| 00:53.30 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:53.30 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.6.0 (2015/10/09), 11.20.0 (2015/10/09); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:59.08 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 00:59.19 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:10.31 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 01:18.14 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 01:24.48 | *** join/#asterisk mlhess (~mlhess@drupal.org/user/102818/view) |
| 01:41.49 | snadge | [2015-12-17 22:51:12] NOTICE[21548] chan_iax2.c: Peer '09544614' is now UNREACHABLE! Time: 70 |
| 01:41.54 | snadge | what does the time value at the end mean? |
| 01:46.05 | WIMPy | The last qualify time before becomming unreachable. |
| 02:07.01 | snadge | but what kind of time is 70 ? |
| 02:07.08 | snadge | 70 ms? 70 past 2 |
| 02:11.52 | *** join/#asterisk jasonwert (~wert@71.89.137.28) |
| 02:18.57 | *** join/#asterisk superscrat (~asanders@173.21.89.217) |
| 02:42.09 | *** part/#asterisk superscrat (~asanders@173.21.89.217) |
| 02:59.52 | ChannelZ | I means 70 hookers have died since the last time it was able to reach the peer |
| 03:20.43 | *** join/#asterisk iq (~iq@pool-173-74-16-217.dllstx.fios.verizon.net) |
| 04:51.12 | *** join/#asterisk Lope (~Lope@192.228.155.122) |
| 05:23.41 | *** join/#asterisk jploh (~textual@122.2.37.42) |
| 05:31.10 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 05:44.20 | Lope | Is there a way to determine how long the Dial()'ed extension rang for before it was picked up? |
| 05:54.19 | [TK]D-Fender | CDR |
| 05:54.49 | [TK]D-Fender | Or check the time before, and compare it upon answer using one of the evident options |
| 05:56.27 | snadge | so tonight i want to bench test asterisk 13.. we have some pretty complex AGI and specific cdr requirements which has boned us |
| 05:56.47 | Lope | What is CDR? |
| 05:57.23 | snadge | asterisk 13 has deprecated some compatibility options.. and its been difficult for me to figure out how to update my config |
| 05:57.42 | Lope | As far as I understand it, the statement after Dial() only get's executed at the end of the call? How can I run a statement when the extension picks up? |
| 06:00.38 | Lope | I'm thinking to use the cURL function to send requests before the extension is Dialed and after the Dial command ends. |
| 06:00.57 | Lope | (to my app which will do something with the data) |
| 06:02.13 | [TK]D-Fender | "core show application dial" <- |
| 06:02.23 | [TK]D-Fender | Always read the app's instrucitons |
| 06:02.36 | Lope | I found this https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification#Asterisk12CDRSpecification-CDROverview |
| 06:05.09 | Lope | The only place I could find the string "CDR" in the string resulting from "core show application dial" was 'C: Reset the call detail record (CDR) for this call." |
| 06:08.18 | Lope | okay, I see it's a function called CDR |
| 06:12.38 | *** join/#asterisk jploh (~textual@122.2.37.42) |
| 06:13.45 | *** join/#asterisk Aamit (~Amit@115.254.72.179) |
| 06:14.08 | Lope | thanks very much, I think I've got something that will work, will test a little later. NoOp(${CDR(start)} - ${CDR(answertime)} - ${CDR(end)}) |
| 06:14.41 | Lope | I mean answer... not answertime. |
| 06:21.13 | [TK]D-Fender | progress |
| 06:21.21 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 06:21.21 | [TK]D-Fender | keep running with that |
| 06:21.28 | [TK]D-Fender | time for bed again, back tomorrow |
| 06:25.38 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 06:53.15 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 06:55.02 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:58.20 | *** join/#asterisk goddva (~goddva@77.40.154.242) |
| 06:58.37 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 07:21.37 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 07:38.30 | *** join/#asterisk jonno11 (~Jon@cpc1-walt12-2-0-cust582.13-2.cable.virginm.net) |
| 07:56.49 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:15.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:23.35 | *** join/#asterisk Tim_Toady (~fuzzy@snf-33276.vm.okeanos.grnet.gr) |
| 08:55.48 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 09:23.27 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 09:25.42 | *** join/#asterisk Guest76385 (danjenkins@gateway/shell/firrre/x-ysqfzrerxczibght) |
| 09:30.33 | Guest76385 | x |
| 09:42.35 | *** join/#asterisk wonderworld (~ww@46.189.28.48) |
| 09:59.54 | *** join/#asterisk defsdoor (~andy@cpc73037-sutt4-2-0-cust62.19-1.cable.virginm.net) |
| 10:06.34 | *** join/#asterisk Draecos (~Draecos@203-121-194-11.e-wire.net.au) |
| 10:44.41 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 10:56.12 | *** join/#asterisk keithf (~keithf@ool-2f151dc2.static.optonline.net) |
| 11:01.19 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 11:04.25 | *** join/#asterisk troyt (~troyt@2601:681:4600:3381:44dd:acff:fe85:9c8e) |
| 11:05.27 | *** join/#asterisk C1ph3r5 (~C1ph3r@179.179.128.69) |
| 11:16.07 | *** join/#asterisk troyt (~troyt@2601:681:4600:3381:44dd:acff:fe85:9c8e) |
| 11:17.03 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:64bf:97ac:da5f:aad2) |
| 11:30.02 | *** join/#asterisk Badbit (~badbit@unaffiliated/badbit) |
| 11:30.27 | Badbit | Anyone here familier with the pjsip libary? |
| 11:32.34 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 12:02.44 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 12:15.01 | file | Yes. Whats your question? |
| 12:18.09 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 12:20.49 | *** join/#asterisk Draecos (~Draecos@203-121-194-11.e-wire.net.au) |
| 12:21.19 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 12:26.17 | *** join/#asterisk ModFather (~ModFather@unaffiliated/modfather) |
| 12:36.30 | *** join/#asterisk flujan (~flujan@200.160.115.22) |
| 12:48.05 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 12:49.53 | Chainsaw | file: It could just be a survey I suppose. |
| 12:50.05 | file | Perhaps. |
| 12:52.35 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 12:52.54 | file | There was a thread on the PJSIP mailing list asking a similar thing thus it caught my eye. |
| 12:56.24 | *** join/#asterisk aurs (~aurs@51.84-48-191.nextgentel.com) |
| 12:59.22 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:10.57 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 13:26.34 | Badbit | file: Is there a way to pass userdata through to a pjsip event or is a linked list the only way? |
| 13:27.08 | Badbit | Chainsaw: Tony, I'm not a survey :\ |
| 13:27.13 | file | Don't know off the top of my head. We don't use it like that. |
| 13:28.30 | Badbit | Hmm, okay. |
| 13:32.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:32.24 | *** join/#asterisk O47m341 (~Suzeanne@75-103-145-152.ccrtc.com) |
| 13:45.32 | dan_j | I note that CDR(accountcode) has been depreciated and replaced with CHANNEL(accountcode). Is that inherited by child channels when the dialplan does a Dial ? |
| 13:46.30 | dan_j | ie. does it function in the same way as CDR(accountcode) ? |
| 13:57.52 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 13:58.12 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-ydayoqhjgoxqfnrb) |
| 13:58.12 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:58.57 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 14:07.06 | Chainsaw | Badbit: I know. It was just odd to have 1 question with no follow-up :) |
| 14:09.10 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:11.33 | Badbit | Chainsaw: I was afk, sorry |
| 14:12.20 | *** join/#asterisk jasonwert (~wert@75-134-81-98.static.aldl.mi.charter.com) |
| 14:12.34 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.124.43) |
| 14:19.32 | *** join/#asterisk acidfoo (45467202@gateway/web/freenode/ip.69.70.114.2) |
| 14:19.38 | *** join/#asterisk C1ph3r5 (~C1ph3r@179.179.128.69) |
| 14:20.02 | acidfoo | does the allow= order is really respected by asterisk ? |
| 14:20.06 | acidfoo | for codecs |
| 14:20.32 | [TK]D-Fender | Yes |
| 14:20.43 | [TK]D-Fender | When * is the side that gets to choose its preference |
| 14:24.01 | acidfoo | ok thank you |
| 14:24.27 | acidfoo | I'm wondering if I should put allow=opus above all others |
| 14:25.18 | [TK]D-Fender | Has * finally added full transcode support for it? Last I heard it was only passthrough |
| 14:26.45 | acidfoo | I used a patch I found online |
| 14:27.22 | acidfoo | so does the best practice is to use WB codec first, and then the others next ? |
| 14:27.24 | [TK]D-Fender | For transcode or passthrough? |
| 14:27.45 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.124.43) |
| 14:28.55 | acidfoo | for the list in sip.conf allow= etc, |
| 14:29.13 | acidfoo | so depending on what the phones support that might passthrough or transcode I guess |
| 14:29.43 | [TK]D-Fender | This is not a guess. |
| 14:29.47 | [TK]D-Fender | What does that patch SAY? |
| 14:30.08 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 14:31.00 | acidfoo | well if I ask a more general question without speaking about opus |
| 14:31.06 | acidfoo | should I put let say g729 first |
| 14:34.51 | [TK]D-Fender | Why would you? |
| 14:36.07 | acidfoo | so you have the best codec first - what I'm trying to grasp here is what are the pro and cons of the order you set your codecs... from wide band to low band or low band to wide band etc |
| 14:36.23 | acidfoo | what drives the choice of the order people set their codecs |
| 14:37.04 | [TK]D-Fender | Their priorities obviously |
| 14:41.12 | *** join/#asterisk flujan (~flujan@200.160.115.22) |
| 14:44.59 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 14:56.45 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 14:59.41 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 15:06.13 | *** join/#asterisk kharwell (kharwell@nat/digium/x-mycysefngynaiqxm) |
| 15:12.39 | *** join/#asterisk zafu (~pif@84-73-105-33.dclient.hispeed.ch) |
| 15:13.10 | zafu | hi, I'm having a long delay between deleting a voice message and disappearance of notification on the (polycom) phone, any idea? |
| 15:13.17 | *** join/#asterisk flujan (~flujan@200.160.115.22) |
| 15:13.18 | zafu | like 15min |
| 15:19.55 | newtonr | Odd. I'd watch the SIP messages to see what gets sent to Asterisk when you delete the message and how Asterisk responds. I'd also see what happens on the filesystem. |
| 15:20.22 | newtonr | That is, see if the voicemails get deleted immediately. |
| 15:24.57 | zafu | thanks |
| 15:29.30 | newtonr | Is your name zafu as in zafu and zabuton? |
| 15:30.00 | zafu | unknown reference :) |
| 15:30.11 | newtonr | Cool, probably not then :) |
| 15:30.20 | zafu | it's zafu as in the meditation cushion |
| 15:30.34 | newtonr | Oh, yeah it is then, the zabuton is the pad that goes under the zafu |
| 15:30.42 | zafu | ah right |
| 15:30.53 | zafu | forgot about that one :) |
| 15:31.23 | zafu | do you meditate? |
| 15:31.46 | newtonr | Sure do, going on nearly three years of daily practice |
| 15:32.22 | zafu | that's great, I wish I was that regular |
| 15:34.09 | newtonr | Yeah it is sort of like going to the gym or working out in general, takes a particular persistence and sometimes just time to build a habit |
| 15:35.14 | zafu | I keep a daily asana practice though, and usually sit still for a little at the end |
| 15:35.59 | newtonr | that is excellent - so good for you. exercise is wonderful |
| 15:36.31 | newtonr | i think it is a must when you are sitting at computers for many hours for work and play. |
| 15:36.50 | zafu | oh yes, |
| 15:38.56 | *** join/#asterisk tompaw (U2FsdGVkX1@tompaw.xxx) |
| 15:40.16 | newtonr | now I will continue to meditate on working out a new Asterisk sounds release |
| 15:40.41 | *** join/#asterisk thothcastel (~chatzilla@213.146.163.66) |
| 15:40.55 | thothcastel | Does Switchvox support hunt groups?? |
| 15:40.58 | zafu | ooomm |
| 15:41.10 | thothcastel | or how to link a did to a group of extensions?? |
| 15:43.00 | newtonr | thothcastel, Switchvox isn't supported in here. Most people here are using straight up Asterisk. You can checkout the answers forum or the knowledge base. There is also a Linkedin group |
| 15:43.03 | newtonr | http://support.digium.com/Answers#!/feedtype=RECENT_REPLY&dc=All&criteria=ALLQUESTIONS |
| 15:43.10 | newtonr | https://www.digium.com/support/switchvox |
| 15:45.35 | thothcastel | thanks newtonr |
| 15:47.59 | *** join/#asterisk jeffspeff (~jeff.clay@12.49.160.131) |
| 15:49.24 | *** join/#asterisk Docfxit (~Docfxit@108-201-143-79.lightspeed.mtryca.sbcglobal.net) |
| 15:51.04 | jeffspeff | when defining variables in realtime sip using the "setvar" field, should multiple variables be comma separated or semi-colon separated? should i put anything in quotes to make the variable values literal (just in case the var contains a comma or something) ? |
| 16:02.02 | *** join/#asterisk KNERD (~KNERD@netservisity.com) |
| 16:10.20 | *** join/#asterisk c0rnoTa (~c0rnoTa@91.221.232.65) |
| 16:10.23 | *** part/#asterisk c0rnoTa (~c0rnoTa@91.221.232.65) |
| 16:15.51 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-rbvzzgmjguwarjdt) |
| 16:16.57 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 16:22.06 | *** part/#asterisk lunaphyte (~lunaphyte@unaffiliated/lunaphyte) |
| 16:22.41 | *** join/#asterisk ruied (~ruied@125.44.62.94.rev.vodafone.pt) |
| 16:35.07 | *** join/#asterisk aness (~aness@2a02:fe0:c310:4830:b5cd:eacf:bbd6:c738) |
| 16:37.13 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 16:43.53 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 16:45.03 | acidfoo | is it possible to "reload" res_pgsql.conf ? |
| 16:49.32 | jeffspeff | module reload res_config_pgsql.so |
| 16:51.46 | *** join/#asterisk jhester (jhester@nat/digium/x-alltdfeqowfkypgv) |
| 16:54.41 | jeffspeff | when defining variables in realtime sip using the "setvar" field, should multiple variables be comma separated or semi-colon separated? should i put anything in quotes to make the variable values literal (just in case the var contains a comma or something) ? |
| 17:01.51 | *** part/#asterisk flujan (~flujan@200.160.115.22) |
| 17:20.00 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 17:38.32 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 17:48.23 | *** join/#asterisk italorossi (~Adium@177.65.201.177) |
| 17:58.40 | *** join/#asterisk Tristan-Speccy (tristan@213.163.67.18) |
| 18:22.52 | *** join/#asterisk boson (~boson@cpe-24-29-241-97.neo.res.rr.com) |
| 18:24.02 | *** join/#asterisk ModFather (~ModFather@unaffiliated/modfather) |
| 18:32.54 | *** join/#asterisk corretico (~luis@181.193.0.162) |
| 18:34.35 | jeffspeff | when defining variables in realtime sip using the "setvar" field, should multiple variables be comma separated or semi-colon separated? should i put anything in quotes to make the variable values literal (just in case the var contains a comma or something) ? |
| 18:41.26 | [TK]D-Fender | Still asking the same question after an hour & half? |
| 18:41.33 | [TK]D-Fender | 2 second Google search answers this... |
| 18:41.38 | [TK]D-Fender | https://www.google.ca/#q=asterisk+realtime+setvar |
| 18:41.50 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 18:42.09 | *** join/#asterisk ruied (~ruied@125.44.62.94.rev.vodafone.pt) |
| 18:42.17 | [TK]D-Fender | http://lists.digium.com/pipermail/asterisk-users/2006-January/133678.html |
| 18:45.00 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 18:48.47 | drmessano | Also, repeating a question when your last is still not scrolled off the screen is extremely bad form |
| 18:49.33 | [TK]D-Fender | Or one you could prove in all of 2 minutes. Add a peer. Set 2 vars. Place call. Call Dumpchan. The End. |
| 18:49.47 | [TK]D-Fender | Or just set 2 on an existing peer |
| 19:02.04 | *** join/#asterisk overyander (~jeff.clay@12.49.160.131) |
| 19:02.52 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 19:04.05 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 19:18.54 | *** part/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 19:20.11 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 19:23.39 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 19:47.19 | acidfoo | there is the module snmp, is there other module to export information about the internal of asterisk ? |
| 19:54.34 | drmessano | AMI, ARI.. |
| 19:54.58 | drmessano | Asterisk has API out the Wazoo |
| 19:55.36 | acidfoo | wazoo ? :P |
| 19:56.30 | drmessano | If you need import, export, report, or deport, it can do it |
| 19:58.33 | *** join/#asterisk Echo6 (~Echo6@64.136.247.50) |
| 20:00.23 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 20:01.32 | newtonr | acidfoo, a variety of interfaces are described under this page https://wiki.asterisk.org/wiki/display/AST/Interfaces |
| 20:06.06 | acidfoo | thank you |
| 20:06.24 | acidfoo | i'll check that to see if you could use something else than snmp |
| 20:14.07 | *** join/#asterisk Mango45 (~Mango45@142.59.247.66) |
| 20:15.17 | Mango45 | Is there a way I can find out the Caller ID of other calls in progress, using the dial plan? I'm getting a minor DoS of sorts, in which someone calls me a dozen times in the same minute to use up all my channels. |
| 20:15.37 | Mango45 | I was hoping to do someting like "if a call from this number is already in progress, do CONGESTION". |
| 20:16.11 | [TK]D-Fender | Not directly, but you could use things "core show functions like GROUP" to keep count |
| 20:16.23 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 20:16.40 | [TK]D-Fender | This is technically setting a value first rather than looking at other channels on any individual basis |
| 20:16.53 | Mango45 | That should work; thanks. |
| 20:22.56 | *** join/#asterisk C1ph3r (~C1ph3r@177.41.70.124.dynamic.adsl.gvt.net.br) |
| 20:25.18 | *** part/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 20:26.55 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 20:35.28 | *** join/#asterisk jasonwert (~wert@71.89.137.28) |
| 20:37.12 | *** join/#asterisk sekil (~Ognjen@82.117.222.227) |
| 20:42.50 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 20:50.11 | *** join/#asterisk jeffspeff (~jeff.clay@12.49.160.131) |
| 21:10.06 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 21:21.52 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 21:27.52 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 21:27.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:28.55 | *** topic/#asterisk by putnopvut -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.7.0 (2016/01/15), 11.21.0 (2016/01/15); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 21:31.12 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 22:04.17 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 22:06.44 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:07.21 | *** join/#asterisk jpastore (~jpastore@50-200-51-210-static.hfc.comcastbusiness.net) |
| 22:08.41 | jpastore | hi, is there some kind of guidelines to follow when spec'ing out hardware for a server? what attributes shoudl I focus on for a * server? |
| 22:12.37 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 22:14.31 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 22:15.17 | [TK]D-Fender | jpastore, what are you expecting to support on it? |
| 22:18.34 | jpastore | well, that's a good question. we're a vici shop, I'm new to both vici and asterisk, vici appears to be just php scripts and a mysql db which is offloaded to a separate server in the cluster, the rest are all agent and dialer servers. so a lot of outbound calling |
| 22:19.48 | DivideBy0 | jpastore: there's a bunch of benchmarks from astricon here https://www.youtube.com/watch?v=tYKWE8EYD9Q&list=PLighc-2vlRgT3sWvW0RL7qyD2THenFIfZ&index=47 |
| 22:20.05 | jpastore | DivideBy0, thanks! I'll review =) |
| 22:20.24 | DivideBy0 | and Matt, the author of vici has some specs here in his presentation: https://www.youtube.com/watch?v=tw5J2Ug_7XI&list=PLighc-2vlRgT3sWvW0RL7qyD2THenFIfZ&index=44 |
| 22:30.18 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 22:32.33 | klow | newbie question - would asterisk by default only open the asterisk.conf file, and any other configs need to be included? or does it open other config files automatically, such as modules.conf, logger.con , sip.conf etc |
| 22:33.23 | klow | i guess it must be loading other files automatically since I haven't included any and i see log entries related to other config files |
| 22:33.40 | klow | is there a list somewhere of all the files it would automatically open , or does that depend on what modules are loaded? |
| 22:33.50 | klow | trying to sort out by chan_sip is failing to load |
| 22:36.07 | newtonr | klow, modules.conf! |
| 22:36.27 | newtonr | klow, https://wiki.asterisk.org/wiki/display/AST/Troubleshooting+Asterisk+Module+Loading |
| 22:36.43 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Configuring+the+Asterisk+Module+Loader |
| 22:36.55 | newtonr | Configuration files are read for the modules that are loaded |
| 22:37.04 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 22:37.07 | xochilpili | hi there! |
| 22:37.29 | xochilpili | im having troubles with odbc real time with asterisk and mysql |
| 22:37.37 | xochilpili | http://pastebin.com/Fq8qFeuX << this is all my configuration |
| 22:37.41 | xochilpili | anyhand ? |
| 22:37.43 | xochilpili | please? |
| 22:38.13 | newtonr | xochilpili, You should describe the problem you are having :) |
| 22:38.24 | xochilpili | No such connection 'asterisk' in the 'first' section of cdr_adaptive_odbc.conf. < |
| 22:38.40 | xochilpili | i want to users from mysql table register into asterisk |
| 22:38.41 | xochilpili | :D |
| 22:38.48 | xochilpili | but odbc show is empty |
| 22:38.53 | xochilpili | also sip show peers |
| 22:39.16 | newtonr | i think connection should be connection=asterisk-kamailio |
| 22:39.27 | xochilpili | i did that too |
| 22:39.30 | xochilpili | but the same result |
| 22:39.32 | [TK]D-Fender | dns => asterisk-kamailio <- FAIL |
| 22:39.58 | xochilpili | [TK]D-Fender, hey there! |
| 22:40.03 | xochilpili | how are you been? |
| 22:40.11 | xochilpili | why fail because of the dash? |
| 22:41.08 | [TK]D-Fender | because that shouldn't be "dns" and you aren't paying attention to the most basic and obvious keywords |
| 22:41.16 | *** join/#asterisk dimitry7 (~dimitry7@gate.aaamerica.com.mx) |
| 22:41.59 | xochilpili | dns?? |
| 22:42.08 | xochilpili | domain ? like localhost? |
| 22:42.25 | [TK]D-Fender | facepalms |
| 22:42.33 | [TK]D-Fender | DSN <--------------------------------------------- |
| 22:42.40 | [TK]D-Fender | Data Source Name |
| 22:43.00 | [TK]D-Fender | Source/Store |
| 22:43.03 | xochilpili | but data source name is asterisk-kamailio in odbc.ini <<<<< |
| 22:43.11 | [TK]D-Fender | no |
| 22:43.39 | [TK]D-Fender | dns => asterisk-kamailio <- FAIL |
| 22:44.25 | newtonr | xochilpili, he is saying it should be "dsn => asterisk-kamailio" |
| 22:45.55 | newtonr | also the connection in cdr_adaptive_odbc.conf needs to be set to the name in res_odbc.conf which is "asterisk-kamailio" not "asterisk". Though what TK is pointing out is the primary issue |
| 22:46.22 | [TK]D-Fender | should start using people in here as masonry.... |
| 22:46.28 | [TK]D-Fender | :| |
| 22:47.17 | [TK]D-Fender | Parameter names are important. |
| 22:47.54 | [TK]D-Fender | "My value is right" means nothing when you set the wrong parameter name |
| 22:48.20 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 22:49.45 | [TK]D-Fender | allowed=ulaw |
| 22:49.50 | [TK]D-Fender | I set it to ulaw!!!!! |
| 22:57.35 | xochilpili | i dont get it |
| 22:57.43 | xochilpili | i have change asterisk-kamailio but still nothing |
| 22:58.07 | xochilpili | i have change in cdr_adaptive.conf the name of the section as asterisk |
| 22:58.18 | xochilpili | which is the same in res_odbc.conf |
| 22:58.40 | newtonr | xochilpili, and you changed "dns" to "dsn" ? |
| 22:59.14 | klow | anyone know what conditions might cause this error? DEBUG[29923] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. I have my tlsbindaddr set to 0.0.0.0 , have also tried the actual IP |
| 22:59.17 | xochilpili | ah fuck |
| 22:59.21 | xochilpili | sorry... |
| 22:59.31 | xochilpili | [TK]D-Fender, now i get it ! |
| 22:59.51 | xochilpili | newtonr, now odbc show is connected :D |
| 22:59.52 | [TK]D-Fender | MASONRY |
| 22:59.53 | newtonr | Data Source Name, D S N :) |
| 23:00.43 | *** join/#asterisk Lope (~Lope@192.228.155.122) |
| 23:00.49 | xochilpili | :D |
| 23:06.00 | klow | does TLS / SRTP work in chan_pjsip ? |
| 23:06.27 | [TK]D-Fender | https://www.google.ca/#q=TLS+%2F+SRTP+work+in+chan_pjsip |
| 23:06.29 | [TK]D-Fender | ^ |
| 23:38.28 | *** join/#asterisk war9407 (war@static-72-73-18-14.clppva.fios.verizon.net) |
| 23:45.01 | *** join/#asterisk Qwell (~north@asterisk/developer/Qwell) |
| 23:45.01 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 23:57.45 | *** part/#asterisk kharwell (kharwell@nat/digium/x-mycysefngynaiqxm) |