| 00:08.06 | *** part/#asterisk kharwell (kharwell@nat/digium/x-bmrosyzvbqriktyo) |
| 00:28.54 | *** join/#asterisk aness (~aness@cm-84.209.155.137.getinternet.no) |
| 00:41.18 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 00:46.45 | *** join/#asterisk miralin (~Thunderbi@194.8.128.50) |
| 00:48.33 | *** join/#asterisk overyander (~Jeff@209.141.208.197) |
| 00:57.19 | *** join/#asterisk qloogkm[m] (qloogkmmat@gateway/shell/matrix.org/x-jwbvwhtvxqsymbur) |
| 01:07.46 | Kobaz | http://ichiveugc-a.akamaihd.net/__8482b0aedeb6921c88454a2f430a8046_width-600.gif |
| 01:19.53 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:19.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); 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 |
| 01:23.00 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
| 01:44.05 | *** join/#asterisk Dovid (~dovid@ool-321d614a.dyn.optonline.net) |
| 02:53.13 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 02:54.40 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 02:55.23 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 05:17.26 | *** join/#asterisk Aboba (~Bob@node-1w7jr9sqc1awwajjrix4g9a3v.ipv6.telus.net) |
| 05:24.14 | *** join/#asterisk vstemen (~vince@ip72-202-187-29.ok.ok.cox.net) |
| 06:10.15 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 06:25.11 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 06:31.52 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 06:56.36 | *** join/#asterisk envisean (~envisean@c-67-183-38-111.hsd1.wa.comcast.net) |
| 06:57.02 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 06:57.51 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 07:07.46 | *** join/#asterisk KValchev (~KValchev@ns.atsoftconsult-bg.com) |
| 07:15.24 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:ba27:ebff:fee4:a52f) |
| 07:16.26 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 07:23.46 | *** join/#asterisk trelane (trelane@unaffiliated/trelane) |
| 07:27.54 | *** join/#asterisk KValchev (~KValchev@ns.atsoftconsult-bg.com) |
| 07:28.40 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 07:58.08 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 08:07.15 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:17.16 | *** join/#asterisk pchero_work (~pchero@2a00:c80:1072::141c) |
| 08:20.17 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:28.58 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 08:35.55 | *** join/#asterisk KValchev (~KValchev@ns.atsoftconsult-bg.com) |
| 08:36.06 | *** join/#asterisk s-mutin (~s-mutin@mail.ska-irk.ru) |
| 08:40.58 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 08:41.07 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 08:57.50 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 09:19.44 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 09:31.31 | *** join/#asterisk phrearch (~spindle@34.tbb.spindle.osso.nl) |
| 09:37.50 | *** join/#asterisk pdugas (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 09:51.40 | phrearch | hi |
| 09:52.27 | phrearch | i just got a webrtc demo working with audio. woohoo! now i was wondering whether it would be possible to have video as well between two clients. anyone an idea if that can be done with asterisk? |
| 09:52.53 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 09:53.43 | *** join/#asterisk jkroon (~jkroon@165.16.204.44) |
| 09:55.32 | *** join/#asterisk TriJetScud (~TriJetScu@69.172.162.252) |
| 09:58.47 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 10:09.31 | *** join/#asterisk Eloy_ (~Eloy@5.149.168.66) |
| 10:09.34 | *** join/#asterisk Oatmeal (~Suzeanne@c-68-45-50-54.hsd1.nj.comcast.net) |
| 10:14.54 | *** join/#asterisk Kaian (~kaian@6.62-99-78.static.clientes.euskaltel.es) |
| 10:34.44 | *** join/#asterisk Y04NN (~y04nn@178.18.54.206) |
| 10:39.12 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 11:12.40 | DanQuinney | Morning all, I have `send_pai` and `send_rpid` set to "yes" in pjsip.conf which rightly sends an in-dialog INVITE to the peer whenever the connected line info changes but these INVITES are exposing the internal extension numbers - how do I stop this? |
| 11:12.56 | DanQuinney | ask for any relevant configs :) |
| 11:19.38 | *** join/#asterisk bof23 (~Thunderbi@185.13.183.107) |
| 11:40.55 | *** join/#asterisk Kaian (~kaian@6.62-99-78.static.clientes.euskaltel.es) |
| 11:44.23 | ConSi | hello |
| 11:44.35 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 11:44.44 | ConSi | I've faced same issue during asterisk startup as in https://issues.asterisk.org/jira/browse/ASTERISK-26744 |
| 11:45.19 | ConSi | It's on debian testing using openssl 1.1.0e |
| 11:53.29 | sriharsha | DanQuinney, are the INVITES exposing the internal extensions in the To field or in the PAI, RPID fields? |
| 11:57.11 | DanQuinney | sriharsha: http://paste.codebasehq.com/pastes/uvpqawdno8036fqk54 |
| 11:57.14 | sriharsha | I also wonder if there is any way to have separate numbers in the To: and PAI,RPID fields |
| 11:57.55 | stefan27 | if you disable send_pai and send_rpid, you can tailor your own headers at least in chan_sip with SIPAddHeader() application |
| 11:58.24 | sriharsha | DanQuinney, what numbers are you setting to CONNECTEDLINE(num)? |
| 11:59.13 | DanQuinney | sriharsha: we're not calling CONNECTEDLINE at all, the in-dialog invite is being sent when the call is connecting to an queue agent |
| 11:59.49 | sriharsha | stefan27, sure, with PJSIP you have to set PJSIP_HEADER(add, Header). However, if I disable send_pai, send_rpid, PJSIP connected line info changes are not being propagated; is this possible? |
| 11:59.53 | DanQuinney | stefan27: that's our current work around in * v1.8, currently upgrading to v13.14.0 and wanted to get around having to do the same |
| 12:00.45 | sriharsha | DanQuinney, perhaps then you could change the CONNECTEDLINE(num) to a valid public extension? |
| 12:00.52 | stefan27 | Sorry can't help you sriharsha |
| 12:01.16 | DanQuinney | sriharsha: where would one set CONNECTEDLINE? |
| 12:01.18 | stefan27 | I never needed connectedline |
| 12:03.06 | *** join/#asterisk aiksa[LV] (5f822365@gateway/web/freenode/ip.95.130.35.101) |
| 12:03.21 | aiksa[LV] | Hello; |
| 12:04.11 | aiksa[LV] | quick question regarding ARI. what is correct way of setting CALLERID(num) when doing channel create through POST /channels/create |
| 12:04.56 | stefan27 | DanQuinney are you upgrading from chan_sip to chan_sip? |
| 12:05.08 | DanQuinney | to chan_pjsip yes |
| 12:05.13 | *** join/#asterisk sekil (~sekil@net120-34-245-109.mbb.telenor.rs) |
| 12:05.27 | stefan27 | all right |
| 12:05.34 | sriharsha | DanQuinney, I wrote an example here: http://paste.codebasehq.com/pastes/c2f8puw6gxa9ke6bls |
| 12:06.50 | DanQuinney | thanks sriharsha, I'll give that a shot |
| 12:07.30 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 12:18.09 | *** join/#asterisk Y04NN_ (~y04nn@178.18.54.206) |
| 12:53.25 | phrearch | anyone know if asterisk can handle video streams as well using webrtc? |
| 13:07.27 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:09.29 | stefan27 | I think you need a third party vp8 patch https://github.com/meetecho/asterisk-opus , that one was built for asterisk 11.1.2 |
| 13:10.01 | stefan27 | But I dont know, you need a recent version of asterisk either way, because there were a few video related bugs in older versions |
| 13:10.33 | stefan27 | recent versions of webrtc seem to support h264 and other codecs, I have not looked into it after that happend though |
| 13:11.00 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 13:12.22 | phrearch | does asterisk 13 support vp8? |
| 13:13.31 | stefan27 | not my version (13.5.0), but it supports vp8 passthrough if you apply that patch |
| 13:15.15 | stefan27 | what does "core show codecs" output for you? |
| 13:19.05 | stefan27 | frankly I don't remember what didn't work without that patch... un-edited asterisk 13.13.1 seems to know about codec vp8, but I'm not sure if it's capable of transmitting RTCP FIR, and it may not have a format for vp8 |
| 13:20.16 | stefan27 | video rtp is always passthrough either way |
| 13:38.49 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 13:46.25 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 13:52.39 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:57.22 | sriharsha | stefan27, I just experimented with turning send_pai, send_rpid on and off and found that the CONNECTEDLINE updates are only being sent when either of them are set to yes. They are not set when they are both no, which is the default in PJSIP. |
| 13:57.25 | DanQuinney | this is the correct way of adding PJSIP headers right? exten => peer15,42,Set(PJSIP_HEADER(add,X-Test)=testing) |
| 13:57.44 | sriharsha | DanQuinney, yes |
| 13:57.48 | DanQuinney | I'm having a really bad day today with nothing working lol |
| 13:58.23 | sriharsha | You have to be careful though, perhaps you are setting the header on an incoming channel? |
| 13:59.04 | DanQuinney | I'm setting it just before I do a Dial |
| 13:59.18 | sriharsha | So, do you want that header to be sent to the SIP peer on peer15 extension? If so, you should set it in the callee's pre-dial handler |
| 13:59.32 | DanQuinney | I do |
| 14:00.08 | DanQuinney | ah, pre -dial handers were added in v11 |
| 14:00.11 | DanQuinney | makes sense |
| 14:00.30 | sriharsha | So are you using pre-dial handlers? |
| 14:00.48 | DanQuinney | I am not, but I shall add one |
| 14:00.51 | sriharsha | ok |
| 14:01.05 | DanQuinney | as * v1.8 and chan_sip didn't need them :) |
| 14:01.28 | sriharsha | yeah, I too had to deal with that shit while upgrading to PJSIP |
| 14:01.58 | sriharsha | I hope PJSIP stays for a while, else all the time spent will not be worth |
| 14:02.11 | DanQuinney | ha, I too hope it's here to stay! |
| 14:08.39 | stefan27 | i'd think so, listening to discussions of asterisk developers they always gab about pjsip, while chan_sip is dead :) |
| 14:23.50 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 14:26.11 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 14:32.24 | *** join/#asterisk Y04NN (~y04nn@178.18.54.206) |
| 14:43.12 | *** join/#asterisk vince1 (~vince@ip72-202-187-29.ok.ok.cox.net) |
| 14:43.54 | igcewieling | chan_pjsip is the future of Asterisk. I, however, prefer to live in the present, where there are lots of docs for chan_sip. |
| 14:44.34 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 14:46.29 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-masrtpqvwxdyityu) |
| 14:46.29 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:00.28 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:00.28 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:04.09 | *** join/#asterisk kharwell (kharwell@nat/digium/x-ugdknqztpgzdsinh) |
| 15:04.09 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:05.46 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 15:05.46 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 15:09.42 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 15:11.54 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 15:13.10 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 15:15.55 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-nqlpbxssmfreqpuf) |
| 15:17.36 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 15:51.50 | *** join/#asterisk yoink (~yoink@unaffiliated/yoink) |
| 15:57.48 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-tusmoafluwyqrpff) |
| 15:57.48 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 16:02.03 | *** join/#asterisk Alblasco1702 (~Alblasco1@ip5456b46b.speed.planet.nl) |
| 16:07.40 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 16:09.31 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 16:19.47 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 16:20.08 | [TK]D-Fender | <igcewieling> chan_pjsip is the future of Asterisk. I, however, prefer to live in the present, where there are lots of docs for chan_sip. <- yyou're on 11. THat is the past. |
| 16:20.17 | [TK]D-Fender | 11 = Disco |
| 16:20.38 | igcewieling | [TK]D-Fender: get over it [TK]D-Fender, not everyone is moving to Asterisk 13 |
| 16:21.06 | [TK]D-Fender | Nothing to get over |
| 16:21.12 | [TK]D-Fender | 11 is the past, not present |
| 16:21.25 | [TK]D-Fender | I accept that you're sticking with it past its support date. |
| 16:21.32 | [TK]D-Fender | But don't call it "present" |
| 16:21.40 | WIMPy | The good thing about the past is that we had working things then. |
| 16:21.41 | [TK]D-Fender | * 14 is RELEASED |
| 16:21.59 | [TK]D-Fender | And chan_sip works just fine on 13 the same as 11 really |
| 16:22.09 | [TK]D-Fender | So trying to stay on a version for 7 years... |
| 16:22.10 | igcewieling | Bitching about it isn't going to convince anyone who has not already upgraded. |
| 16:22.11 | [TK]D-Fender | well that's on you |
| 16:22.22 | [TK]D-Fender | Again, I'm not bitching. |
| 16:22.30 | [TK]D-Fender | I'm jsut saying that is is not "present" |
| 16:22.41 | igcewieling | It is not EOLd yet. |
| 16:22.52 | igcewieling | But my reasons don't matter to you so just drop it. |
| 16:24.08 | [TK]D-Fender | Secfix only |
| 16:24.18 | [TK]D-Fender | I'm not pushing you to upgrade |
| 16:24.27 | [TK]D-Fender | just don't call it "present" |
| 16:24.28 | [TK]D-Fender | :) |
| 16:24.42 | WIMPy | For many it's still the future. |
| 16:24.43 | [TK]D-Fender | #preoccupiedwith1985 |
| 16:24.50 | [TK]D-Fender | ^ |
| 16:25.05 | [TK]D-Fender | And some people still use Windows XP |
| 16:25.12 | [TK]D-Fender | time to get over that. |
| 16:25.18 | WIMPy | Indeed |
| 16:31.31 | Samot | I'm not sure how a version release over 15 months ago is the "future" |
| 16:31.59 | Samot | Sorry, 29 months ago. |
| 16:33.04 | WIMPy | It is if you're still using 1.2 or 1.4 as some people happily do. |
| 16:33.32 | Samot | And you would support that? |
| 16:33.33 | Samot | Now? |
| 16:33.54 | igcewieling | Even I admit people should not be using versions quite that old. |
| 16:33.59 | WIMPy | That's the point: They don't need support. |
| 16:34.06 | Samot | Until they do. |
| 16:34.11 | Samot | No one needs support |
| 16:34.22 | Samot | Until something actually breaks. |
| 16:34.41 | igcewieling | software does not just break. |
| 16:35.04 | WIMPy | Do you remember that good old "never change a running system"? It's never been more true than now. |
| 16:35.13 | igcewieling | exactly!@ |
| 16:35.19 | [TK]D-Fender | or the unduerlying hardware. Then they want to install onto a new OS... who doesn't support the old ZAPTEL. |
| 16:35.21 | [TK]D-Fender | Or such |
| 16:35.28 | [TK]D-Fender | maybe other libs run into conflict |
| 16:35.42 | WIMPy | In the last years I regretted so many upgrades. |
| 16:35.54 | Samot | I havent. |
| 16:37.06 | igcewieling | I used to be very proactive about upgrading and giving customers the latest features. Customers HATED it and I got bitten too many times with issues not seen in production. I'm not in this job to get yelled at by customers every day. Now, if the customer doesn't PAY for it, it doesn't get done. |
| 16:38.41 | igcewieling | I suppose I should switch to the Certified version of Asterisk 11 |
| 16:42.27 | Samot | I wasn't referring to dedicated customer boxes that they have setup to their own specs/configs. |
| 16:42.31 | [TK]D-Fender | Cert means nothing if you aren't paying Digium for support |
| 16:42.36 | [TK]D-Fender | it's the same as tthe gen release |
| 16:42.39 | Samot | I was talking about the actual servers running the infrastructure itself. |
| 16:42.46 | [TK]D-Fender | And is about to be fully disco'd |
| 16:42.52 | [TK]D-Fender | that doesn't get you longer support |
| 16:49.36 | *** join/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 16:49.37 | Samot | I can't count how many clients I've had that just deployed the "It's up, it's running, it's all good" mentality come back and bite them in the ass on their projects they wanted done. Not just with Asterisk, in general. |
| 16:50.18 | *** part/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 16:52.50 | Samot | 11:34:43 AMÂ <igcewieling>Â software does not just break. <-- No but it is released with bugs that can be exploited. Same bugs that are fixed with updated versions. |
| 16:53.33 | WIMPy | And new versions contain new bugs. |
| 16:53.38 | Samot | Yes. |
| 16:53.40 | igcewieling | Asterisk 11 still receives security updates. |
| 16:53.41 | Samot | They do. |
| 16:53.44 | Samot | For now. |
| 16:54.20 | Samot | My point is if you're on 1.8 of something and its 8 years old and the software is on 5.6 |
| 16:54.21 | WIMPy | Let's put it that way: For some people the purpose of any software seems to be the possibility to upgrade, while others are in to more productive uses. |
| 16:54.32 | Samot | And you get hit with a bug from 1.8 that brings you down... |
| 16:54.36 | Samot | Dumb. |
| 16:54.43 | Samot | Could have been totally avoided. |
| 16:54.57 | Samot | That's BS. |
| 16:55.33 | Samot | When I say "upgrade" I don't just mean update the current box. |
| 16:55.35 | WIMPy | Or you avoid all the new and unknown bugs. |
| 16:55.41 | Samot | You can't do that in all cases. |
| 16:55.47 | Samot | You have to do a migration to to the stuff. |
| 16:56.11 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-14.wall.uls.co.za) |
| 17:07.02 | *** join/#asterisk sekil (~sekil@cable-89-216-225-104.dynamic.sbb.rs) |
| 17:07.43 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 17:13.42 | *** join/#asterisk NirS (~Nir_S@141.226.192.98) |
| 17:14.27 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 17:28.19 | jkroon | is it possible to send a NOTIFY (check-sync) to a SIP endpoint from the dialplan? |
| 17:29.16 | Samot | Dialplan only executes when a call happens. |
| 17:29.24 | Samot | There are notify options for peers for that. |
| 17:29.27 | WIMPy | System() |
| 17:29.49 | WIMPy | You can decide yourself if that makes a yes or a no for you :-) |
| 17:30.00 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 17:30.31 | jkroon | Samot, yes, that's the idea, during a call I'd like to send a NOTIFY back to the calling extension to notify the phone that it needs to recheck it's config. |
| 17:30.58 | Samot | You want the phone to possible reboot when it gets a call? |
| 17:31.00 | jkroon | WIMPy, System() can work, but spawns pointless extra processes (which in this case might be acceptable come to think of it) |
| 17:31.05 | WIMPy | I think some phones ignore that while in use. |
| 17:31.10 | Samot | They will |
| 17:31.23 | Samot | Pretty much any decent phone will. |
| 17:31.27 | Samot | As will ATAs |
| 17:31.32 | jkroon | hmm, ok, think hot desk. |
| 17:31.48 | Samot | Dialplan only executes when a call happens |
| 17:31.54 | WIMPy | Why is that decent? It's not a reboot request. |
| 17:32.03 | Samot | If the profile is different it is. |
| 17:32.07 | jkroon | so when the call gets received from the phone, issue FUNC_ODBC things to update the config to link the MAC to the right extension, then request the phone to recheck config. |
| 17:32.09 | Samot | It has to reload the profile. |
| 17:32.28 | Samot | Phones have to reboot generally to apply config changes. |
| 17:32.44 | Samot | Especially when pulling profiles. They compare and then update, if needed. |
| 17:32.49 | WIMPy | Decent phones can do it without reboot. |
| 17:32.56 | Samot | Depends on the setting. |
| 17:32.58 | jkroon | ok, so issue Hangup() and then in the h context send the NOTIFY if required. |
| 17:33.36 | jkroon | System will have to do. |
| 17:33.37 | WIMPy | I guess you have to experiment a little with that one. |
| 17:33.40 | Samot | If the phone is doing something it will wait until its done to update |
| 17:33.56 | WIMPy | That would be ok. |
| 17:34.17 | Samot | Why does the phone need to change profiles? |
| 17:34.39 | WIMPy | You came up with profiles. |
| 17:34.52 | Samot | config = profile |
| 17:34.59 | Samot | 12:32:09 PMÂ <jkroon>Â so when the call gets received from the phone, issue FUNC_ODBC things to update the config to link the MAC to the right extension, then request the phone to recheck config. |
| 17:35.00 | Samot | ^^^^ |
| 17:35.11 | Samot | Why does it need to do this during a call? |
| 17:37.02 | jkroon | the idea is that all phones by default configure to some mac-based account, and that when the first call goes out the caller gets prompted for the extension (and secret/password) to then enable auto-generating the right configuration for the phone. |
| 17:37.24 | jkroon | so today I sit by desk A, tomorrow I sit by B. |
| 17:37.34 | Samot | OK |
| 17:37.41 | jkroon | obviously some log-off sequence needs to be implemented too :) |
| 17:37.49 | Samot | But what is the phone updating? |
| 17:38.05 | Samot | You can "hot desk" a phone without having to change it's profile. |
| 17:38.08 | jkroon | account settings, names, BLF monitoring lists etc ... |
| 17:38.13 | Samot | OK. |
| 17:38.28 | Samot | So you have to regenerate the config each time. |
| 17:38.43 | Samot | Because MAC based is how phones pull their configs to begin with. |
| 17:39.06 | jkroon | if there is a better way ... then please do point me at it. if it was only which "extension" was required keeping a map of extension/account in astdb would have been adequate. |
| 17:39.13 | sekil | that's usually done out of provisioning |
| 17:39.19 | sekil | via some xml script |
| 17:39.29 | Samot | Depends on the phone, but yes. |
| 17:39.34 | sekil | err xml based http script |
| 17:39.38 | Samot | Depends on the phone, but yes. |
| 17:39.40 | jkroon | oh? |
| 17:39.43 | Samot | Yes. |
| 17:39.43 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 17:39.49 | jkroon | documentation? |
| 17:39.54 | Samot | Depends on the phone. |
| 17:40.04 | sekil | jkroon: phones need to be xml enabled |
| 17:40.08 | WIMPy | o.O |
| 17:40.11 | Samot | No they do not. |
| 17:40.14 | sekil | jkroon: so go look for the API for the phone |
| 17:40.15 | Samot | Please stop saying that |
| 17:40.29 | Samot | No all phones use an XML based config |
| 17:40.38 | sekil | not the config |
| 17:40.44 | WIMPy | What do you guys think the phone should check when it receives that BOTIFY? |
| 17:40.49 | WIMPy | NOTIFY |
| 17:40.54 | Samot | check-syncy? |
| 17:40.56 | Samot | check-sync? |
| 17:41.03 | sekil | but xml app enabled |
| 17:41.07 | Samot | that means resync so check the provisioning server. |
| 17:41.19 | jkroon | WIMPy, because pretty much all phones I've worked with re-checks their config when it receives a NOTIFY with check-sync. |
| 17:41.57 | Samot | Yes, the check-sync means resync with the provisioning server... |
| 17:42.07 | Samot | Or resync it's profile. |
| 17:42.20 | Samot | config/profile whatever you want to call it. |
| 17:42.42 | WIMPy | jkroon: You could also set up mac based hints and change them on the server. Not sure if that makes more sense though. Probably depends on what or rather how many types of phones you want to use. |
| 17:43.08 | Samot | The real issue is.. |
| 17:43.14 | Samot | How are you changing those configs? |
| 17:43.30 | WIMPy | Why change? |
| 17:43.40 | Samot | He has to change the BLF monitorings |
| 17:43.44 | WIMPy | He only needs to links them to some MAC. |
| 17:43.48 | Samot | Right |
| 17:43.55 | Samot | He wants five users to use the same DEVICE |
| 17:43.58 | Samot | With the same MAC |
| 17:44.07 | Samot | With five different configurations. |
| 17:44.14 | Samot | Or even two |
| 17:44.30 | Samot | The phone looks for ITS MAC when pulling configs |
| 17:44.39 | WIMPy | Yes, that's the point. |
| 17:45.00 | Samot | So how can User A and User B both have their own configs for the phone.. |
| 17:45.07 | igcewieling | Seems to be a lot of work for something which should be able to be done from Asterisk. |
| 17:45.11 | Samot | And that phone pull based on MAC and get the right config? |
| 17:45.15 | WIMPy | By having their config. |
| 17:45.29 | igcewieling | mapping dialed number to device peer in the dialplan is a lot easier than changing phone configs all the time. |
| 17:45.30 | WIMPy | You just have to find out which users config to send to a given MAC. |
| 17:45.34 | Samot | Why would it be Asterisks job to update the phones display name? |
| 17:45.42 | jkroon | WIMPy, current need is for Yealink phones, which does seem to have some interesting hot desk options. busy reading up. |
| 17:45.43 | Samot | Or assign it's BLF monitoring? |
| 17:46.31 | jkroon | Samot, i use some rewrite rules to rewrite the path to point at a php script which then generates the configuration on the fly. |
| 17:46.50 | Samot | Yes, I'm not saying it can't be done. |
| 17:47.15 | Samot | Sending *45123 to Asterisk to trigger something not that hard. |
| 17:47.24 | jkroon | so the idea is to use func_odbc from dialplan to update the mapping in the SQL database, then request the phone to recheck config. |
| 17:48.10 | jkroon | my question really now is whether it's the right way - because looking at the yealink think I suspect being able to use a nice menu really is the better way to go than having users need to know what to dial to log out and in, for example. |
| 17:48.18 | sekil | jkroon: do you have acd server option? |
| 17:48.24 | jkroon | acd? |
| 17:48.56 | WIMPy | Don't they have some sort of xml browser you could use to do a login screen? |
| 17:49.01 | jkroon | yes, simple account/extension mapping is inadequate due to display name and BLF related things. |
| 17:49.15 | jkroon | WIMPy, that's what I'm trying to figure out now. |
| 17:49.19 | sekil | jkroon: acd login server |
| 17:49.20 | *** part/#asterisk igcewieling (~ewieling@162-236-85-155.lightspeed.moblal.sbcglobal.net) |
| 17:49.39 | WIMPy | But you could change BLF things server side as well. |
| 17:49.52 | Samot | OK.. |
| 17:49.58 | Samot | So first off.. |
| 17:50.02 | Samot | If you have Yealinks.. |
| 17:50.16 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 17:50.17 | Samot | Their Hot Desking feature requires the phone to be idle. |
| 17:50.27 | Samot | They have to enter a username and password.. |
| 17:50.31 | jkroon | Yealink + SNOM + Polycom (usage volumes in that order) |
| 17:51.00 | Samot | jkroon: OK so... |
| 17:51.07 | sekil | jkroon: with linksys/cisco you have acd server..you get a login password to type...and then it sends subscribe with event presence to server...that's where one can use a script to send check-sync with those values |
| 17:51.40 | sekil | jkroon: other way would be to build some custom http xml app to provide similar thing.. |
| 17:51.53 | sekil | jkroon: but normaly you would need to check api.. |
| 17:51.58 | Samot | jkroon: On the Asterisk side, you can dynamically assign users to devices.. |
| 17:52.05 | Samot | So that's one part of your "hot desking" |
| 17:52.55 | Samot | So users can dial a login/logout code with their user and pin, however you want... |
| 17:53.10 | jkroon | Samot, that's the easy part :) |
| 17:53.16 | Samot | Now.. |
| 17:53.20 | Samot | When they do all that.. |
| 17:53.28 | Samot | You add a System() or something. |
| 17:53.43 | Samot | Thats going to send them to your database to check their user/pin.. |
| 17:53.51 | Samot | Where you have their configs stored somewhere.. |
| 17:53.59 | jkroon | ... asterisk -rx "sip notify check-cfg endpoint" - for example. |
| 17:54.13 | Samot | that writes them to the provisioning location. |
| 17:54.30 | Samot | Then sends the check-sync when they end the call.. |
| 17:54.38 | WIMPy | If you do it via browser, you might even be able to trigger reconfiguration that way. |
| 17:54.52 | Samot | So the phone will reach out to the provisioning server and get the currently exported configs |
| 17:54.59 | Samot | You can do all this via a GUI too. |
| 17:55.18 | Samot | Or you can have them dial some digits and still execute the same script. |
| 17:55.48 | jkroon | ok, so i'm thinking that the xml mechanism would be nice. because the user just needs to enter his username/password on the phone UI. |
| 17:56.11 | jkroon | but i'm not sure if it'll pull all the configs that I need (BLF most notably) |
| 17:56.57 | WIMPy | You should be able to all options that way. |
| 17:58.18 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 17:59.01 | sekil | jkroon: bear in mind that change of key option like new blf usualy reboots phone |
| 17:59.16 | jkroon | sekil, that's not a problem. |
| 17:59.29 | jkroon | a reboot after logging in/out is not major. |
| 17:59.45 | WIMPy | The Snoms don't reboot. |
| 18:00.15 | sekil | hates snoms |
| 18:09.24 | *** join/#asterisk Inf0r (uid2810@gateway/web/irccloud.com/x-ldvvlqdibtzuzqqj) |
| 18:14.28 | *** join/#asterisk giesen (~ggiesen@2001:19f0:0:1019:5400:ff:fe25:bda6) |
| 18:21.22 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 18:38.27 | jkroon | WIMPy, Samot, sekil - the Yealink Hot Desk is adequate if the ONLY thing you need to change is the SIP credentials for the account. |
| 18:38.56 | jkroon | as soon as ANY other settings (eg, BLF) enters the fray you're reasonably stuffed unfortunately. |
| 18:39.53 | jkroon | I suspect similar will apply for others. not sure if the XML browser could possibly be used as an alternative, but for now I'm going with the voice prompt and reboot thing. |
| 18:40.01 | *** join/#asterisk igcewieling (~ewieling@162-236-85-155.lightspeed.moblal.sbcglobal.net) |
| 18:41.07 | WIMPy | If you don't build everything yourself... |
| 18:41.18 | jkroon | also, the hot desk option doesn't give me any protection on which extensions may be hot desked and which not (not that this is really much protection in the bigger scheme of things) |
| 18:41.18 | jkroon | even just pulling display name is problematic. |
| 18:41.42 | jkroon | so a user's address book won't even follow him/her, so if you've got personalized anything ... |
| 18:42.36 | WIMPy | has reverted to using the browser as phone book. |
| 18:44.18 | jkroon | so i'll look into the browser as an option to allow hot desking via that as alternative config option. |
| 18:49.06 | *** join/#asterisk gswain (uid91227@gateway/web/irccloud.com/x-ogfmlxqddjurfcuv) |
| 19:02.36 | jkroon | eesh, the more I think about this the more problems there are with the basic yealink mechanism - eg, the same extension on multiple phones (presumably one could force a log-out when a user logs in on a new phone). |
| 19:02.52 | jkroon | really basic things ... |
| 19:04.05 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 19:05.58 | sekil | jkroon: https://www.3cx.com/docs/yealink-hot-desking/ |
| 19:07.32 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 19:27.59 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 19:52.20 | *** join/#asterisk pdugas (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 20:03.47 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 20:06.21 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 20:09.13 | *** join/#asterisk sekil (~sekil@cable-89-216-225-104.dynamic.sbb.rs) |
| 20:38.41 | *** join/#asterisk linux4life (~linux4lif@cpe-24-209-136-148.wi.res.rr.com) |
| 20:39.19 | *** join/#asterisk sekil (~sekil@cable-89-216-225-104.dynamic.sbb.rs) |
| 20:39.54 | linux4life | hello all. how can I capture users sending invalid dial strings to the pbx and play an invalid message rather than getting: Call from '595' to extension '075498' rejected because extension not found in context |
| 20:40.09 | linux4life | I tried e, i, and _X. |
| 20:40.24 | linux4life | e and i don't work because it's not background or waitexten, |
| 20:40.26 | *** join/#asterisk lankanmon (~LKNnet@2607:fea8:d1f:fc17:f882:25ff:b942:d21b) |
| 20:40.53 | linux4life | and _X. doesn't work because it captures everything, even valid numbers that should allow for dialing out and dialing in.... :-( |
| 20:43.18 | [TK]D-Fender | It does work |
| 20:43.47 | [TK]D-Fender | Pattern matching works from most precise to least specific |
| 20:44.04 | linux4life | unless you're running realtime. ;-) |
| 20:44.07 | linux4life | which I'm doing. |
| 20:44.50 | linux4life | I have the specific number listed in the DB, but the _X. catches the call rather than following the dialplan listed in the DB for the number. ;-) |
| 20:48.09 | [TK]D-Fender | Shouldn't affect relatime |
| 20:50.44 | WIMPy | You can always prioritize extensions by using includes. |
| 20:51.30 | linux4life | ***** for WIMPy!!! |
| 20:51.41 | linux4life | that's what I just did and tested. working perfectly!!!! :-) |
| 20:52.30 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 20:53.15 | DivideBy0 | why does WIMPy get five asterisks? :) |
| 20:53.36 | WIMPy | 've got more than 5 running :-) |
| 20:53.49 | DivideBy0 | I think he wants to give you his five to run! |
| 20:55.02 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 20:55.39 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-tvoumdwnpyksrlkm) |
| 20:55.49 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 20:58.43 | linux4life | WIMPy get's 5 "stars" for listing the solution that worked for me.... ;-) |
| 21:01.24 | WIMPy | Five star accommodation? A five star menu? Or what is it? |
| 21:01.54 | linux4life | Five star accommodation |
| 21:02.10 | WIMPy | Where? How long? |
| 21:12.55 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 21:17.49 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 21:22.13 | *** join/#asterisk miralin (~Thunderbi@194.8.128.120) |
| 21:24.28 | *** join/#asterisk sekil (~sekil@cable-89-216-225-104.dynamic.sbb.rs) |
| 21:36.18 | *** join/#asterisk sekil (~sekil@cable-89-216-225-104.dynamic.sbb.rs) |
| 21:57.57 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 22:01.02 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 22:01.14 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 22:07.20 | *** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy) |
| 22:20.14 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 22:21.14 | *** join/#asterisk sekil (~sekil@cable-89-216-225-104.dynamic.sbb.rs) |
| 22:34.34 | *** join/#asterisk sekil (~sekil@cable-89-216-225-104.dynamic.sbb.rs) |
| 22:38.18 | *** join/#asterisk sekil (~sekil@cable-89-216-225-104.dynamic.sbb.rs) |
| 22:45.57 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 22:49.43 | *** join/#asterisk overyander (~Jeff@209.141.208.197) |
| 23:09.50 | *** join/#asterisk j0hnd0e (~Mutter@24.48.86.58) |
| 23:14.49 | *** join/#asterisk Y04NN_ (~y04nn@nayon.fr) |
| 23:19.59 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 23:24.20 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 23:33.19 | *** join/#asterisk miralin1 (~Thunderbi@194.8.128.54) |
| 23:47.29 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 23:59.42 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
