| 00:20.44 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 00:35.04 | *** join/#asterisk imcdona (imcdona@2001:470:e916:3:a03a:7230:d912:453b) |
| 00:49.03 | zopsi | Anyone in here familiar with DPMA and Alert-Infos? I can't seem to get my D70 to recognize PJSIP_HEADER entry for ringtone and change ringtone. |
| 00:53.37 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 01:00.15 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 01:15.55 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 01:15.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:20.31 | *** join/#asterisk TodWulff (~TodWulff@unaffiliated/todwulff) |
| 01:27.58 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 01:52.16 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 02:02.30 | *** join/#asterisk roramirez (~rodrigo@186-78-10-162.baf.movistar.cl) |
| 02:03.41 | roramirez | Hello, i have a estranger problem with local channels. I set NoCDR in both context i'm using but the calls is logging en cdr |
| 02:05.58 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:06.04 | roramirez | mi test dialplan and .call is http://pastebin.com/sWubkLMJ |
| 02:06.47 | roramirez | what i do wrong? |
| 02:14.57 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 02:31.40 | *** join/#asterisk CeBe (~CeBe@port-92-206-56-170.dynamic.qsc.de) |
| 03:08.02 | *** join/#asterisk bipolar (~bipolar@offsite.guru) |
| 03:12.39 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 03:36.18 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 03:52.26 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 03:57.11 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 04:05.21 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
| 04:43.51 | *** join/#asterisk stevePearPear (~stevePear@202.166.82.164) |
| 04:45.45 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 04:48.56 | *** join/#asterisk DraugTheWhopper (32f5e4b6@gateway/web/freenode/ip.50.245.228.182) |
| 04:49.38 | DraugTheWhopper | Anyone got time to help me diagnose an issue? |
| 04:53.11 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 04:53.55 | [TK]D-Fender | ~ask |
| 04:53.56 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 04:54.38 | DraugTheWhopper | gotta love a pat answer... :P |
| 04:55.07 | ChannelZ | Diagnosis: You have dysentery |
| 04:57.15 | [TK]D-Fender | If you asked that on WebMD you'd already be in stage 4 cancer... |
| 04:58.26 | coppice | I've had dysentery. It was bloody awful |
| 05:01.33 | [TK]D-Fender | fires off a rim-shot |
| 05:04.25 | DraugTheWhopper | Alright then, does anyone know why I might have issues with the g723 codec when I'm pretty sure I'm running ulaw everywhere. If I set my inbound route to an extension, it works. If I set to an IVR, it hacks and coughs and never gets anywhere. Pastebin http://pastebin.com/qDS90BSz FreePBX 12.0.1beta4, Asterisk 12.3.2 |
| 05:05.12 | [TK]D-Fender | * can't translate G.723 unless you have a rather expensive transcoder card |
| 05:05.36 | [TK]D-Fender | That debug also does not include any SIP leve debug |
| 05:06.17 | [TK]D-Fender | The call has also been accepted anonymously so it doesn't have peer-specific settings either |
| 05:06.22 | [TK]D-Fender | Overall not-great |
| 05:06.36 | [TK]D-Fender | [2014-06-24 00:36:50] WARNING[31747][C-00000039] channel.c: Unable to find a codec translation path from (g723) to (slin) |
| 05:06.42 | [TK]D-Fender | Can't translate as it says... |
| 05:06.47 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 05:06.50 | DraugTheWhopper | So should I increase logging levels and bring it back, or take it to the FreePBX folks? |
| 05:06.57 | [TK]D-Fender | change codecs |
| 05:07.58 | DraugTheWhopper | Thats my problem, I have no idea how to. My gateway and phones all have ulaw as the highest priority, so I don't see how 723 got into my system in the first place. |
| 05:08.39 | [TK]D-Fender | It's a question of WHOSE priority matters as well |
| 05:08.53 | [TK]D-Fender | And you allowed it at all which is a mistake |
| 05:09.03 | coppice | my priorities matter |
| 05:09.13 | tm1000 | [TK]D-Fender: also “beta” software |
| 05:09.45 | [TK]D-Fender | tm1000: One noose at a time! :p |
| 05:10.00 | coppice | beta than nothing |
| 05:10.28 | DraugTheWhopper | Eh. I think my "Standard: Asterisk 12.3.2 (2014/06/13)" should be sufficient. |
| 05:10.50 | tm1000 | DraugTheWhopper: But freepbx being beta doesnt mean anything to you? hmmm |
| 05:10.55 | DraugTheWhopper | Lemme see if I can clobber 723 on my phone and gateway. |
| 05:11.08 | [TK]D-Fender | DraugTheWhopper: Clobber it on your SERVER |
| 05:11.18 | [TK]D-Fender | Stop putting the responsibility on the remote ends |
| 05:11.25 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 05:19.58 | DraugTheWhopper | Epic facepalm. |
| 05:21.09 | DraugTheWhopper | Disabling it on the endpoints fixed it. Is it still terribly important to squash it on the server? |
| 05:21.41 | [TK]D-Fender | Do you want end service picking is and your allowing it to shoot you in the head? |
| 05:21.56 | [TK]D-Fender | Lesson: don't drive like a schmuck |
| 05:22.39 | DraugTheWhopper | Lesson learned. After all, this is the first SIP system I've created, and the second one I've ever touched. |
| 05:23.22 | *** join/#asterisk Qwell (north@pdpc/sponsor/digium/Qwell) |
| 05:23.23 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 05:38.57 | *** join/#asterisk Qwell (north@pdpc/sponsor/digium/Qwell) |
| 05:38.58 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 05:54.53 | *** join/#asterisk Qwell (north@pdpc/sponsor/digium/Qwell) |
| 05:54.53 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 05:57.33 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.82.71) |
| 05:58.54 | ruben23 | hi guys is it possible to dial and bridge calls with extension 100 to extension 1000 |
| 06:01.49 | [TK]D-Fender | ? |
| 06:03.39 | ruben23 | i mean i have asterisk with extension 1000 and asterisk 2 with extension 100, when i ring this 1000 - it will ring the asterisk2 100 |
| 06:03.57 | stian | you have two pbx's you mean? |
| 06:04.23 | [TK]D-Fender | Your dialplan will do whatever you confiigure it to do. |
| 06:04.28 | stian | ^^ |
| 06:04.58 | ruben23 | stian: yes - how do i create dialplan on this functions somehow |
| 06:05.02 | ruben23 | any idea |
| 06:05.14 | [TK]D-Fender | exten => 1000,1,DOSOMETHING |
| 06:05.55 | [TK]D-Fender | make a peer to your other server. DIAL IT |
| 06:06.10 | [TK]D-Fender | This is no different that dialing a # over an ITSP |
| 06:07.46 | ruben23 | [TK]D-Fender: the two asterisk is peer now already |
| 06:08.05 | [TK]D-Fender | So go dial it over |
| 06:08.41 | [TK]D-Fender | Dial(SIP/somepeer/thenumbertodial) |
| 06:10.09 | *** join/#asterisk Qwell (north@pdpc/sponsor/digium/Qwell) |
| 06:10.09 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 06:10.17 | *** join/#asterisk stevePearPear (~stevePear@202.166.82.164) |
| 06:10.28 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 06:24.32 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 06:28.06 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 06:32.59 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 06:43.51 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 06:46.20 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 06:53.08 | *** join/#asterisk DataWraith (~s1gny@37-4-2-201-dynip.superkabel.de) |
| 07:02.36 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-56-12.rn.hr.cox.net) |
| 07:05.47 | *** join/#asterisk kayfox (~kayfox@orca.zerda.net) |
| 07:07.42 | *** join/#asterisk jsjc (~Adium@90.174.2.18) |
| 07:10.53 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 07:11.11 | *** join/#asterisk Nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 07:37.11 | *** join/#asterisk BakaKuna (~Thunderbi@office.voys.nl) |
| 08:18.57 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 08:39.12 | *** join/#asterisk wonderworld (~ww@p5084A999.dip0.t-ipconnect.de) |
| 08:53.44 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:53.58 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 08:53.59 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 09:08.15 | *** join/#asterisk tgu (~tgu@host-212-68-194-46.brutele.be) |
| 09:20.37 | *** join/#asterisk danjenkins__ (~dan@nat/digium/x-jhurcifjgaooncdb) |
| 09:43.30 | *** join/#asterisk jetlag (~jetlag@pool-71-168-195-42.cmdnnj.east.verizon.net) |
| 09:45.46 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 09:46.35 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.82.71) |
| 09:49.05 | ruben23 | hi guys i have asterisk1 with extension 100 on it and wanted to call asterisk2 extension 702 - how do i create the internal dial planto recieve the call from asterisk2 locals somwho.. |
| 09:49.11 | ruben23 | any suggestions..? |
| 10:04.49 | *** join/#asterisk Elliott9 (95fe33e9@gateway/web/cgi-irc/kiwiirc.com/ip.149.254.51.233) |
| 10:06.32 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-rprwlbuiziwfwjsw) |
| 10:11.53 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 10:27.55 | Elliott9 | Hi Guys, hope you're all well, I'm trying to mute recordings with ari in Asterisk 12.2.0 . In the api you pass "recordingName" of the currently in progress recording. I've tried the filename with/without extension, channel id, caller and some other values, sadly all receiving is the "Resource not found" message. Anyone idea what I should be using |
| 10:27.56 | Elliott9 | for "recordingName"? :) |
| 10:56.22 | *** join/#asterisk go (~puck@unaffiliated/go) |
| 10:56.24 | go | hi everyone |
| 10:56.44 | go | does anyone have a working ConfBridge configuration with pins and admins? |
| 10:57.00 | go | or any RTFM? I'm reading http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge but I can't seem to figure out where the pin is set |
| 10:59.07 | Chainsaw | go: You store it externally. |
| 10:59.23 | Chainsaw | go: ConfBridge is very minimalist, compared to MeetMe. It's certainly not a drop-in replacement. |
| 11:00.00 | go | so basically I need to build up the whole dialplan and ConfBridge just connects users to the given room that I specify before it |
| 11:00.49 | Chainsaw | go: Yes, you have none of the luxuries of MeetMe. It will not deal with the PIN for you, that's dialplan logic now. |
| 11:06.06 | *** join/#asterisk ketas- (ketas@ketas6-sixxs.si.pri.ee) |
| 11:06.35 | go | great |
| 11:06.38 | go | digs himself in |
| 11:12.00 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:17.52 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:18.44 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 11:23.41 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:29.30 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:35.23 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:36.06 | *** join/#asterisk wonderworld (~ww@p5084A999.dip0.t-ipconnect.de) |
| 11:36.35 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 11:41.15 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:46.59 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:47.42 | *** join/#asterisk wonderworld (~ww@p5084A999.dip0.t-ipconnect.de) |
| 11:48.19 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 11:52.51 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:58.42 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 12:02.44 | file | moo |
| 12:04.33 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 12:08.45 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:10.23 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 12:11.29 | *** join/#asterisk udp` (~kvirc@bzq-79-177-103-80.red.bezeqint.net) |
| 12:11.43 | udp` | hi to all |
| 12:13.11 | ruben23 | hi guys how to dial a local extension using asterisk CLI> any idea..? |
| 12:13.48 | WIMPy | channel originate ... |
| 12:14.26 | ruben23 | WIMPy: what you mean..? |
| 12:15.14 | WIMPy | That is how you do it. |
| 12:15.32 | ruben23 | thats the command..? where do i put the extension..? |
| 12:16.09 | udp` | im trying to dial with .call files and getting this error on the log file |
| 12:16.09 | udp` | ERROR[1624][C-00000006] netsock2.c: getaddrinfo("Test1", "(null)", ...): Name or service not known |
| 12:16.14 | WIMPy | Probably in a local channel. |
| 12:16.21 | WIMPy | It will tel you how it works. |
| 12:16.26 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 12:17.35 | udp` | any ideas why is that? |
| 12:19.14 | WIMPy | Looks like Test1 does not exist. |
| 12:19.36 | udp` | i tried with various Channel |
| 12:19.37 | udp` | s |
| 12:19.54 | udp` | that i defined in the sip.conf file |
| 12:20.07 | udp` | i am probably missing something |
| 12:20.10 | *** join/#asterisk roramirez (~rodrigo@pc-28-175-160-190.cm.vtr.net) |
| 12:20.14 | WIMPy | That doesn't exist as a host name. |
| 12:20.53 | udp` | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 12:20.57 | udp` | im using this guide |
| 12:21.21 | udp` | and im trying this simple hello world.call |
| 12:21.30 | udp` | Channel: SIP/trunkname/18882223333 |
| 12:21.30 | udp` | Application: Playback |
| 12:21.30 | udp` | Data: hello-world |
| 12:21.51 | udp` | where trunkname i replaced with Test1 |
| 12:22.10 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 12:22.11 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 12:22.11 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 12:22.24 | WIMPy | And Test1 is either not a configured peer or it has the same hostname configured that isn't valid. |
| 12:22.30 | ruben23 | i tried this and nothing happens --> originate SIP/107@cloud extension s@from-internal |
| 12:22.37 | udp` | it configured as user |
| 12:22.43 | udp` | not peer |
| 12:23.13 | udp` | so i need to change it to peer ? |
| 12:23.33 | WIMPy | Usually you use freind. |
| 12:24.13 | *** join/#asterisk andy09usa (~andy@unaffiliated/andy09usa) |
| 12:24.13 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 12:27.50 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 12:27.50 | udp` | YES ! |
| 12:27.53 | udp` | it worked! |
| 12:27.56 | udp` | :D :D :D |
| 12:28.39 | udp` | what is the difference between a user and peer ? |
| 12:29.58 | ruben23 | anyone..? |
| 12:33.41 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 12:36.45 | *** join/#asterisk pii3 (~pii3@unaffiliated/pii3) |
| 12:39.40 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 12:40.38 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 12:45.19 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 12:47.17 | *** join/#asterisk iulhk (iulhk@182.189.114.217) |
| 12:51.02 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 12:56.50 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 13:02.12 | *** join/#asterisk shmzadmin (uid28588@gateway/web/irccloud.com/x-ipaxmfjwkyebqnrt) |
| 13:02.45 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 13:06.27 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 13:08.11 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 13:08.33 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 13:13.50 | *** join/#asterisk wonderworld (~ww@p5084A999.dip0.t-ipconnect.de) |
| 13:14.33 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 13:19.25 | udp` | http://pycall.readthedocs.org/en/v2.0/usage.html |
| 13:19.36 | udp` | im trying to run the hello world code from this page |
| 13:20.08 | udp` | but i dont get anything on the log files |
| 13:20.11 | udp` | any idea? |
| 13:20.32 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 13:22.31 | Katty | yes. |
| 13:22.42 | Katty | vanilla ice cream, root beer soda, and vanilla vodka. |
| 13:23.55 | udp` | i preffer whisky :) |
| 13:23.58 | udp` | ;) |
| 13:24.13 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:26.22 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 13:32.12 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 13:38.16 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 13:44.23 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 13:50.48 | *** join/#asterisk cw1972 (~cw1972@145.255.240.32) |
| 13:51.41 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 13:54.08 | udp` | anyone using pycalls here? |
| 13:54.52 | *** join/#asterisk nicknam1232 (5ad12e74@gateway/web/freenode/ip.90.209.46.116) |
| 13:56.53 | roramirez | Hello, i have a estranger problem with local channels. I set NoCDR in both context i'm using but the calls is logging en cdr. My test dialplan and .call is http://pastebin.com/sWubkLMJ |
| 14:04.42 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 14:06.51 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:07.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:07.53 | *** join/#asterisk BitEvil_ (~quassel@tor/regular/SpeedEvil) |
| 14:08.34 | ruben23 | hi guys ---the situation is - freepBX server - with extension 100 being dialer will pass calls to another asterisk Server peered already to extension 1000 |
| 14:08.57 | ruben23 | elastix and another asterisk server is locally peered |
| 14:09.42 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-gpwdcipllitigack) |
| 14:14.52 | jmls | jmls |
| 14:15.24 | TodWulff | hi jmls, u testing highlighting? ;) <-- should help |
| 14:18.12 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:18.12 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:25.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 14:33.30 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 14:41.27 | *** join/#asterisk LiuYan (~LiuYan@222.125.134.157) |
| 14:44.35 | *** join/#asterisk imcdona (~imcdona@75-151-101-98-Washington.hfc.comcastbusiness.net) |
| 14:46.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 14:51.33 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 14:51.35 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 14:53.06 | TodWulff | [TK]D-Fender, after further digging, it seems that the OBi202 was actually designed with proxying in mind: http://i.imgur.com/a0dlzFk.png http://i.imgur.com/fJpWMd6.png I might actually be able to implement a solution as considered and discusseed yesterday. |
| 14:54.29 | [TK]D-Fender | unclear as to what that EXT1 truly offers, but it looks like that might be the implication |
| 14:56.14 | TodWulff | nods. |
| 14:56.45 | [TK]D-Fender | can't hurt to go look... |
| 14:57.16 | TodWulff | Agreed. I will advise if i ever get it working as intended. Thanks for the dialog, regardless. I appreciate the time. |
| 14:58.01 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:58.16 | [TK]D-Fender | http://blog.obihai.com/2012/08/use-your-obi202-as-google-voice-gateway.html |
| 14:58.46 | [TK]D-Fender | So you set up som other profile and point it to your server... |
| 14:59.53 | TodWulff | yep. makes sense. replace the ip phone in that image with an asterisk implementation. |
| 15:02.59 | *** part/#asterisk LiuYan (~LiuYan@222.125.134.157) |
| 15:05.34 | ipengineer | Is there a way when setting monitor on a channel to have it not start recording unless the call is picked up? I would have thought the “b” option would have done that. What happens is if I set mixmonitor on an incoming call/channel and then dial a device, if that device doesnt answer and I send it to voicemail it records 1 second of dead air. https://gist.github.com/zconkle/cf52e6a65cd3968c6dcc |
| 15:06.26 | WIMPy | And what did you expect? |
| 15:10.13 | ipengineer | I was hoping that unless the device answered it would not start recording |
| 15:10.29 | ipengineer | Having a bunch of 1 second recordings just takes up space and clutters everything |
| 15:10.39 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 15:10.44 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 15:11.34 | WIMPy | Yes, that doesn't make sense. |
| 15:11.49 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-dpeqmcjvouaadqpp) |
| 15:13.13 | ipengineer | Is it because the incoming call is on chan_sip and I am dialing out to a device using PJSIP possibly |
| 15:13.40 | [TK]D-Fender | I think we should have an actual call to look at... |
| 15:14.44 | ipengineer | K. How would you like me to provide that? Just the CLI output? |
| 15:15.36 | [TK]D-Fender | with full debug. |
| 15:15.48 | [TK]D-Fender | shouldn't need core at first |
| 15:15.51 | [TK]D-Fender | but verbose + sip |
| 15:16.10 | ipengineer | k |
| 15:18.28 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:18.28 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:22.06 | *** join/#asterisk MarcoZink (~marcozink@187-178-254-98.dynamic.axtel.net) |
| 15:27.25 | ipengineer | [TK]D-Fender: WIMPy Here you go: https://gist.github.com/zconkle/03b60af18bbcd88de173 |
| 15:28.56 | [TK]D-Fender | <PROTECTED> |
| 15:29.13 | [TK]D-Fender | MixMonitor(filename.extension[,options[,command]]) |
| 15:29.21 | [TK]D-Fender | options is the THIRD parameter |
| 15:29.35 | [TK]D-Fender | you put in the SECOND |
| 15:29.42 | [TK]D-Fender | umm.. |
| 15:29.44 | [TK]D-Fender | hold that. |
| 15:29.48 | [TK]D-Fender | me checks his eyes.. |
| 15:30.02 | [TK]D-Fender | gah |
| 15:30.42 | [TK]D-Fender | strike that... |
| 15:30.42 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-keowdwwhlnqvuwzs) |
| 15:31.21 | ipengineer | K |
| 15:31.54 | [TK]D-Fender | it does seem to just kick off instantly there... |
| 15:34.44 | ipengineer | Yea.. The only thing I could come up with is using SIP and PJSIP together.. But I would still think asterisk would know when the call is bridged |
| 15:35.36 | [TK]D-Fender | it starts immeditealy after.. there is no bridge yet... |
| 15:35.45 | [TK]D-Fender | You haven't even gotten close to a dial or anything that should answer |
| 15:37.40 | ipengineer | The only other thing is I am on 12.0.0 but I didnt see any bugs that addressed this and I am assuming it prob hasnt been touched |
| 15:39.40 | [TK]D-Fender | "didn't see" isn't a a great reason not to update. |
| 15:39.46 | [TK]D-Fender | always just do it |
| 15:40.19 | ipengineer | No.. I know. I would love to upgrade but I am waiting on file to build out dialog-info support. There is no support for that in the later versions. There are a lot of other reasons I NEED to upgrade but I am stuck until that happens |
| 15:41.34 | wdoekes | are you saying there was support in 12.0.0 but it was dropped? |
| 15:42.30 | file | no, it's a patch I did but putnopvut rewrote the pubsub support afterwards before it got in |
| 15:42.45 | ipengineer | wdoekes: No. file had submitted on the reviewboard a fix for it in 12.0.0 and I manually merged and recompiled. The underlying architecture changed after that so there it is no longer compatible and needs to be rewritten. See: https://issues.asterisk.org/jira/browse/ASTERISK-21443 |
| 15:45.14 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:2f70:21b:63ff:fe31:8426) |
| 15:49.38 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-qbxpjjzebdrwdsit) |
| 15:49.38 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:55.11 | *** join/#asterisk RaMcHiP (~RaMcHiP@wsip-98-190-149-42.ph.ph.cox.net) |
| 16:05.42 | jeremy_laine | file: I have been experimenting with your DTLS-SRTP branch |
| 16:06.14 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 16:06.16 | jeremy_laine | file: with force_avp=yes and dtlsverify=fingerprint, things are working nicely between asterisk / chrome |
| 16:06.33 | file | works with Firefox now too |
| 16:07.04 | jeremy_laine | file: ah, need to do an svn update then, I was getting a OK on the negotiation but no sound |
| 16:08.17 | jeremy_laine | file: I stopped at r417184 |
| 16:08.45 | file | 417185 was a Firefox fix |
| 16:19.35 | jeremy_laine | file: ok, need to refresh my patch and rebuild |
| 16:21.39 | jeremy_laine | file: should you need to point someone to a patch to slap onto asterisk 11.10.2 for tests http://anonscm.debian.org/gitweb/?p=pkg-voip/asterisk.git;a=blob;f=debian/patches/dtls_srtp.patch;hb=refs/heads/dtls-srtp-patch |
| 16:35.36 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 16:40.58 | jeremy_laine | file: hm, still one way audio with Firefox (I hear on the PC side, but not on the remote side) |
| 16:41.19 | file | what version? (and I'm still working on the branch) |
| 16:41.28 | jeremy_laine | but it might be that Firefox is at fault, interacting with the mic |
| 16:41.42 | jeremy_laine | r417185 |
| 16:41.49 | file | I meant Firefox |
| 16:42.12 | jeremy_laine | ah sorry 30.0 on Linux |
| 16:42.53 | jeremy_laine | file: I do see RTP frames flowing if I enable rtp debug, so I'll run some other webrtc demos to rule out a local problem |
| 16:43.46 | file | if media flows, nothing else I can do ^_^ |
| 16:46.18 | [TK]D-Fender | The media must flow... |
| 16:48.52 | jeremy_laine | file: anyway, focusing on the positive side of things, back to my initial message: chrome works just dandy so thanks for the progress |
| 16:49.00 | file | the world goes on |
| 16:57.38 | file | jeremy_laine, do you see media coming in AND going out? |
| 16:59.31 | *** join/#asterisk riscphree (~riscphree@h198.16.28.71.dynamic.ip.windstream.net) |
| 17:00.21 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:01.52 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 17:03.17 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:09.19 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:11.44 | *** join/#asterisk jpoz_ (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 17:29.55 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 17:30.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 17:33.28 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:36.19 | *** join/#asterisk BakaKuna (~Thunderbi@82-169-251-128.ip.telfort.nl) |
| 17:45.19 | *** join/#asterisk danjenkins__ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:47.39 | *** join/#asterisk Xaviertoor (~Xaviertoo@177.99.205.154) |
| 17:49.29 | file | tacos! |
| 18:06.22 | *** join/#asterisk amizraa4 (~amizraa@gateway/tor-sasl/amizraa) |
| 18:24.56 | *** join/#asterisk pgrace (~pgrace@108.161.133.150) |
| 18:25.47 | pgrace | Hi! I'm trying to get a Cisco 7925 working with chan_skinny. I've got the phone registered and I can call the phone/the phone can call an extension on asterisk but in neither case is RTP audio transmitted from either side. Does anyone have experience with this phenomena? |
| 18:39.29 | *** join/#asterisk bkruse (~Adium@64.89.97.127) |
| 18:50.51 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 18:55.24 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 18:55.55 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:00.27 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:07.10 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 19:12.41 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 19:14.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 19:20.39 | *** join/#asterisk saint_ (~saint@209.222.15.231) |
| 19:23.28 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:25.19 | *** join/#asterisk g_r_eek (~g_r_eek@46-29-7.adsl.cyta.gr) |
| 19:25.25 | *** join/#asterisk pjensen00 (~per@ip-64-21-247-189.far.ideaone.net) |
| 19:30.13 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 19:33.17 | aross42 | I'm sorry to ask what seems like a simple question. I've created accounts where the extension + pw set from users.conf can be used to authenticate. I'd like to have an alphanumeric username + extension instead. I tried the username field but it doesn't seem to do what I expect. Any pointer in the right direction is appreciated/ |
| 19:35.19 | [TK]D-Fender | ~book |
| 19:35.19 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:35.27 | WIMPy | And what exactely is the question? |
| 19:35.44 | WIMPy | Asterisk doesn't care if usernames or extensions are numeric or not. |
| 19:36.03 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 19:37.30 | malcolmd | ...and users.conf should be run away from |
| 19:38.15 | Qwell | malcolmd: I'm quoting you on that. |
| 19:38.39 | WIMPy | ~users.conf |
| 19:38.40 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 19:39.19 | aross42 | What is a better mechanism to manage users then? |
| 19:39.24 | WIMPy | Reading that I wonder if I should switch to using users.conf instead of the other files. hrrrg. |
| 19:39.30 | aross42 | Definitely open minded to a better way. |
| 19:40.18 | WIMPy | Use the channels config (mainly sip.conf) and extensions.conf |
| 19:40.28 | malcolmd | Qwell: it's got's good grammar and stuff |
| 19:43.10 | aross42 | WIMPy: Currently our extensions/accounts are numeric. One SIP client one of the team is using only accepts alpha usernames. I'm learning how to specify both the extension & an alpha username. I'm sure this is embarrassingly easy. Maybe if there's a decent example I can look at somewhere? |
| 19:44.01 | aross42 | Google was pretty poor for signal vs. noise for this. Didn't learn much after reviewing a few pages worth. |
| 19:44.52 | aross42 | Again, I am very grateful for any help. And the response for something so simple. |
| 19:46.59 | *** join/#asterisk dw5304 (~dw5304@nas4free/dw5304) |
| 19:47.03 | WIMPy | As I said: Asterisk doesn't care. Just change the numeric name to a non-numeric one. |
| 19:47.23 | [TK]D-Fender | [ThIsis12345AlPhAnUmErIc] |
| 19:47.41 | dw5304 | after noon wondering what i should buy for a t1/pri card that works with asterisk |
| 19:48.04 | dw5304 | thinking i should get one with hardware noice canncelation |
| 19:48.14 | aross42 | OK, WIMPy [TK]D-Fender thanks I'll give that a go. Much appreciated. |
| 19:48.15 | [TK]D-Fender | dw5304: Digium and Sangoma are the 2 solid manufacturers of cards for * |
| 19:48.35 | WIMPy | dw5304: What protocoll? |
| 19:48.47 | dw5304 | pri to voip |
| 19:49.01 | Xaviertoor | r2 isdn? |
| 19:49.31 | dw5304 | let me see if i can locate some documentation on the existing unit |
| 19:49.32 | WIMPy | And IMHO HWEC only makes sense if you need some extremely low spec host. |
| 19:49.49 | WIMPy | r2 and isdn are exclusive. |
| 19:50.37 | *** join/#asterisk fauxalliance (gerald@irc.hacknl.org) |
| 19:50.50 | dw5304 | planning on replaceing a samsung officeserv MP20S |
| 19:52.12 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:52.12 | dw5304 | my bad 7200 i mean |
| 19:53.13 | WIMPy | What kind of line? |
| 19:53.20 | WIMPy | Or where? |
| 19:53.23 | WIMPy | T1? US? |
| 19:53.47 | dw5304 | t1 us |
| 19:54.12 | aross42 | OK, I'm good with the account now. Thanks again. I'll need to tweak my extensions.conf to adjust but I should be OK. Very grateful. I knew it was likely something very simple. |
| 19:54.30 | WIMPy | Ok, that makes easier. You need DAHDI anyway, so Digium or Sangoma are the best bets, I guess. |
| 19:54.53 | dw5304 | ah ok |
| 19:55.01 | dw5304 | kinda comming into this blind |
| 19:55.04 | Xaviertoor | Digium is the best in my opinion |
| 19:55.09 | dw5304 | lol so not sure what to expect |
| 19:55.57 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 19:57.55 | dw5304 | so http://www.ipphone-warehouse.com/Digium-TE133F-p/1te133f.htm |
| 19:57.58 | dw5304 | should work good? |
| 20:02.58 | Xaviertoor | http://digium.com/en/products/telephony-cards/digital/single-span |
| 20:03.49 | [TK]D-Fender | dw5304: that should do |
| 20:04.42 | dw5304 | ok thanks |
| 20:05.03 | WIMPy | Well, that depends on your definition of "good". |
| 20:05.51 | dw5304 | well i came here asking what i should buy so im playing the what do u guys thing |
| 20:05.57 | dw5304 | think* |
| 20:06.45 | [TK]D-Fender | dw5304: winpis just a little jilted for some particular shortcomings DAHDI has in general with regards to PRI handling. I doubt it's a model issue |
| 20:06.49 | WIMPy | In your place I'd go for Digium. |
| 20:06.59 | WIMPy | But I'd let the host do the EC if you need it. |
| 20:07.42 | WIMPy | Indeed. It's about DAHDI, not the hardware, but it's mainly not about the PRI handling. |
| 20:07.44 | dw5304 | its only 200 bucks |
| 20:07.55 | dw5304 | why does it matter |
| 20:08.21 | dw5304 | i mean wouldnt it be better on the hardware vs host? |
| 20:08.49 | WIMPy | If you do it on the host, you've got different algorithms to choose from. |
| 20:09.01 | dw5304 | ah ok |
| 20:09.23 | dw5304 | so say i have 24 lines with ec on host side of things what do i need in terms of a proc |
| 20:10.44 | WIMPy | IIRC, I managed about 16 channels on a PIII-1266. |
| 20:10.53 | dw5304 | o good why? |
| 20:10.55 | dw5304 | god* |
| 20:11.11 | WIMPy | Why what? |
| 20:11.17 | dw5304 | P3 |
| 20:11.24 | dw5304 | thats like sooooo old |
| 20:11.37 | WIMPy | Because that was to spare. |
| 20:11.39 | dw5304 | was thinking about using an i3 with 8gb of rm |
| 20:11.41 | dw5304 | ram |
| 20:11.54 | WIMPy | only used that for testing. |
| 20:12.39 | WIMPy | And testing on some ather lame hardware can be a pretty good idea. You see possible issues much faster. |
| 20:14.39 | Xaviertoor | Here in Brazil personal use without Clone Digium echo cancellation. |
| 20:14.39 | Xaviertoor | As you use Digium for $ 600 we have to buy $ 500 by false Digium and use Oslec. |
| 20:15.27 | Xaviertoor | And it works well |
| 20:16.22 | WIMPy | prefers hardware supported by Linux. |
| 20:16.37 | WIMPy | But Linux only supports DSS1. |
| 20:20.03 | WIMPy | Err, no. You can actually do R2, AFAIK. |
| 20:24.11 | *** join/#asterisk Sprocks (~Sprocks@BMTNON3746W-LP130-03-1242451514.dsl.bell.ca) |
| 20:32.19 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 20:51.21 | *** join/#asterisk shido6 (~shido6@98.234.178.27) |
| 20:52.44 | *** join/#asterisk kalz (~kalz@2602:fff6:f:1::d979:58e8) |
| 20:52.50 | *** join/#asterisk FrozenFire (~FrozenFir@pdpc/supporter/active/frozenfire) |
| 20:53.27 | FrozenFire | Is there a saner approach to polling information out of Asterisk for monitoring purposes, than trying to parse the output of the "show" commands available via asterisk -rx? |
| 20:53.39 | FrozenFire | We can't add any components, as SNMP would require. |
| 20:55.11 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:55.37 | rrittgarn | listen to AMI? |
| 20:55.40 | rrittgarn | or use AMI for polling? |
| 20:55.49 | rrittgarn | or AMI over HTTP... or ARI if you're using 12 |
| 20:56.32 | *** join/#asterisk shuggans (~shuggans@ucomnetworks-209-203-168-138.ucom.net) |
| 20:56.32 | FrozenFire | Ah, that could be good. I'll take a look at that. Thank you. |
| 20:56.43 | shuggans | anyone in the US (preferably close to kansas, but not a deal breaker) do any freepbx consulting work? |
| 20:58.18 | WIMPy | You should ask in #freepbx. It's not supported in here. |
| 20:59.56 | shuggans | Thanks |
| 21:13.12 | *** join/#asterisk bkruse (~Adium@172.56.20.227) |
| 21:16.25 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 21:23.21 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 21:25.46 | *** join/#asterisk TazzNZ (~pieter@office-nat1.ytechops.co.nz) |
| 21:28.24 | *** join/#asterisk kalz (~kalz@2602:fff6:f:1::d979:58e8) |
| 21:34.17 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 21:43.36 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 21:56.32 | *** join/#asterisk bkruse (~Adium@64.89.97.127) |
| 22:17.55 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:28.59 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 22:32.19 | *** join/#asterisk OmSai (quassel@2600:3c03::f03c:91ff:feae:5caf) |
| 22:36.53 | *** part/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 22:40.16 | *** join/#asterisk theron (~theron@173-29-74-236.client.mchsi.com) |
| 22:40.51 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 22:46.42 | *** join/#asterisk theron (~theron@173.252.71.189) |
| 22:55.37 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 23:07.23 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 23:08.50 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 23:23.31 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 23:33.59 | *** join/#asterisk kalz (~kalz@2602:fff6:f:1::d979:58e8) |
| 23:34.23 | *** join/#asterisk BakaKuna (~Thunderbi@82-169-251-128.ip.telfort.nl) |
| 23:35.13 | *** join/#asterisk beanie (~beanie@bgareth2.plus.com) |
| 23:36.45 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 23:43.34 | *** join/#asterisk zhando (~user@c-50-132-88-182.hsd1.wa.comcast.net) |
| 23:43.37 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 23:54.57 | *** join/#asterisk SALZH (~Administr@49.89.119.121) |
| 23:59.17 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
