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| 01:20.19 | igcewieling | heh, "He was a great man, but only when seen from a distance." |
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| 02:18.15 | aandrew | < InFeDeL> i keep getting illegal instructions |
| 02:18.18 | aandrew | < pjensen00> Lawyer up |
| 02:18.24 | aandrew | hahaha, best response to that I've ever seen |
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| 02:30.44 | sarsaeol | hi there i'm trying to figure out what a predecessor has done and having a bit of trouble figuring out what they've done here. they are using func_odbc to grab some DB values. this is an example of one, they exec an app called ivr_setLocal |
| 02:30.58 | sarsaeol | the trouble is, i have no idea what ivr_setLocal actually does, and I cannot find it |
| 02:31.44 | igcewieling | show us the line, if more than one or two lines, use a pastebin |
| 02:31.50 | sarsaeol | http://pastebin.com/64cDNqwh |
| 02:32.50 | sarsaeol | so i'm assuming ivr_setLocal is a known context or routine to asterisk somehow, but i just cannot find it. |
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| 02:34.02 | sarsaeol | i kinda assumed it would be included in extensions.conf or something, but it doesn't appear to be anywhere on the server |
| 02:34.02 | igcewieling | I have no idea what thats. here is a sample from one of my configs: writesql=UPDATE `accounting` SET `answer_timestamp`=NOW() WHERE `sid`='${VAL1}' |
| 02:34.48 | sarsaeol | RIGHT! i totally expected actual SQL statements in the readsql.. it seems they are running a script which would contain the sql, but holy hell it's hidden |
| 02:34.50 | igcewieling | everything after the = should be be standard SQL |
| 02:35.00 | sarsaeol | that was my understanding as well |
| 02:35.04 | sarsaeol | but this apparently works. |
| 02:35.10 | igcewieling | use grep 8-) |
| 02:35.34 | sarsaeol | i did try, but not against the whole filesystem⦠i guess ill just let it run overnight |
| 02:35.43 | sarsaeol | thanks for looking igcewieling =] |
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| 08:35.38 | WIMPy | What is sys/vmmeter.h and why does Asterisk want it? Never complained before. |
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| 08:43.08 | WIMPy | Hmm. snmp? But that's disabled. |
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| 08:55.06 | samwierema | Morning. Does the ARI websocket send a (last) close frame when restarting/stopping Asterisk? |
| 08:55.08 | samwierema | There does seem to be an implementation for it: https://github.com/asterisk/asterisk/blob/master/res/res_http_websocket.c#L291 |
| 08:56.17 | WIMPy | Hmm. Aparently a change in 11.17. I had 11.19 runnong, though. Does it suddenly think I'm on BSD? |
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| 09:39.26 | GeneralSpongebob | Hi, I'd like to have an easy toggle to block all new incomming calls with a busy signal but I'm not sure where to start. Can anyone provide any tips? |
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| 09:41.44 | WIMPy | Well, you have to do it on all extensions. Or send all calls to some catch-all and then distribute them from there. |
| 09:45.04 | GeneralSpongebob | Can this be implemented as an override so I don't have to remove all my existing extensions? |
| 09:49.48 | GeneralSpongebob | Also, I've been thinking of doing the Asterisk Essentials course. Has anyone here done it and found it useful? |
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| 15:40.53 | EnrgySmth | Sorry if this is a dumb question, I am a newbie and reading up dialplans, trying to understand... Is the dialplan syncronous/sequential in nature? If you have a dialplan that calls the Record( ) application for 15 seconds, and then Playback(hello-world), will the 15 second record complete before Playback is executed? |
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| 15:42.48 | [TK]D-Fender | yes |
| 15:43.00 | [TK]D-Fender | almost all dialplan applications are "blocking" |
| 15:43.47 | EnrgySmth | Ok, excellent, thank you |
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| 16:25.13 | *** join/#asterisk yorick (~yorick@oftn/oswg-member/yorick) |
| 16:28.16 | yorick | hello! I'd like to --with-pjproject-bundled, but my build process is unable to do network IO for... reasons. is there a way to put the thing it wants to download in place beforehand? or should I just use unbundled pjproject? |
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| 16:30.20 | yorick | I realize this is a weird problem to have |
| 16:30.30 | Samot | I think with that it builds it. |
| 16:30.47 | Samot | You shouldn't have to comple pjproject separately. |
| 16:30.58 | Samot | It will compile at the time with Asterisk. |
| 16:31.04 | Samot | With that flag I believe. |
| 16:31.18 | yorick | Samot: yeah, but it tries to download it from the internet and the build server has no internet |
| 16:31.40 | Samot | I haven't done it with that flag yet. |
| 16:31.54 | Samot | I have compiled them separately when I've done it. |
| 16:32.13 | Samot | So it's possible that requires a connection to pull the pjproject stuff. |
| 16:33.49 | Samot | What version of Asterisk are you trying to install? |
| 16:35.16 | Samot | "The configure and make processes will download the correct version of pjproject, patch it, configure it, build it, and finally link Asterisk to it statically. No changes in runtime configuration are required." <-- straight from the Wiki page |
| 16:36.02 | Samot | yorick: You can download pjproject 2.4 or higher and compile it on its own |
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| 17:02.34 | rmudgett | yorick: You can cache the pjproject tarball and use the --with-externals-cache=/home/rmudgett/cache_pjproject |
| 17:02.34 | rmudgett | <PROTECTED> |
| 17:03.06 | yorick | rmudgett: okay, that helps! thanks! |
| 17:03.33 | yorick | rmudgett: is that documented anywhere? |
| 17:04.22 | rmudgett | ./configure --help lists it. |
| 17:04.35 | Samot | We all learned today. |
| 17:04.41 | Samot | Cuz I didn't know that. |
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| 17:04.52 | yorick | rmudgett: yeah, but that talks about the binary modules, not that tarbalal |
| 17:04.56 | yorick | tarball* |
| 17:06.23 | rmudgett | On another system you can ./configure to get the tarball. Then on the internet restricted machine you place the downloaded file into your cache directory. |
| 17:06.38 | yorick | ah yeah, that'll work # DOWNLOAD_DIR := $(or $(EXTERNALS_CACHE_DIR),$(TMPDIR),$(wildcard /tmp),.) |
| 17:08.09 | yorick | rmudgett: thanks a lot! |
| 17:09.24 | rmudgett | IIRC that externals cache directory is also used for caching the external modules like codec_opus, codec_silk, codec_siren7/14, codec_g729a, and res_digium_phone |
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| 17:11.34 | igcewieling | rmudgett: not things like sounds? |
| 17:12.43 | rmudgett | Sounds are cached using a different parameter: --with-sounds-cache=PATH |
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| 17:13.25 | yorick | sounds are in the release tarball, I thought |
| 17:13.52 | igcewieling | yorick: not since...2002? The install downloads them automatically. |
| 17:14.21 | yorick | someone put me in charge of fixing the nixos asterisk package |
| 17:14.44 | yorick | oh, they patch it to remove that download |
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| 19:24.43 | wasanzy | hi |
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| 19:26.32 | wasanzy | Is it possible to get if elsif else in dialplan? |
| 19:28.39 | [TK]D-Fender | huh? |
| 19:29.04 | scgm11_ | I would use GotoIf nested |
| 19:29.50 | [TK]D-Fender | No, there is no "elseif" |
| 19:31.00 | wasanzy | scgm11_: I will aprreciate if you can show me the nested |
| 19:31.26 | wasanzy | I want to be able to have more than two conditions |
| 19:32.18 | [TK]D-Fender | there is no nesting |
| 19:32.24 | [TK]D-Fender | just jump accordingly |
| 19:32.36 | [TK]D-Fender | Do your Gotoif. Then do the next. |
| 19:32.39 | [TK]D-Fender | then the next |
| 19:32.42 | [TK]D-Fender | Thre is no "sample |
| 19:34.16 | wasanzy | ok I get it |
| 19:39.41 | scgm11_ | you can use this to design and maintain dialplan https://github.com/IntegraCCS/integradesigner |
| 19:59.30 | DanQuinney | hi all, we're seeing the following warning "No UDPTL ports remaining" within one of Asterisk servers - a quick google suggests that Asterisk isn't releasing the ports correctly for none T.38 calls |
| 20:00.04 | DanQuinney | not sure where to begin to investigate/fix - anyone got any pointers? |
| 20:00.28 | igcewieling | DanQuinney: first, update to the latest version of the Asterisk branch you are using. |
| 20:01.16 | DanQuinney | We have, and before anyone wants to kill me, we're running v1.8 ... (plans are afoot to upgrade to the latest v13 by Q2 of this year |
| 20:01.34 | igcewieling | ah, then you are on your own. |
| 20:01.57 | DanQuinney | don't say that igcewieling :p |
| 20:02.19 | igcewieling | Personally I prefer to go from 1.8 to Asterisk 11, since there are not all that many changes which would affect config files. |
| 20:03.11 | DanQuinney | I'm not too scared of the upgrade to 13 - it's just finding time to do it and the required testing |
| 20:03.43 | igcewieling | I'm far too scared to upgrade to v13 |
| 20:05.37 | DanQuinney | got to take the bull by the horns |
| 20:07.02 | igcewieling | DanQuinney: I prefer to run mostly the same version on all servers. I'm not upgrading 70+ Asterisk boxes just to have the latest. |
| 20:08.28 | igcewieling | I also don't feel like rewriting the scripts we use to handle whatever changes Asterisk 13 would require. |
| 20:12.50 | Samot | Are you going to eventually? |
| 20:21.01 | igcewieling | Samot: I expect to upgrade to Asterisk 15 about 6 months after v15 is released. |
| 20:22.03 | igcewieling | well, start upgrading at least, it will take a while to get all servers upgraded. |
| 20:28.07 | Samot | Got a 14 box running? |
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| 20:48.42 | blscoe | Hello! |
| 20:49.15 | blscoe | Is there a way to save asterisk config (for compilation) like the kernel does in .config? |
| 20:53.44 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 20:54.36 | madduck | can I have a sanity check, please? client behind NAT firewall (linux, running the conntrack_sip modules) talks to asterisk out there using SIP just fine. server receives RTP and sends plenty back |
| 20:54.40 | madduck | e.g. |
| 20:54.43 | madduck | 7648 77.197539 188.174.253.168 â 100.64.45.19 UDP 216 Source port: 13844 Destination port: 8958 |
| 20:55.01 | madduck | but the firewall at 100.64.45.19 actually never sees incoming RTP from that IP |
| 20:58.02 | madduck | rtp debug on the asterisk server only says it's sending packets too⦠|
| 20:58.05 | madduck | [Jan 25 21:57:20] Sent RTP packet to 100.64.45.19:8322 (type 08, seq 029011, ts 103288, len 000160) |
| 20:58.15 | madduck | (different call) |
| 20:59.18 | blscoe | madduck: did you tested the conectivity without asterisk via udp? (eg: netcat) |
| 21:00.17 | blscoe | maybe a firewall in the path dropping this kind of traffic, depending type of connection, ISP, etc... |
| 21:00.22 | madduck | good idea |
| 21:00.35 | madduck | i was thinking along those lines too |
| 21:01.12 | blscoe | there's more choices, but this one should be the first to test |
| 21:02.20 | madduck | well, well, that already doesn't work |
| 21:02.27 | madduck | so I am fairly sure the ISP is at fault. |
| 21:04.17 | blscoe | madduck: do you know if there's a way to save asterisk config (for compilation) like the kernel does in .config? |
| 21:04.31 | madduck | i have no idea, sorry |
| 21:05.29 | blscoe | ok, don't worry |
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| 21:43.20 | *** join/#asterisk dasjoe (~dasjoe@americangirlscouts.org) |
| 21:45.04 | deweydb_ | Frequently when using my softphone (Bria App, Android) I get a low network quality icon, and the audio quality drops off horribly for 30-60 seconds. then it recovers. My phone is about 10 feet from my wifi access point when this happens, with a clear line of sight. The access point is a ubuiquity unifi. The AP is connected via Cat6 ethernet to my router (netgate s1000) which in turn is connected to my modem (150 mbit |
| 21:45.04 | deweydb_ | <PROTECTED> |
| 21:49.09 | *** join/#asterisk gusto (~gusto@2001:4c50:62e:443:ee34:53bd:4cbf:22a1) |
| 21:52.31 | igcewieling | does it work if you hardwire it instead of using WiFi |
| 21:53.01 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 21:55.49 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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| 22:01.00 | *** join/#asterisk Maliuta (~nobusines@unaffiliated/maliuta) |
| 22:02.55 | deweydb_ | igcewieling: i have had intermittent connection issues via Bria on my desktop as well. which is connected directly to a gigabit switch via ethernet, which is then connected to my router. |
| 22:03.25 | deweydb_ | <PROTECTED> |
| 22:38.44 | *** join/#asterisk apb1963 (~apb1963@107-146-220-94.res.bhn.net) |
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| 23:06.40 | wonderworld | has res_pjsip matured enough for use with webRTC (signalling via websockets)? i setup a test environment 2 years ago but wasn't able to make it work in a stable way. chan_sip used to work fine back then. |
| 23:10.51 | *** join/#asterisk rwb (~Thunderbi@65.183.151.239) |
| 23:27.52 | dan_j | When using voicemail realtime, is it possible to set the 'fromstring' on a per-voicemail basis ? |
| 23:32.32 | apb1963 | I've setup my android as an extension using zoiper and GV. I can get it to dial out, and to ring... but I cannot get sound. Upon investigation the logs show an immediate hangup. Any ideas? Log available, just let me know... |
| 23:33.07 | [TK]D-Fender | Show the log |
| 23:33.15 | [TK]D-Fender | Always just show the log |
| 23:33.33 | [TK]D-Fender | waiting gets you nothing except slower answers eventually after being asked for it.... at best |
| 23:36.14 | apb1963 | It's too big for pastebin |
| 23:36.29 | dan_j | first time i've heard that |
| 23:36.36 | apb1963 | which is kind of odd because it's only 700 lines |
| 23:36.47 | apb1963 | so I'm trying to figure that out |
| 23:36.51 | dan_j | split it? |
| 23:36.52 | apb1963 | be a minute plz |
| 23:37.31 | apb1963 | I take that back... 7600 lines from INVITE to end |
| 23:39.08 | apb1963 | [TK]D-Fender, ok if I send the file to you directly? |
| 23:40.50 | [TK]D-Fender | host it if you can |
| 23:41.05 | [TK]D-Fender | my attention is fractured and others should have a chance as well |
| 23:41.12 | [TK]D-Fender | but I'll skim it as I can |
| 23:44.00 | apb1963 | Found a new pastebin. https://fnpaste.com/8D3Q [TK]D-Fender |
| 23:46.29 | [TK]D-Fender | f'n paste indeed |
| 23:47.12 | [TK]D-Fender | o=Zoiper 0 1 IN IP4 127.0.0.1 |
| 23:47.15 | Samot | So, this is FreePBX |
| 23:47.19 | Samot | And there's that. |
| 23:47.24 | [TK]D-Fender | Zoiper seems to be handing LOCALHOST as the media address |
| 23:47.46 | [TK]D-Fender | that looks kinda shitty |
| 23:48.19 | [TK]D-Fender | and I don't see what was sent to it in the first palce |
| 23:48.43 | [TK]D-Fender | And you included a full reload worth of garbage in there, etc |
| 23:48.50 | [TK]D-Fender | You REALLY could have trimmed this down |
| 23:50.43 | Samot | So this is an incoming call from Google Voice and you're using Follow-Me to send the call BACK out the PSTN. |
| 23:51.30 | Samot | ?? |
| 23:52.56 | drmessano | Who the hell does that |
| 23:55.53 | *** join/#asterisk d00gster (~d00gster@unaffiliated/d00gster) |
| 23:56.31 | apb1963 | [TK]D-Fender, I don't know how to respond. The reload is included because apparently that's what it did - I did not manually reload it, so I don't know why it does that. |
| 23:57.26 | apb1963 | [TK]D-Fender, As for what was sent to it in the first place... again, I don't know how to respond. I started from the INVITE. |
| 23:57.41 | apb1963 | I'll look for more prior to that, but I don't know what I'm looking for. |
| 23:58.16 | [TK]D-Fender | P-Asserted-Identity: "6618365653" <sip:6618365653@127.0.1.1> |
| 23:58.24 | [TK]D-Fender | actually, your own INVITE going out started with that |
| 23:58.43 | [TK]D-Fender | Which means you have a failed DNS lookup for your WAN IP most likely |
| 23:58.54 | [TK]D-Fender | Which I'm betting on a shitty lOOPBACK of your DDN name |
| 23:58.58 | [TK]D-Fender | DDNS |
| 23:58.58 | [TK]D-Fender | ^^^ |
| 23:59.13 | [TK]D-Fender | because you're not resolving it externally due to a hosts entry |
| 23:59.45 | apb1963 | Yep |