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| 00:18.49 | *** 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 |
| 00:19.33 | adonaros | akay: |
| 00:19.50 | adonaros | i am currently cleaning/fixing exactly this problem |
| 00:20.04 | adonaros | what software are you going to use? are you thinking freepbx? |
| 00:20.43 | adonaros | great, here is something to consider. asterisk is "mostly secure" - keep it up to date, which may be a pain in freepbx, to walk from asterisk core 13.0.0 to 13.20.0 for example |
| 00:20.57 | adonaros | but its not asterisk that has bugs and "holes" -- its freepbx webui |
| 00:21.41 | adonaros | you would need to enable some kind of 2fa to get into freepbx webui like "vitel-shield" (though i hate vitelity too) |
| 00:22.00 | adonaros | or you would need to block off web access to only you or who your clients are et al |
| 00:23.04 | adonaros | some kind of firewall/fail2ban that only allows webif access if there is an existing sip registration, let's say. This might be implemented in freepbx, and i have not RTFM to figure it out, but i dont have this problem. because i run vanilla asterisk cos i got tired of getting burned by freepbx. |
| 00:24.55 | adonaros | if you do not have shitty endpoints, and your endpoints only take sip commands/access from your proxy and your firewall acutally works (which it may not) you do not need to fear murderous calls except for ones made by your clients. |
| 00:31.49 | adonaros | pm me more for other questions et al, as you will probably get lost in the buffer and im leaving the office. |
| 00:33.29 | akay | great input! i'll /q you |
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| 07:35.56 | Blinky_ | Morning all, need some help please. Our phone server has decided not to make any outbound calls, worked find yesterday. It will make inbound calls though. When I try to make a call from a user to my mobile this is the output with the debug on - https://pastebin.com/QtiVr7ss - Could someone please help, I do not have a huge knowledge on debugging asterisk and need this line up fast. |
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| 10:37.39 | UNNNE7A7PT | hello, is there any way to have _date:[pP]:. => digits/p-m for multiple day separation (instead of two) Ex: morning, noon, afternoon, evening, night? |
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| 15:24.03 | clebig | good day here |
| 15:24.51 | clebig | Can someone tell me if using trunkfreq on a iax.conf actually works ? |
| 15:25.26 | clebig | because when I tcpdump on the iax traffic, even when I change this to 500 or 1000, I still see packets every 20ms |
| 15:27.39 | clebig | My version is 13.20, I have checked the code of chan_iax2.c here : https://github.com/asterisk/asterisk/blob/13.20/channels/chan_iax2.c , line 13721 there is a condition that if trunkfreq is over 1000, a NOTICE message must appear in the log saying trunkfreq must be between 10ms and 1000ms, using 1000ms instead |
| 15:28.05 | clebig | But when I set a trunkfreq superior to 1000, I never get this notice message |
| 15:28.56 | [TK]D-Fender | Sounds like an oversight in that it should warn you you're out of range |
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| 15:30.31 | clebig | yes, that's why I'm wondering if someone could confirm this trunkfreq parameter is really taken into account in this version |
| 15:30.55 | igcewieling | clebig: very few people still use IAX |
| 15:31.03 | [TK]D-Fender | I'd confirm that you are indeed trunking properly with your peer |
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| 15:31.58 | clebig | igcewieling: ok. |
| 15:32.17 | clebig | [TK]D-Fender: I think so, I'm controlling both sides |
| 15:32.28 | [TK]D-Fender | Time to verify |
| 15:33.08 | clebig | also, I have read about a bug affecting jitterbuffer* settings when using IAX in trunk mode, has this been fixed already ? |
| 15:33.18 | clebig | [TK]D-Fender: ok |
| 15:34.57 | clebig | on one side I have : https://pastebin.com/raw/mERyKktp |
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| 15:36.18 | [TK]D-Fender | clebig, in the sample config I see trunkfreq as a [general] option, not under a peer itself. |
| 15:36.24 | [TK]D-Fender | clebig, That could be the issue |
| 15:36.33 | clebig | ah ! |
| 15:36.48 | clebig | on the other host, it is in general |
| 15:36.51 | clebig | I'm gonna test |
| 15:38.02 | clebig | but that would be strange, because we might wan't to set it differently according to our remote trunks properties (connectivity) |
| 15:38.05 | clebig | ? |
| 15:39.18 | clebig | nope, still seeing IAX traffic every 20ms |
| 15:39.30 | [TK]D-Fender | I'd like to imagine that being independently settable but perhaps there is an impact in how it times multiple trunks, etc |
| 15:40.46 | [TK]D-Fender | Check your porition for related parameters as well |
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| 15:43.31 | clebig | ok, but I don't see any other parameters where timing is involved in iax.conf except those for jitters which are commented, and those codec ptime |
| 15:48.57 | jkroon | clebig, yes it works. |
| 15:49.16 | jkroon | you do need to have multiple concurrent calls between the same hosts to actually see the effect though. |
| 15:50.00 | clebig | jkroon: you mean that with only one call, it is not taken into account ? I mean datagrams are still passed to the interface every 20ms ? |
| 15:50.02 | igcewieling | I think the break even point is 3 calls, any fewer and trunking would have more overhead than it saves. |
| 15:50.48 | jkroon | clebig, it'll only transmit at those intervals if there are actually frames to be transmitted, otherwise it's wasting bandwidth, so if you set it to 10ms, with only a single call, ptime=20ms, then why send empty stuff every other packet? |
| 15:50.48 | clebig | This trunk is going to support 8 simultaneous call most of the time |
| 15:51.33 | jkroon | clebig, then it'll work correctly. i believe the last timing related patch wrt trunkfreq has been patched. i've never seen the JB issue, but you should still set the option to trunk timestamps |
| 15:51.44 | clebig | ok jkroon, that is smart and that would explain why I see what I see, I'm doing a tests with only one call |
| 15:52.08 | jkroon | trunkfreq=10; trunktimestamps=yes; trunkmtu=1240 |
| 15:52.22 | jkroon | that's typically the three settings that I have in [general] |
| 15:52.23 | clebig | trunkmtu is not set, but the rest is |
| 15:52.39 | clebig | trunkmtu=1240, isn't it the default ? |
| 15:52.44 | clebig | when not set |
| 15:52.45 | clebig | ? |
| 15:53.24 | jkroon | i have no idea ... it wasn't when i discovered it probably, or i decided i wanted to be explicit (I often do that though). |
| 15:53.36 | jkroon | i don't like defaults I care about changing without me knowing about it. |
| 15:54.02 | clebig | yes it is, it is written as comment in the iax.conf |
| 15:54.57 | jkroon | the code agrees. |
| 15:55.20 | jkroon | trunkmtu can be set per peer. |
| 15:55.57 | clebig | jkroon: do you know if jitterbuffer and related jitter settings work in iax trunk mode ? |
| 15:56.20 | jkroon | why should they? it's jitter in, jitter out, let the endpoint deal with it. |
| 15:57.48 | jkroon | having said that, i remember there were some rather bad devices out there that was receiving RTP that needed to have LOW jitter to cope, and in those cases I did set the forcejb settings, and from what I recall it used to work. haven't specificaly timing-tested it recently as ... well, it just works. |
| 15:57.54 | clebig | if the paths in between the trunk peers is unstable, delays some dgram, it's worth having on on both side, or am I understing it bad ? |
| 15:58.22 | jkroon | clebig, if possible i'd say let the endpoint (phone) device deal with it if possible. |
| 15:58.51 | clebig | ok |
| 15:58.53 | jkroon | I note this is still in my configs: |
| 15:58.54 | jkroon | jitterbuffer=yes |
| 15:58.54 | jkroon | forcejitterbuffer=yes |
| 15:59.33 | jkroon | haven't hunted down any rtp timing related bugs in at least 5 years so i'm guessing it's either still working or no longer needed. |
| 15:59.59 | clebig | https://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer |
| 16:00.07 | jkroon | i do recall something about jitterbuffers should actually be used on TX to outputs (DAHDI) that can't accept jitter. |
| 16:00.08 | clebig | it is indicated : Trunking distorts frame timestamps by up to "trunkfreq" milliseconds (usually 20msec). Accurate timestamps are critical to the operation of the current jitterbuffer - so trunking and dejittering are currently incompatible. |
| 16:00.41 | clebig | because I had quality problem when jitterbuffers were set |
| 16:00.46 | clebig | with normal values |
| 16:01.00 | jkroon | hmm, then i wonder if the trunkfreq patch ever made it to mainline. |
| 16:01.04 | clebig | like cuts 2 times per second |
| 16:01.32 | clebig | https://issues.asterisk.org/jira/browse/ASTERISK-26766 |
| 16:02.20 | clebig | it is said : Resolved. With jitterbuffer and trunking enabled in IAX it is necessary to include trunktimestamps=yes in iax.conf. |
| 16:02.21 | jkroon | https://issues.asterisk.org/jira/browse/ASTERISK-19521 test the patch there. |
| 16:02.36 | clebig | but I could notice this problem even with trunktimestamps=yes |
| 16:02.44 | jkroon | try that patch :) |
| 16:03.08 | jkroon | anyway, i've got to be off sorry. |
| 16:03.12 | clebig | I have checked into the sources |
| 16:03.14 | clebig | on github |
| 16:03.18 | clebig | and it has been included already |
| 16:03.30 | clebig | ok jkroon, thank you for your time ! |
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| 17:00.22 | igcewieling | Has anyone seen issues with the chan_sip keepalive= setting with using tcp transport? I get warnings in the CLI. |
| 17:01.20 | igcewieling | "[2018-04-10 12:58:01] WARNING[13123] chan_sip.c: sip_send_keepalive to 10.0.0.28:5060 returned 0: Interrupted system call" and "[2018-04-10 12:58:21] WARNING[13123] chan_sip.c: sip_send_keepalive to 10.0.0.28:5060 returned 0: No such file or directory" |
| 17:01.52 | igcewieling | I switched back to UDP, but was wondering if anyone else saw anything similar. |
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| 19:08.33 | LunaLovegood | I have multiple SIP phones in my home and they all ring simultaneously when someone calls my DID. I can pick up from any of them, but then the only phone in that conversation will be that one. Is there a way I could set it up so someone could pick up another phone (like say the one in the kitchen), push a button, and join the conversation? |
| 19:08.48 | LunaLovegood | My router runs Asterisk 14, but I can upgrade to 15 if needed. |
| 19:10.02 | [TK]D-Fender | Use a dial option that sends both parties to a conference room |
| 19:10.11 | [TK]D-Fender | Yourother phones could then dial that conf room to jion |
| 19:11.01 | LunaLovegood | What happens when a second calls comes in? In that case I want the other phones to ring. I don't want it to automatically join the first call. |
| 19:11.19 | [TK]D-Fender | it's your dialplan. |
| 19:11.28 | [TK]D-Fender | You'll have to do whatever checks you want to implement |
| 19:11.34 | LunaLovegood | ok, I guess I'll read the docs |
| 19:11.59 | [TK]D-Fender | There's no doc to tell you how to build your exact thing |
| 19:12.31 | [TK]D-Fender | You can get instructions on how to connect 2 LEGO bricks together, but noby is making an instruction sheet for the specific thing in your head |
| 19:12.42 | LunaLovegood | but I can build out of the existing applications, right? |
| 19:12.51 | [TK]D-Fender | You dop checks |
| 19:13.01 | LunaLovegood | dop? |
| 19:13.04 | [TK]D-Fender | you use dial when and how you want to based on circumstances. |
| 19:13.07 | [TK]D-Fender | do* |
| 19:13.11 | LunaLovegood | ah |
| 19:13.18 | LunaLovegood | I should be fine then, thanks |
| 19:13.18 | [TK]D-Fender | dialplan = programming |
| 19:13.47 | LunaLovegood | Is there a stack limit I should be aware as far as macros and gosubs go? |
| 19:13.54 | LunaLovegood | *aware of |
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| 19:14.03 | [TK]D-Fender | <LunaLovegood> What happens when a second calls comes in? In that case I want the other phones to ring. I don't want it to automatically join the first call. <- you'll have to count if there is somebody i a conf room already and decide how to act instead |
| 19:14.22 | [TK]D-Fender | Gosub doesn't have the old 7-deep macro limit |
| 19:14.35 | [TK]D-Fender | but I'd have to wonder why your setup would even get that far |
| 19:15.07 | LunaLovegood | There's already a bunch of stuff in my dialplan. I should probly clean it up a bit before adding more stuff. |
| 19:15.32 | [TK]D-Fender | up to you |
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| 20:08.30 | taschendiebLAUCH | hey there short questiuon about jumping in my dialplan |
| 20:08.42 | taschendiebLAUCH | when i have a extension like that: https://bpaste.net/show/8937fd149cec |
| 20:09.08 | taschendiebLAUCH | and i am in another extension with my call that cam in |
| 20:09.29 | taschendiebLAUCH | how could i jump to that (normalrun) label? |
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| 20:17.28 | [TK]D-Fender | You have to jump to that extension & label |
| 20:18.03 | [TK]D-Fender | so Goto(YOURCONTEXT,+49234501,normalrun) |
| 20:18.11 | [TK]D-Fender | however the CID would have to also match |
| 20:18.34 | [TK]D-Fender | which means maybe you should already put part of that code in another extension and GOTO/GOSUB to it from your initial match |
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| 21:21.37 | qakhan | hi all, there is ABANDON event in queue_log table. but some records show call went to all agents (RINGNOANSWER) and some ABANDON event records show call went to some agents (RINGNOANSWER) |
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