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| 08:42.37 | ntz | hello |
| 08:45.25 | ntz | one stupid Q: |
| 08:45.43 | ntz | why does: core shutdown gracefully never works o.O ???? |
| 08:45.56 | ntz | core shutdown now works but the graceful one never |
| 08:52.00 | wdoekes | ntz: not a stupid question. the graceful shutdown should work, but waits for all calls to end. you may need to debug what is holding back asterisk (core show channels, sip show channels, sip show objects, ...) |
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| 09:04.16 | ntz | wdoekes: only what I have still up are established a T38 peers, perhaps this |
| 09:04.33 | ntz | okay, explained, understood, taken .... thanks !!!! |
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| 09:23.09 | cbeyerlein | heya, I'm having troubles getting the dialplan matching logic in pbx.c logging debug output.. already added logger channel with DEBUG and core set debug 9 |
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| 10:20.01 | melmoth81_ | hello everybody, I need some help |
| 10:22.14 | ntz | melmoth81_: cools story, go on :) |
| 10:22.25 | ntz | s/s//1 |
| 10:23.19 | melmoth81_ | Hello ntz :) |
| 10:24.37 | ntz | in order to help you (assumingly by answering your question) we need to know the question |
| 10:25.14 | melmoth81_ | When Dian an extension, I run a bash script than send via telnet the CALLERID to another system. The problem is than I need to send another script when the call is hangup |
| 10:26.08 | melmoth81_ | The problem is that i'm trying to use the h extension, buy the CALLERID es always "h" |
| 10:26.22 | ntz | melmoth81_: wait, what protocol .... you shouldn't be sending that via telnet but read it enwrapped in the trunk within the call by receiving application |
| 10:26.54 | melmoth81_ | I only need to send a string |
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| 10:30.12 | melmoth81_ | <ntz> the receiving application are listeing the telnet port. There is any problem? |
| 10:32.13 | ntz | melmoth81_: depends, perhaps not ... that was my initial reaction because it sounds a bit overengineered ... I'm also sending CALLERID[0..?8] but I can read it by receiving application from sip trunk |
| 10:32.37 | ntz | or generally speaking, I can postprocess that on client while read from sip trunk |
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| 10:44.53 | ntz | melmoth81_: eg from your dialplan do something like this: Set(CALLERID(number)=${EXTEN}); |
| 10:51.47 | ntz | melmoth81_: https://www.voip-info.org/wiki/view/Setting+Callerid |
| 10:57.50 | melmoth81_ | <ntz> I have this... |
| 10:57.54 | melmoth81_ | exten => _[1234567]0001,1,Answer() same => n,Set(EXT_DESTINO=${DB(DESVIO/${EXTEN})}) same => n,GotoIf($["${EXT_DESTINO}" = ""]?NORMAL:DESVIO) same => n(NORMAL),System(/etc/asterisk/pServer/cctv_integracion.sh 1 ${EXTEN} ${CALLERID}) ;Integramos con CCTV OPERACION LLAMADA same => n,Dial(SIP/${EXTEN}) same => n,Hangup same => n(DESVIO),Dial(SIP/${EXT_DESTINO}) same => n,Hangup exten => h,1,System(/etc/asterisk/pSe |
| 10:58.43 | melmoth81_ | ups... sorry. I think it's hard to see fine |
| 10:59.05 | ntz | use pastebin next time please |
| 10:59.06 | melmoth81_ | in this line same => n(NORMAL),System(/etc/asterisk/pServer/cctv_integracion.sh 1 ${EXTEN} ${CALLERID}) |
| 10:59.30 | melmoth81_ | i run a scritp with 3 params (1), EXTEN and CALLERID |
| 10:59.51 | melmoth81_ | y need to run the script again on hangup so I try to use this line |
| 10:59.57 | melmoth81_ | exten => h,1,System(/etc/asterisk/pServer/cctv_integracion.sh 0 ${EXTEN} ${CALLERID}) |
| 11:01.22 | melmoth81_ | the problem is that EXTEN is allways "h"... |
| 11:03.36 | *** join/#asterisk Rico (~Rico@unaffiliated/rico29) |
| 11:03.37 | Rico | hio |
| 11:05.19 | Rico | I have a question about logger : I only have one line containing "full => notice,warning,error,debug,verbose,dtmf,fax" in logfiles section, but logs are still sent to syslog. how can I disable that ? |
| 11:06.39 | Rico | (Asterisk 13.10.0) |
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| 11:18.14 | ntz | Rico: perhaps you havent reload() yet |
| 11:18.35 | Rico | I've reload logger module |
| 11:18.41 | Rico | CLI>logger reload |
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| 11:19.39 | ntz | Rico: show us output from `logger show channels' |
| 11:20.57 | Rico | ntz: http://fpaste.scsys.co.uk/554474 |
| 11:21.29 | Rico | everything goes to /var/log/asterisk/full, and to /var/log/syslog TOO |
| 11:21.32 | ntz | well, this clearly says, that syslog.xxx chans ain't enabled |
| 11:21.48 | ntz | try restarting asterisk please |
| 11:22.05 | Rico | I can not |
| 11:22.06 | ntz | entirely, logger show channels just says what it does |
| 11:22.55 | ntz | Rico: why can't you restart that when you can modify it ??? |
| 11:23.05 | Rico | ntz: I can reload logger module |
| 11:23.18 | Rico | but I can not restart asterisk, having about 100calls running on it |
| 11:25.13 | ntz | Rico: to be honest, you see what logger show channels says, so why it writes still to syslog is kinda mysterious |
| 11:25.29 | Rico | ntz: yes, that's why I'm asking here :) |
| 11:38.41 | ntz | Rico: does it still persist ? I'm asking because I've player with logging a relatively much, configuring it for logstash and so on and I've never hit into this issue |
| 11:38.50 | ntz | **played |
| 11:39.11 | Rico | yes I'm still having the issue |
| 11:39.16 | Rico | but I did not restart asterisk |
| 11:40.19 | ntz | couldn't it be some crazy rsyslog queue ??? |
| 11:43.10 | Rico | nothing related to '/var/log/asterisk*' in rsyslog config |
| 11:44.34 | ntz | you've said it still goes in via /dev/log (eg processed by rsyslog based on facility.prio) |
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| 11:56.51 | melmoth81_ | <ntz> Do you have any idea?? |
| 12:00.26 | ntz | melmoth81_: yes, I've already pointed you https://www.voip-info.org/wiki/view/Setting+Callerid .... just Set() within your dialplan exporting whatever you need and read it on client side, no idea with telnet, really |
| 12:04.19 | melmoth81_ | <ntz> I allready know that, but if i use the h extension, I cat use CALLERID. I'm exporting the information at the beginning, but i can't export at the hangup. |
| 12:04.35 | melmoth81_ | Not is a telnet problem |
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| 13:18.56 | [TK]D-Fender | melmoth81_, the problem is that EXTEN is allways "h"... <--- because that is where you are |
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| 14:25.18 | MacroMan | My WebSocket client gets randomly booted by asterisk with the following error: https://paste.ngx.cc/b6543216e518dfb2 |
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| 14:26.03 | MacroMan | I can register again straight away with no problem, but I don't think getting the error is a good think. I'm using a SSL certificate from Letsencrypt |
| 14:26.32 | ntz | MacroMan: https://issues.asterisk.org/jira/browse/ASTERISK-22750 |
| 14:31.49 | Samot | ntz: I'm not sure a Asterisk 1.8 issue is the problem. |
| 14:31.57 | Samot | MacroMan: What version of Asterisk? |
| 14:34.40 | ntz | MacroMan: unresolved is the status |
| 14:35.30 | Samot | ntz: https://issues.asterisk.org/jira/browse/ASTERISK-25312 <-- This is actually closer to the issue he is having. |
| 14:35.44 | Samot | Because he's using WSS. |
| 14:36.05 | Samot | It's why I'm asking what version is running. |
| 14:36.25 | ntz | yeah, cool |
| 14:37.32 | Samot | 1.8 is dead at the end of the day. So if you are running 1.8 and are hit with bugs, your fix is upgrading to a supported version. |
| 14:38.15 | MacroMan | I'm using 14 |
| 14:38.30 | Samot | 14.0.0? |
| 14:38.34 | Samot | 14.1? |
| 14:39.18 | Samot | This should have been fixed in 14.0, if it's not I would submit a ticket. |
| 14:39.19 | MacroMan | Erm. 14 latest when I built it about 3 weeks ago |
| 14:39.47 | Samot | Or try updating to the current version of 14. |
| 14:40.32 | MacroMan | Asterisk 14.2.1 built by root @ asterisk on a x86_64 running Linux on 2017-02-01 14:47:14 UTC |
| 14:41.01 | file | the referenced issue is not the same thing |
| 14:41.37 | [TK]D-Fender | Standard: 14.3.0 (2017/02/13) |
| 14:41.46 | file | it was if a fatal error occurred and we killed the websocket it could still remain open when it shouldn't |
| 14:42.32 | MacroMan | I haven't had it happen during a call, so I don't know if it would interrupt a call or not |
| 14:42.49 | MacroMan | [TK]D-Fender, Thanks, I'll upgrade |
| 14:43.39 | Samot | I just said it was the closest of the two tickets. |
| 14:43.51 | Samot | I wasn't sure if it was 100% related. |
| 14:44.00 | Samot | But more related than the 1.8 ticket. |
| 14:46.00 | MacroMan | Ticket 22750 doesn't really describe what is happening. 25312 is close. I'll upgrade to 14.3.0 and monitor |
| 14:52.22 | *** join/#asterisk phrearch (~spindle@34.tbb.spindle.osso.nl) |
| 14:52.29 | phrearch | hi |
| 14:54.54 | phrearch | i wonder where i can find a working modules.conf for asterisk 14 |
| 14:55.33 | phrearch | i moved the existing modules.conf that works with asterisk 11 to asterisk 14, removed chan_local, but i still get an error about pjsip |
| 14:55.43 | phrearch | `symbol lookup error: /usr/lib/asterisk/modules/res_pjsip_phoneprov_provider.so: undefined symbol: ast_phoneprov_provider_register` |
| 14:56.35 | MacroMan | phrearch, Here is modules.conf for 14 produced by 'make samples': https://paste.ngx.cc/be8989eb4a2bf748 |
| 14:57.00 | phrearch | MacroMan: ah thanks! i'll try make samples next time |
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| 14:57.28 | MacroMan | yw |
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| 15:15.04 | phrearch | is pjsip an alternative to chan_sip? |
| 15:16.33 | file | yes. |
| 15:17.00 | jkroon | hi all, it would seem VALID_EXTEN doesn't deal with labels for priority? |
| 15:17.12 | phrearch | file: do i need to disable chan_sip to use pjsip or is it ok to keep them both enabled? |
| 15:17.23 | file | you can use both but they can't be on the same ports |
| 15:17.30 | phrearch | ah i see |
| 15:17.33 | jkroon | eg, VALID_EXTEN(,${EXTEN},tech-${CHANNEL(channeltype)}) ? |
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| 15:19.21 | phrearch | https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial states that a client needs a generated client certificate, but how does that work in the browser? |
| 15:19.53 | phrearch | trying to figure out how to enable webrtc in asterisk |
| 15:20.54 | file | for TLS a client certificate is needed, for WSS it is not needed |
| 15:21.01 | file | same for DTLS-SRTP |
| 15:21.10 | file | the browser generates a certificate itself to use for DTLS-SRTP |
| 15:21.29 | *** join/#asterisk yoink (~yoink@unaffiliated/yoink) |
| 15:21.35 | phrearch | ah, so for the browser, it should be fine without the additional client certificate action |
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| 15:22.06 | file | yes |
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| 15:40.34 | igcewieling | heh, nice. I'm seeing 16x compression ratios on a zfs deduplicated volume which is being used to serve polycom firmware and configs to phones. |
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| 15:54.19 | MacroMan | Disconnecting my WSS client doesn't unregister the client with Asterisk. How can I go about unregistering disconnected clients automatically. |
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| 15:57.41 | MacroMan | file, phrearch, That's not correct. You will need a valid SSL cert to use WSS |
| 15:58.00 | file | you need an SSL certificate on the server |
| 15:58.17 | file | and there is no method to have it automatically unregister on disconnect |
| 16:01.28 | MacroMan | phrearch, This is the script I wrote that will install Asterisk 14 with WSS support on Ubuntu 16.04 (should also work on 16.10): https://gist.github.com/MacroMan/c40df7545516aec2cb7836bf0b0d1a63 |
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| 16:19.33 | samwierema | I have a dialplan where I answer an incoming channel and do a playback. Then I Hangup(). Is it correct that it sends 603 Declined on Hangup()? |
| 16:21.25 | [TK]D-Fender | no |
| 16:21.43 | [TK]D-Fender | 603 is what * gives if you don't answer and have no other status to forward |
| 16:22.37 | samwierema | [TK]D-Fender: hmm, ok. I'll dive into it some more. Thanks |
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| 17:03.59 | Samot | 603 is basically the "I got your call but I don't/won't handle for whatever reason". |
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| 17:04.39 | [TK]D-Fender | "didn't answer and have nothing else to say" |
| 17:04.46 | *** part/#asterisk mknooihuisen (~mknooihui@12.150.48.70) |
| 17:04.48 | Samot | 603 = raspberry of call responses. |
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| 17:05.15 | tompaw | Morning |
| 17:06.19 | tompaw | Anyone having issues with WebRTC and 13.14? I have exactly the same config as in 13.13.1 and Asterisk can't call my WS endpoints. There are no error/debug messages - the last thing I see is "Called X" and nothing happens. Even SIP messages are not generated. |
| 17:06.25 | Samot | I feel like 603 should return that "wha wha whaaaa" trumpet sound when you get it. |
| 17:07.00 | *** join/#asterisk sh1rtybird (~quassel@beepbeep.serverpit.com) |
| 17:07.55 | [TK]D-Fender | tompaw, PASTEBIN <- |
| 17:08.39 | tompaw | [TK]D-Fender: http://pastebin.com/zQaL5FZM |
| 17:08.50 | tompaw | That's all logs including pjsip enabled, the next step is debug |
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| 17:18.49 | MajesticFudgie | Having a wierd issue where as soon as an internal call is answered between my deskphone and X-Lite the call drops. The console doesnt show an error either |
| 17:19.39 | [TK]D-Fender | And we should already have a call to look at right now.... |
| 17:19.39 | igcewieling | that sounds like directmedia needs to be no. |
| 17:19.48 | [TK]D-Fender | Sounds like nothing to me |
| 17:20.38 | MajesticFudgie | igcewieling where should I set that? Incoming calls work fine when answered by my desk phone, not tested with X-Lite |
| 17:21.16 | igcewieling | look at sip.conf.sample |
| 17:21.56 | igcewieling | if calls to a different phone at the same location works, then I'd look at codec isues, though that should be logged in the CLI. |
| 17:23.24 | MajesticFudgie | kk, I'll try your suggestion |
| 17:23.44 | MajesticFudgie | igcewieling, fixed and working :D |
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| 17:23.50 | igcewieling | what was it? |
| 17:23.56 | MajesticFudgie | Literally what you suggested |
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| 17:25.40 | tompaw | [TK]D-Fender: http://pastebin.com/255Mn5Jr |
| 17:25.45 | tompaw | nice clean debug |
| 17:27.32 | [TK]D-Fender | I see no call coming in. No SIP debug . No dialplan processing. No call ending. Nothing. |
| 17:27.41 | *** part/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 17:29.45 | tompaw | Because no dialplan is involved. This simply needs to call '2' and bridge it to an ARI app. |
| 17:30.06 | tompaw | Source of transaction state change is TRANSPORT_ERROR << could this be a clue |
| 17:30.17 | [TK]D-Fender | we should see SIP DEBUG for this |
| 17:30.38 | tompaw | [TK]D-Fender: pjsip logging is enabled when this happens and produces exactly 0 output |
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| 18:12.00 | tompaw | The inv session does NOT have an invite_tsx |
| 18:12.06 | tompaw | What's "tsx"? |
| 18:13.34 | tompaw | BTW the reason why I have to move from 13.13.1 to 13.14 is that 13.13.1 goes shit bananas when recordings.mute or recordings.unmute is used. It triggers hundreds of thousands of "INTERNAL_OBJ: FRACK!, Failed assertion bad magic number 0x0..." errors, which allegedly are fixed in 13.14 |
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| 18:27.06 | file | INVITE transaction |
| 18:34.54 | tompaw | file: a-ha. any idea why this happens on 13.14? |
| 18:35.22 | file | no, I would need to see a full pjsip set logger on from start to finish |
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| 19:13.51 | tompaw | file: what do you mean from start to finish - it's not showing ANYTHING for this originate |
| 19:14.02 | file | so it's not even sending the INVITE? |
| 19:14.15 | file | then you'd need to provide configuration details and Dial string |
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| 19:49.49 | dan_j | Hi. In Asterisk 11, users said that when they transferred a call, their phones updated with the transferred calls CLI. In Asterisk 13, that isn't happening and the receiving phone still shows the transferer's CLI. Can the original functionality be restored? |
| 19:50.11 | dan_j | this is when an attended transfer managed by the SIP phones feature buttons. |
| 19:54.41 | igcewieling | the depends on how you are doing the transfer. |
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| 19:55.59 | igcewieling | There are at least three things which are "transfers" and one other which could be considered "Transfer" |
| 19:58.06 | dan_j | Three? |
| 19:58.26 | dan_j | DTMF transfer, transfers using the SIP feature buttons, whats the third? |
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| 20:14.41 | tompaw | file: it's not even sending the INVITE. It's a simple PJSIP/2 in this case (which is a WS endpoint, if it matters). I can extract more stuff, but I don't know what to go after. |
| 20:15.22 | tompaw | Again, it's a 1:1 same environment (ari daemons, configs, originate call, dialplan, etc.) between 13.13.1 and 13.14 - the only difference is the asterisk version. |
| 20:15.26 | file | is it WSS or WS? |
| 20:15.38 | tompaw | WSS being WS + SSL? |
| 20:15.41 | file | yes. |
| 20:15.51 | tompaw | Then yes, it uses a valid cert. |
| 20:16.01 | tompaw | Ooh, I actually haven't checked the web console |
| 20:16.03 | tompaw | One sec. |
| 20:16.05 | file | rmudgett is currently working on that. |
| 20:16.25 | tompaw | As in? It's a bug? |
| 20:16.47 | file | there is a bug in WSS |
| 20:17.19 | file | https://issues.asterisk.org/jira/browse/ASTERISK-26796 |
| 20:18.04 | tompaw | (when are you going to fix this https issue for chris sake it's been years now) |
| 20:18.14 | rmudgett | That hasn't changed between v13.13.1 and v13.14.0 |
| 20:18.29 | file | yes it did |
| 20:18.30 | tompaw | yeah it says 14.x |
| 20:18.37 | tompaw | 14.3.0 |
| 20:18.52 | file | that's what it was reported against, but it's applicable to 13.14.0 |
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| 20:23.01 | tompaw | So I can pick between INTERNAL_OBJ: FRACK! in 13.13.1 and WSS error in 13.14 :F |
| 20:24.16 | file | it's not hard to undo the WSS change |
| 20:24.43 | tompaw | I need to confirm this is really the issue here, no errors in Firefox web console |
| 20:24.57 | tompaw | unless it somehow happens on the Asterisk side |
| 20:25.29 | file | https://www.irccloud.com/pastebin/vAgimfjg/ |
| 20:31.52 | igcewieling | dan_j: The Transfer dialplan application. Dial could be be called a "transfer" too. |
| 20:32.02 | tompaw | file: awesome, thanks |
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| 20:46.10 | dan_j | igcewieling: ah. Ok. In this case its not via the dial plan. The user presses the Transfer key on the sip phone, does a normal dial (which does use the dialplan), speaks to the recipient and the presses the transfer key again to complete the transfer. So i assume the actual transfer is done via sip signalling. In asterisk 11, that caused the recipients phone |
| 20:46.11 | dan_j | to update the display with the transferred call's callerid. But in 13, that has stopped. |
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| 23:07.09 | SpaceInvaders | Hi! I'm running Asterisk at home using OBi110s (2) and ready to start testing with a service provider. It seems many of them have upped their prices significantly. Is it still possible, in the US, to find a SP that'll give me good service and unlimited calls in the US for US$5.00/month per line? |
| 23:11.01 | igcewieling | I doubt it. Most people go with per/min rates. |
| 23:11.51 | SpaceInvaders | do you have a preferred provider? and what do you like about them? |
| 23:12.53 | igcewieling | I work for a phone company, but I really hate talking on the phone. I use Vitelity, but my usage is very low. |
| 23:13.29 | igcewieling | most of my usage is test faxes. |
| 23:13.46 | SpaceInvaders | My usage is pretty low but I am not sure about my wife's. |
| 23:15.45 | SpaceInvaders | are you paying 1.44 cents per minute for US and CA? |
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| 23:25.31 | [TK]D-Fender | that's pricey |
| 23:25.41 | [TK]D-Fender | easy to find 1c/m |
| 23:25.56 | SpaceInvaders | That's what I was thinking. It's what they advertise on the Vitelity web site |
| 23:26.10 | SpaceInvaders | Hi TK :-) |
| 23:26.13 | [TK]D-Fender | https://www.flowroute.com/voice/ |
| 23:26.56 | [TK]D-Fender | https://voip.ms/ |
| 23:27.02 | [TK]D-Fender | All 1c/m tops |
| 23:27.44 | SpaceInvaders | TK if I'm running Asterisk I want the 2-way SIP trunking, correct? |
| 23:28.21 | [TK]D-Fender | Depends on you |
| 23:28.33 | [TK]D-Fender | Sometimes you'll want 1 provider for inbound, maybe another for outbound |
| 23:28.37 | [TK]D-Fender | it's entirely up to you |
| 23:28.41 | SpaceInvaders | ahhh ya |
| 23:28.52 | SpaceInvaders | I never even thought about splitting providers |
| 23:28.54 | [TK]D-Fender | Often mutilple for outbound as that cost teends to vary depending where you call |
| 23:29.28 | SpaceInvaders | I just want a quality and low-cost provider that'll offer me a solution that won't require me to pay for things I already do with my Asterisk server. |
| 23:31.13 | [TK]D-Fender | The last 3 providers mentioned all let you pick exactly what you want in pieces. |
| 23:31.20 | [TK]D-Fender | What is your actual expected usage? |
| 23:31.40 | SpaceInvaders | Mine is low. I'm not sure about my wife's or how to quantify usage. |
| 23:31.51 | SpaceInvaders | her's is definitely more than mine |
| 23:32.00 | [TK]D-Fender | Get a good idea of your expected usage and start shopping |
| 23:32.19 | [TK]D-Fender | but those 3 providers are amongst the best a-la-carte |
| 23:32.35 | SpaceInvaders | thanks! |
| 23:32.59 | [TK]D-Fender | there are "unlimited" providers, but it's good to keep the DIY cost in mind |
| 23:33.06 | [TK]D-Fender | You have a lot of choice. |
| 23:33.36 | SpaceInvaders | Ya. I was hoping to find a $3/month or $5/month provider like i used to see but apparently those plans are all "retired". |
| 23:33.50 | [TK]D-Fender | The age of free lunch came and went |
| 23:34.03 | [TK]D-Fender | now that it's a known thing they go back to more standardized rates |
| 23:34.04 | SpaceInvaders | yup :/ |
| 23:34.16 | [TK]D-Fender | but VoIP still lets you DIY since you control your selection and call handling |
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