| 00:04.03 | *** join/#asterisk Chotaire (chotaire@unaffiliated/chotaire) |
| 00:23.22 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:5db6:9709:acd3:b02b) |
| 00:34.08 | *** join/#asterisk zapata_ (~zapata@2a02:b18:581:10:5db6:9709:acd3:b02b) |
| 00:51.18 | *** part/#asterisk kharwell (kharwell@nat/digium/x-zpsihlqacrrbecsb) |
| 00:55.49 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 01:07.13 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 01:13.54 | *** join/#asterisk degenerate (~quassel@S0106689e199caaf4.no.shawcable.net) |
| 01:14.09 | degenerate | I've got 100s of lines like this in my logs, what does it mean: |
| 01:14.09 | degenerate | [2018-01-09 20:10:38] WARNING[24434] pbx.c: Maximum PBX stack exceeded |
| 01:18.44 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:18.44 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.5 (2017/12/22), Standard: 15.1.5 (2017/12/22); 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 |
| 01:21.31 | rmudgett | degenerate: Your dialplan is either nesting include contexts more than 128 levels or you have a circular context include arrangement. |
| 01:22.54 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 02:02.43 | *** join/#asterisk jjrh (~weechat12@ppp-199-167-117-242.storm.ca) |
| 02:18.44 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-irroillilyadudxs) |
| 02:21.01 | *** join/#asterisk TheFatherMind (~TheFather@cpe-104-34-204-52.socal.res.rr.com) |
| 02:21.04 | *** part/#asterisk TheFatherMind (~TheFather@cpe-104-34-204-52.socal.res.rr.com) |
| 02:38.33 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 02:40.26 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 02:41.18 | *** join/#asterisk kitsune (~kitsune@vulpix.org) |
| 02:42.18 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 02:51.30 | *** join/#asterisk BakaKuna (~BakaKuna@185.23.140.9) |
| 02:58.05 | *** join/#asterisk TandyUK (~admin@TandyUK/staff/James) |
| 03:09.10 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 03:17.50 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 03:26.38 | *** join/#asterisk r0ckp3arl_ (~rock@124.171.234.114) |
| 03:52.18 | *** join/#asterisk J0hnSteel (~J0hnSteel@92.55.116.178) |
| 03:54.18 | *** join/#asterisk cj (~cjac@moonunit.colliertech.org) |
| 03:54.19 | cj | moo |
| 03:55.07 | cj | I'm getting unicode characters echoed when I type things in to my asterisk console. anyone seen this before and maybe know how to fix it? |
| 03:55.19 | cj | Tim_Toady: you've always been able to help me with my unicode problems :-) |
| 03:55.42 | cj | wait, you're not TimToady, are you? |
| 04:07.35 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 04:09.49 | *** join/#asterisk davidbowlby (~textual@99-67-53-141.lightspeed.clmboh.sbcglobal.net) |
| 04:14.45 | *** join/#asterisk elcontrastador (~textual@206.78.110.4) |
| 04:25.58 | elcontrastador | greetings guys...I'm getting a sip 'Forbidden' error call outbound thru my ITSP...inbound calling works fine. https://gist.github.com/elcontrastador/d1ad6e264a0fbeec5498f39262ea59aa |
| 04:35.33 | elcontrastador | I'm using Asterisk 15.1.5, btw |
| 04:36.28 | *** part/#asterisk r0ckp3arl_ (~rock@124.171.234.114) |
| 04:45.36 | elcontrastador | I have resolved the issue. My ITSP put had some fraud detection i needed to disable. |
| 05:04.16 | *** join/#asterisk dandann00dle (uid120491@gateway/web/irccloud.com/x-mazmvwnbhljyumdx) |
| 05:08.30 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 05:11.22 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 05:38.21 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 05:39.46 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 06:19.26 | *** join/#asterisk jamesaxl (~James_Axl@109.172.62.242) |
| 06:37.50 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 06:38.52 | *** join/#asterisk r0ckp3arl_ (~rock@124.171.232.6) |
| 06:39.05 | *** part/#asterisk r0ckp3arl_ (~rock@124.171.232.6) |
| 06:39.46 | *** join/#asterisk velix_inuse (~Velix@x4e345e31.dyn.telefonica.de) |
| 07:00.37 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 07:05.21 | *** join/#asterisk jamesaxl (~James_Axl@109.172.62.242) |
| 07:19.52 | *** join/#asterisk vandyk (~vandyk@179.232.88.214) |
| 07:25.37 | *** join/#asterisk Penguin (~xwQ5kwYl6@2001:470:1f07:26d:20c:29ff:fe62:be33) |
| 07:47.03 | *** join/#asterisk bllackjack (67d86345@gateway/web/freenode/ip.103.216.99.69) |
| 07:47.25 | bllackjack | Can anyone help me? I cannot find macro-stdexten in extensions. My ultimate goal is to get notified whenever a call happens |
| 07:56.11 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 08:16.52 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:18.22 | *** join/#asterisk sekil (~sekil@cable-89-216-195-85.dynamic.sbb.rs) |
| 08:20.35 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:30.05 | *** join/#asterisk DanB (~DanB@clt-195.192.202.196.ip-anschluss.net) |
| 08:53.08 | *** join/#asterisk hehol (~Adium@ip-109-90-164-163.hsi11.unitymediagroup.de) |
| 09:02.49 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 09:20.01 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 09:23.46 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 09:26.08 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.w1.tele.crt1.net) |
| 09:30.09 | *** join/#asterisk vandyk (~vandyk@179.232.88.214) |
| 09:57.07 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.elen.ucl.ac.be) |
| 10:29.41 | *** join/#asterisk r0ckp3arl_ (~rock@203-206-244-207.dyn.iinet.net.au) |
| 10:31.16 | *** part/#asterisk r0ckp3arl_ (~rock@203-206-244-207.dyn.iinet.net.au) |
| 10:51.32 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 10:51.59 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 11:04.28 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 11:08.14 | velix | [TK]D-Fender: Thx. |
| 11:10.50 | velix | Can I record a phone call while it's running? I mean, start in the middle of the conversation? |
| 11:20.59 | *** join/#asterisk vandyk (~vandyk@201.90.171.130) |
| 11:26.45 | *** join/#asterisk nighty- (~nighty@s229123.ppp.asahi-net.or.jp) |
| 12:11.42 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.w1.tele.crt1.net) |
| 12:20.13 | *** join/#asterisk jkroon (~jkroon@41.21.200.174) |
| 12:25.26 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 12:25.28 | Haris | hello all |
| 12:27.23 | Haris | Is udp 5678 a significant port for voip/voice traffic ? |
| 12:36.12 | *** join/#asterisk gusto (~gusto@2a01:c844:102f:d20:dc2e:8aa9:e52c:1dc3) |
| 12:39.11 | Samot | Sure |
| 12:40.48 | *** join/#asterisk jkroon (~jkroon@41.21.200.174) |
| 12:45.28 | Haris | what does it do ? |
| 12:46.08 | Haris | its like traffic for this port is coming to us via default gw on route to telco 1 |
| 12:50.37 | velix | Sorry for asking in here, but I don't have an idea where to ask else. Does anyone know a service to check some hundret landline numbers for true existence? Sure, I could code it with Asterisk... but I'd love to use an existent service. |
| 12:57.17 | Samot | What do you mean "what does it do?" it's a port. |
| 12:57.41 | *** join/#asterisk lankanmon (~LKNnet@CPE64777dd7e053-CM64777dd7e050.cpe.net.cable.rogers.com) |
| 12:57.55 | Samot | velix: Not sure how Asterisk would do that properly. |
| 12:58.17 | Samot | velix: You need a number/line lookup service. |
| 12:58.21 | velix | Samot: Call a number and check the log? |
| 12:58.31 | Samot | OK |
| 12:58.32 | velix | Samot: Yeah, but 90% are for mobile only. |
| 12:58.39 | Samot | OK. |
| 12:58.47 | Samot | When you say "true existence" you mean? |
| 12:58.47 | velix | Samot: it's called a "ping" call. Like ring 1 sec and cancel the call. |
| 12:58.53 | Samot | It's a valid number format? |
| 12:58.59 | Samot | It's been assigned? |
| 12:59.03 | velix | Samot: Nah, that's easy. I mean, assigned. Yes. |
| 12:59.11 | Samot | Assigned does not mean routed. |
| 12:59.40 | Samot | Your ping call is going to what? Tell you something like "Number not in service?" |
| 12:59.44 | velix | Samot: Can't Asterisk show the debug info from the SIP trunk? |
| 12:59.46 | velix | Samot: yes. |
| 12:59.59 | Samot | Also, Asterisk doesn't handle SIP codes individually |
| 13:00.23 | velix | oh ok. Then I need to use Python and SIP API. |
| 13:00.34 | Samot | Making a "ping call" doesn't prove anything. |
| 13:01.04 | velix | Samot: in my country it does. We've got assigned or not assigned ;) |
| 13:02.09 | Haris | what voip/voice related traffic flows on udp port 5678? |
| 13:02.13 | Samot | Assigned to who |
| 13:02.22 | Samot | Haris: What ever you put on it. |
| 13:02.24 | Samot | It's a port. |
| 13:02.50 | Samot | I could use port 80 for SIP |
| 13:03.15 | Samot | Ports do not support "specific" traffic. |
| 13:03.18 | velix | Samot: I don't need the person's name. I just want to check, if it's assigned or "free" (unregistered). |
| 13:03.27 | Samot | To a person? |
| 13:05.32 | Haris | if there's nothing standard on it, then you just have to say you don't know |
| 13:05.47 | Haris | I already know if its something custom (I put something on it) |
| 13:06.00 | DanQuinney | velix: doing that in the UK would get you in very hot water with the regulator ;) |
| 13:06.02 | Haris | or nothing's on that port |
| 13:06.05 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 13:06.32 | Haris | 18:06:23.180613 IP 172.31.3.58 > 10.20.80.2: ICMP 172.31.3.58 udp port sip unreachable, length 510 |
| 13:06.34 | velix | DanQuinney: I've checked it for Germany, it's legal here right now. |
| 13:07.29 | Samot | Haris: There is a list of "standard" ports available on the Internet. |
| 13:07.42 | Samot | That's about all it is though. |
| 13:07.50 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 13:07.57 | Samot | Just like port 80 on TCP is "standard" for HTTP traffic |
| 13:08.00 | Haris | what does this mean ? it cant reach sip port ? |
| 13:08.06 | Samot | It doesn't mean you have to use it. |
| 13:08.13 | Samot | There's not such thing as a "SIP Port" |
| 13:08.19 | Samot | It's just a "port" |
| 13:08.26 | Samot | That listens on TCP and/or UDP |
| 13:08.37 | Samot | What you bind that to is up to you. |
| 13:09.03 | Samot | Again, you can use port 80 as a port for SIP |
| 13:09.09 | Samot | Instead of HTTP |
| 13:09.14 | Samot | It's up to you. |
| 13:30.18 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:32.58 | *** join/#asterisk jkroon (~jkroon@41.76.202.29) |
| 13:35.34 | *** join/#asterisk flujan (~flujan@189.39.53.65) |
| 13:35.52 | flujan | hi there. Is it possible to exec a dialplan function inside a Agi Script? |
| 13:36.28 | Haris | what does the word unreachable signify in that tcpdump output ? |
| 13:36.32 | Haris | what was unreachable ? |
| 13:37.16 | Haris | and in what direction is something unreachable ? |
| 13:38.25 | Haris | 18:38:09.271894 IP 10.20.80.2.sip > 172.31.3.58.sip: SIP, length: 474 |
| 13:38.25 | Haris | 18:38:09.271912 IP 172.31.3.58 > 10.20.80.2: ICMP 172.31.3.58 udp port sip unreachable, length 510 |
| 13:39.25 | Haris | ah. my asterisk service is down. so its natural its going to say that |
| 13:42.59 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:46.39 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 13:48.10 | [TK]D-Fender | flujan, yes |
| 13:55.42 | flujan | [TK]D-Fender: how so? Can you point me a example? |
| 13:55.53 | [TK]D-Fender | Geet it like any other variable |
| 13:55.58 | [TK]D-Fender | the idea is no different |
| 13:56.08 | [TK]D-Fender | or Set it |
| 13:57.15 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 13:57.38 | Samot | Haris: Unreachable is pretty self explanatory. It can't reach the other side. In the case of SIP that means no response to a request was received. |
| 13:58.01 | Haris | yep. I just realized, this was a simple case |
| 13:58.28 | flujan | [TK]D-Fender: thanks. |
| 14:07.26 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.elen.ucl.ac.be) |
| 14:12.03 | *** part/#asterisk flujan (~flujan@189.39.53.65) |
| 14:31.25 | *** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com) |
| 14:46.34 | *** join/#asterisk casix (~casix@2a01:1c8:d:6::dead) |
| 14:46.39 | casix | hello |
| 14:47.25 | casix | how can I make a dial with pjsip with all information (proxy, user, password) in the cmd, without having any trunk? |
| 14:50.40 | *** join/#asterisk cervajs2 (~cervajs2@gatekeeper.bm.ipex.cz) |
| 14:51.27 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.elen.ucl.ac.be) |
| 14:53.33 | cervajs2 | outgoing call from pjsip webrtc endpoint with rtc-mux negotiate. pjsip endpoit has rtc-mux=yes . setRemoteDescription doesnt have a=rtcp-mux . is it correct? howto confirm that i'm using rtc-mux from asterisk side (and from browser)? thanks |
| 14:56.07 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 14:57.45 | file | the option is rtcp_mux=yes and it would be "rtcp-mux" in the SDP |
| 14:58.00 | [TK]D-Fender | casix, You always have to use an endpoint. You can make it fairly empty though and pass the rest in-line |
| 15:00.09 | cervajs2 | sry. its ok on asterisk side "rtcp_mux : true" |
| 15:00.36 | casix | [TK]D-Fender, where can I find the syntax for the dial to pass the username and password in-line? |
| 15:03.18 | cervajs2 | file: invite from jssip is with a=rtcp-mux but response from asterisk is without a=rtcp-mux . its on ast 13.17.2 . do you think it can be bug? |
| 15:06.12 | file | it's possible but you'd really need to upgrade to the latest version first |
| 15:06.38 | file | I just confirmed in 15 it works as expected |
| 15:06.56 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 15:06.56 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 15:07.28 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.elen.ucl.ac.be) |
| 15:12.03 | cervajs2 | file: how is rtcmux negotiate/require propagated to sip? |
| 15:12.15 | file | I don't understand the question |
| 15:12.37 | cervajs2 | i mean pc = new RTCPeerConnection({rtcpMuxPolicy: "negotiate"}) |
| 15:12.56 | file | it's not, that's policy within the browser over its behavior based on the result of the SDP negotiation |
| 15:13.43 | *** join/#asterisk kharwell (kharwell@nat/digium/x-opyofqqvleheiohs) |
| 15:13.43 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:18.47 | cervajs2 | ops. learning time. rtcpMuxPolicy: "negotiate" is not clear for me. i'll back later |
| 15:22.46 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.elen.ucl.ac.be) |
| 15:25.59 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-ztabdtqgiwnwhypt) |
| 15:26.00 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:31.13 | *** join/#asterisk vandyk (~vandyk@201.90.171.130) |
| 15:34.22 | *** join/#asterisk bford (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 15:34.23 | *** mode/#asterisk [+o bford] by ChanServ |
| 15:37.34 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-vubidefgleasphbs) |
| 15:38.39 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 15:38.53 | *** join/#asterisk znoteer__ (~quassel@107-179-145-194.cpe.teksavvy.com) |
| 15:45.43 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-cogzokbyeseqhisf) |
| 15:45.43 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:47.46 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.elen.ucl.ac.be) |
| 15:52.15 | *** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com) |
| 15:54.29 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 15:59.46 | *** join/#asterisk vandyk (~vandyk@201.90.171.130) |
| 16:00.08 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 16:05.47 | *** part/#asterisk Haris (~haris@unaffiliated/haris) |
| 16:08.28 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 16:15.49 | *** join/#asterisk Worldexe (~Worldexe@95-107-33-134.dsl.orel.ru) |
| 16:32.44 | *** join/#asterisk dl12 (~tyui@luminet-2-130-190.cust.b2b2c.ca) |
| 16:53.01 | *** join/#asterisk Guest91029 (b92009fa@gateway/web/freenode/ip.185.32.9.250) |
| 16:53.21 | Guest91029 | Have anyone seen this odd behaviour? It started when we moved from local database to Amazon RDS. Using Asterisk Realtime |
| 16:53.22 | Guest91029 | http://codepad.org/rvZpQU5L |
| 16:53.44 | Guest91029 | The Asterisk/CLI is frozen or does not respond. Also I did a tcpdump and can see no queries being made. This occurs randomly. |
| 16:55.32 | Guest91029 | Restart of Asterisk service does not work, either does restarting the server |
| 16:55.41 | file | you'd have to get a backtrace to show where it is blocked |
| 16:56.06 | file | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 16:57.18 | Guest91029 | Seems to work when I started with safe_asterisk all of a sudden |
| 16:58.55 | file | any disturbance with the database will make Asterisk upset |
| 16:59.51 | Guest91029 | No core dump was created |
| 16:59.55 | Guest91029 | Meaning it did not crash? |
| 17:00.36 | Guest91029 | Sure database disturbance I get that, I want to know why though so that I can fix it. |
| 17:01.14 | Guest91029 | It SEEMS, that doing a FLUSH HOSTS; in the mysql CLI sorts it out |
| 17:01.17 | Guest91029 | It does not make any sense |
| 17:01.43 | file | it didn't crash |
| 17:01.55 | file | it was likely in a deadlock in database code |
| 17:02.22 | Guest91029 | It worked excellent before we moved the database to RDS. I must say. |
| 17:03.45 | Guest91029 | Any suggestion on how to approach this issue? |
| 17:11.58 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 17:12.50 | *** join/#asterisk awkwardpenguin (~awkwardpe@2607:fb90:285f:62a2:75eb:5450:3952:7091) |
| 17:13.49 | file | a backtrace would be needed to know precisely where it is blocked |
| 17:13.56 | file | the result of which would dictate what to look into |
| 17:19.48 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 17:19.48 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 17:23.31 | Guest91029 | So asterisk needs to crash? Or is one created just by running safe_asterisk? |
| 17:23.51 | file | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock |
| 17:24.06 | file | that details how to get the info |
| 17:29.54 | *** join/#asterisk awkwardpenguin (~awkwardpe@2607:fb90:285f:6888:c8a4:32e9:3a95:3b02) |
| 17:45.43 | *** join/#asterisk awkwardpenguin (~awkwardpe@2607:fb90:825e:da71:4d1:1e3d:8997:9217) |
| 17:49.04 | Guest91029 | @file: yeah ok |
| 17:49.08 | Guest91029 | So I have to recompile asterisk |
| 17:49.27 | Guest91029 | To bad I cannot replicate this on demand |
| 17:57.32 | Guest91029 | Bugger to be having to run asterisk with "seriously impact performance" for weeks until it might occur again :-( |
| 18:14.02 | *** join/#asterisk sekil (~sekil@cable-89-216-195-85.dynamic.sbb.rs) |
| 18:19.39 | *** join/#asterisk awkwardpenguin (~awkwardpe@172.58.40.12) |
| 18:28.45 | *** join/#asterisk SoBlindWolf (~SoBlindWo@go.pcshost.co) |
| 18:30.40 | *** join/#asterisk LunaLovegood (~alice@75.98.139.193) |
| 18:32.50 | LunaLovegood | Is there a way to have a BLF dialplan hint for an extension number instead of for PJSIP devices? Like, I want a BLF light on the SIP phone at the reception desk to go red when someone from the outside dials into the IVR (before any of the office phones rings). |
| 18:33.50 | LunaLovegood | Like, calls coming from the ITSP to a specific number (there's also a fax number and I don't want BLF lights to turn on when we get faxes.) |
| 18:34.15 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_DEVICE_STATE |
| 18:34.22 | file | the primitives exist to do it. |
| 18:36.23 | LunaLovegood | So I can do a Set(DEVICE_STATE(Custom:lamp1)=BUSY) like in the example, but how do I know when to set it to NOT_INUSE? There may be many simultaneous calls. |
| 18:36.52 | file | you'd need to write logic to somehow do it, otherwise there is nothing built in that explicitly does what you want |
| 18:38.01 | [TK]D-Fender | file, is there a hook for GROUP()? |
| 18:38.04 | [TK]D-Fender | for the count |
| 18:38.14 | LunaLovegood | I'd need to deal with race conditions and other multithreading issues in the dialplan? |
| 18:38.23 | file | you can query the current count for a group |
| 18:39.35 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_GROUP_COUNT |
| 18:39.57 | file | unfortunately that's the best I've got |
| 18:40.54 | LunaLovegood | That should work, I think, thanks! |
| 18:41.51 | [TK]D-Fender | Group_count + hangup handlers could probably deal with this pretty efficiently |
| 18:42.09 | [TK]D-Fender | low liklihood of race |
| 18:51.09 | velix | [TK]D-Fender: Are you a rentable Asterisk consultant? |
| 18:52.23 | [TK]D-Fender | possibly. PM me |
| 19:02.37 | LunaLovegood | It does work! Thanks again! |
| 19:07.12 | LunaLovegood | I have to do more dialplan "programming" if I want to have the users be able to decide for themselves whether they want a password on their voicemail or not, right? |
| 19:09.09 | [TK]D-Fender | * will always ask for the VM password |
| 19:09.14 | [TK]D-Fender | I supposethey COULD leave it blank |
| 19:09.18 | [TK]D-Fender | but it will still ask |
| 19:09.31 | [TK]D-Fender | at which point you'd just hit # for instance I believe |
| 19:09.43 | [TK]D-Fender | if you want it to skip outright, then that's in how you call it |
| 19:10.38 | *** join/#asterisk miralin (~Thunderbi@81.177.59.227) |
| 19:31.29 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 19:50.02 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 20:00.51 | *** join/#asterisk awkwardpenguin (~awkwardpe@2607:fb90:8247:6ebd:fd73:a270:e789:df28) |
| 20:37.15 | *** join/#asterisk sekil (~sekil@cable-89-216-195-85.dynamic.sbb.rs) |
| 20:45.13 | *** join/#asterisk znoteer_ (~max@107-179-145-194.cpe.teksavvy.com) |
| 20:58.27 | *** join/#asterisk tzafrir (~tzafrir@62-90-199-247.barak.net.il) |
| 21:02.59 | *** join/#asterisk vandyk (~vandyk@179.232.88.214) |
| 21:18.26 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-uvyyebtrfnsxpqmi) |
| 21:34.47 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 21:39.53 | *** join/#asterisk sarlalian (~sarlalian@107.170.239.102) |
| 21:56.23 | *** join/#asterisk vandyk (~vandyk@179.232.88.214) |
| 22:02.42 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:04.19 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 23:20.28 | *** join/#asterisk mahlon (~mahlon@martini.nu) |