drmessano | snadge: It's pretty obvious |
| 05:39.18 | Samot | Why did I even bother to try and help? |
| 05:39.21 | snadge | and I hate VoIP systems.. I just want to phone it in ;) |
| 05:39.29 | Samot | Then quit. |
| 05:39.34 | drmessano | goes off to do real work |
| 05:39.40 | Samot | follows |
| 05:39.43 | snadge | because I need the money |
| 05:44.48 | snadge | thanks for the advice anyway |
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| 07:47.32 | mubbashar84 | i need to detect Voicemails through asterisk. AMD is creating delays while detecting so due to delays bad customers experience is observed i am in search of better option other than AMD or BackgroundDetect. |
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| 08:56.58 | jamesaxl | Do I have something wrong here https://bpaste.net/show/f2eabd1a118c? cause I received res_odbc.c:448 ast_odbc_print_errors: SQL Execute returned an error: HY000: ERROR: more than one row returned by a subquery used as an expression; |
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| 11:46.25 | jamesaxl | Do I have something wrong here https://bpaste.net/show/f2eabd1a118c? cause I received res_odbc.c:448 ast_odbc_print_errors: SQL Execute returned an error: HY000: ERROR: more than one row returned by a subquery used as an expression; |
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| 13:41.12 | Maliuta_ | what is in the SQL? |
| 13:41.27 | Maliuta_ | you need to look there |
| 13:42.38 | Samot | Where is the DB query in that dialplan? |
| 13:43.58 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:44.46 | Samot | And HY000 means some sort of general error or an error with no related error code happened. |
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| 14:27.18 | jamesaxl | Samot, I have only same => n,Set(NAME=${ODBC_FULLNAME(${EXTEN})}) , this table is seppeers name is A unique COLUMN. |
| 14:27.39 | jamesaxl | unique FIELD* |
| 14:30.02 | Samot | OK that didn't exist in the dialplan you shared. |
| 14:33.12 | Samot | I don't use the ODBC stuff a lot for things in dialplan but I don't recall a function called ODBC_FULLNAME() |
| 14:34.20 | Gugge | ODBC_XXX is user defined functions |
| 14:35.01 | Samot | Yes. |
| 14:35.10 | Samot | Again, I said I don't recall a FULLNAME |
| 14:35.12 | Samot | There is FETCH |
| 14:35.22 | Samot | Rollback, Commit |
| 14:35.28 | Samot | I get there are ODBC_XXX commands. |
| 14:35.34 | Samot | I was referring to a specific one. |
| 14:36.34 | Samot | The fact there is a general HY000 error being return could go along with sending an unknown command via ODBC |
| 14:56.03 | jamesaxl | Samot, thank you for this information. |
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| 16:07.42 | GeneralSpongebob | Hi, when I do "sip show peers" I can see an extension "4038/door" but I don't know where "door" is coming from. Where is that name in sip.conf? |
| 16:08.01 | scgm11_ | is there anywhere were I can find astricon 2017 videos? or they wont be published? |
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| 16:10.39 | [sr] | howdy |
| 16:10.57 | [sr] | for the permit= option for sip extensions |
| 16:11.00 | [sr] | what is best to use |
| 16:11.10 | [sr] | 192.168.1.44/32 or 192.168.1.44/255.255.255.0 |
| 16:11.12 | [sr] | ? |
| 16:15.50 | GeneralSpongebob | The first allows only that IP while the second allows the whole range. It depends on what you want |
| 16:23.28 | [TK]D-Fender | you don't even need to specify /32 for that case |
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| 17:38.13 | [sr] | GeneralSpongebob: ok you're right, the example should be 192.168.1.44/32 or 192.168.50.44/255.255.255.0 |
| 17:38.28 | [sr] | [TK]D-Fender: ok great, for single IP no need for netmask |
| 17:38.29 | [sr] | thanks |
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| 17:49.45 | avb | jamesaxl: your error is sdql related and has nothing todo with asterisk. seems you are using subquery in your main queury which you need to limit to a 1 row result |
| 17:50.17 | avb | sql* |
| 17:51.13 | jamesaxl | avb: that is what I did I limit to 1, But i have sippeers.name unique |
| 17:51.27 | jamesaxl | no problem, I do not have now Any error |
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| 18:58.24 | GeneralSpongebob | Has anyone here ever got a Paxton Mk1 door entry panel working with Asterisk? |
| 19:00.16 | GeneralSpongebob | Server and panel are on the same subnet. The panel registers with Asterisk but goes to unreachable straight away and I can't dial it and it can't dial me. Same thing with qualify=yes and no |
| 19:02.19 | [TK]D-Fender | set a defaultip and stop qualify-ing on it |
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| 19:05.51 | GeneralSpongebob | The panel always tries to register and shows an error if not. |
| 19:07.08 | [TK]D-Fender | If you want an opinion we should be looking at debug |
| 19:07.10 | [TK]D-Fender | and configs |
| 19:10.11 | GeneralSpongebob | the panel log is only that it tried to connect and whether it worked or not. As far as Asterisk goes, " -- Registered SIP '4038' at 10.0.17.61:5060" and that's it at verbosity 5. If I have qualify on it then tells me it's unreachable after it's registered |
| 19:12.13 | GeneralSpongebob | tcpdump shows REGISTER from the panel and the server replying 401 then 200 but that's it |
| 19:18.00 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
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| 19:40.39 | SupaYoshi | Looking for (paid) support, setting up a custom_extensions.conf for e164 formatting. for callerid forwarding, anyone here up to call/help/teamviewer/discord/skype/voice? |
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| 20:59.17 | znoteer_ | [TK]D-Fender: yesterday you asked me to show packets moving through my firewall towards my * server and back during a call attempt (incoming calls from my voip provider no longer reach asterisk) I sent a paste. I was wondering if you've had time to look at it yet |
| 21:02.05 | znoteer_ | it would seem to show that udp traffic on port 4569 is getting properly forwarded through the firewall to my * server. But, incoming calls do not get to me. (iax2 show registry reports the perceived address as my provider, which is wrong) |
| 21:03.46 | znoteer_ | here's the paste again: https://paste.debian.net/hidden/dc3f85e0/ 192.168.ccc.5 is the local ip of my * box |
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