| 00:01.28 | *** join/#asterisk bmurt (~brendan@64-121-3-32.c3-0.upd-ubr2.trpr-upd.pa.cable.rcn.com) |
| 00:06.25 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 00:09.36 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 00:14.50 | *** join/#asterisk ZerOlegend (~ZerOlegen@184-158-70-162.dyn.centurytel.net) |
| 00:37.26 | *** join/#asterisk MrFidget (~MrFidget@r49-2-11-73.cpe.vividwireless.net.au) |
| 00:38.04 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 01:00.49 | *** join/#asterisk vader- (~Adium@pool-173-49-160-70.phlapa.fios.verizon.net) |
| 01:09.39 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 01:11.07 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 01:18.07 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 01:32.07 | *** join/#asterisk sudharshaw (~sudharsha@175.157.6.221) |
| 01:37.50 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 01:40.20 | *** part/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 01:42.13 | *** join/#asterisk tuxd00d (~tuxd00d@ip70-162-151-19.ph.ph.cox.net) |
| 01:43.56 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 01:45.37 | *** join/#asterisk sudharshaw (~sudharsha@175.157.39.200) |
| 02:11.50 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 02:36.39 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-121-144.dynamic.qsc.de) |
| 02:39.42 | *** join/#asterisk vader- (~Adium@pool-173-49-160-70.phlapa.fios.verizon.net) |
| 02:41.14 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:51.40 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-pjquhozodliwipgw) |
| 02:53.58 | *** join/#asterisk mjordan (~mjordan@199.119.131.162) |
| 02:53.58 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:12.40 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 03:16.09 | *** join/#asterisk D30 (~D30@58.71.19.178) |
| 03:20.10 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 04:15.00 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 04:26.24 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 04:30.42 | *** join/#asterisk camerin (~camerin@elite.bshellz.net) |
| 04:51.54 | *** join/#asterisk vader- (~Adium@pool-173-49-160-70.phlapa.fios.verizon.net) |
| 04:54.01 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 05:05.52 | *** join/#asterisk tuxd00d (~tuxd00d@ip70-162-151-19.ph.ph.cox.net) |
| 05:34.37 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 05:45.35 | *** join/#asterisk tuxd00d (~tuxd00d@ip70-162-151-19.ph.ph.cox.net) |
| 05:45.57 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:00.59 | *** join/#asterisk KNERD (~KNERD@198.12.96.132) |
| 06:16.19 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 06:22.18 | *** join/#asterisk D30 (~D30@58.71.19.178) |
| 06:28.12 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 06:29.31 | *** part/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 06:33.29 | *** join/#asterisk MrFidget (~MrFidget@r49-2-11-73.cpe.vividwireless.net.au) |
| 06:34.36 | ChannelZ | did SIPDtmfMode get silently deprecated? I see the app registered in chan_sip.c but can't actually find/figure out where it does anything. |
| 06:51.55 | *** join/#asterisk SaintMoriarty (~SaintMori@ip68-231-100-174.ph.ph.cox.net) |
| 06:52.29 | SaintMoriarty | Are there any good open source CDR apps anyone has used? |
| 06:54.55 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 07:09.25 | ChannelZ | that do what? |
| 07:09.50 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 07:11.57 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 07:12.16 | *** join/#asterisk jeffspeff (~Jeff@12.49.160.131) |
| 07:12.57 | *** join/#asterisk bkruse (~Adium@74.51.115.100) |
| 07:13.36 | linocisco | asterisk on virtualbox has call quality problems using two softphones |
| 07:13.55 | linocisco | it has background noise, not clear voice |
| 07:14.07 | linocisco | what should we do? |
| 07:14.32 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 07:17.23 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 07:19.11 | *** join/#asterisk Acilim (~ahmet@unaffiliated/acilim) |
| 07:19.33 | Acilim | Hello all, I used "Call Forward All Activate" and forward my phone to another extention, using featurecodes. Where in management/database I can see forwarded number of extension? |
| 07:21.06 | *** join/#asterisk zildjian (~zildjian@unaffiliated/zildjian) |
| 07:22.16 | zildjian | anyone sucessfully received SMS/SMSC messages from voip provider vitelity to a DID on an asterisk server? I can't get it to work at all |
| 07:34.35 | ChannelZ | I had it working at one point awhile back; they essentially do it as XMPP if memory serves |
| 07:34.45 | zildjian | that would work great |
| 07:34.49 | zildjian | just want to receive only |
| 07:35.23 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:37.59 | *** join/#asterisk ghostlines (~ghostline@i191002.upc-i.chello.nl) |
| 07:38.29 | ChannelZ | You basically configure xmpp.conf to talk to vitelity, and send it to like an "incoming-sms" context in your dialplan, and then you can use the MESSAGE() function in your dialplan to read them |
| 07:38.58 | ChannelZ | NoOp(SMS from ${MESSAGE(from) - ${MESSAGE(body)}) for instance |
| 07:39.43 | ChannelZ | Acilim: that sounds like FreePBX.. amIright? |
| 07:40.35 | Acilim | yes thats true |
| 07:40.57 | zildjian | ChannelZ: makes sense now |
| 07:41.20 | zildjian | I just logged in via pidgin but no text messages show up despite being SMS enabled on the DID |
| 07:41.31 | zildjian | I will have to log a ticket |
| 07:42.16 | ChannelZ | Acilim, that's a feature of FreePBX and not base asterisk, so you'd have to ask them. Probably in some mysql/*sql database the rest of freepbx uses |
| 07:43.22 | Acilim | ChannelZ do you know where in base asterisk, where these datas stored? |
| 07:43.51 | ChannelZ | As I said 'call forwarding' like that isn't a built-in function of asterisk |
| 07:44.19 | zildjian | I'm using 1.8.32 though so that might be the issue |
| 07:44.52 | ChannelZ | is that still Jabber zildjian? I think I had it working under the old system too |
| 07:45.24 | ChannelZ | jabber.conf or whatever |
| 07:45.28 | Acilim | ChannelZ, Thank you I understand |
| 07:47.11 | zildjian | I've got gchat still working under the old system |
| 07:47.33 | zildjian | yeah jabber.conf |
| 07:47.49 | *** join/#asterisk CeBe (~CeBe@port-92-200-121-144.dynamic.qsc.de) |
| 07:48.17 | ChannelZ | well I ask because under jabber it worked differently as far as how things made it in/out from what I can remember.. I think there was like a JabberSend() dialplan function and maybe JabberReceive or JabberRead or something |
| 07:48.42 | zildjian | yeah you are right about that I think. |
| 07:48.55 | zildjian | though at this point I can't evne get a message to show up in pidgin |
| 07:49.59 | ChannelZ | I never tried that :) |
| 07:50.30 | ChannelZ | (and it looks like it was a function, JABBER_RECEIVE() ) |
| 07:50.45 | zildjian | I just created this DID so maybe it has'nt been added to the SMSC yet |
| 07:50.55 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 07:50.55 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:51.25 | ChannelZ | oh, you have a short code? |
| 07:51.34 | zildjian | nope |
| 07:51.36 | zildjian | no short code |
| 07:52.06 | ChannelZ | oh.. well I think it's relatively instant then, or should be |
| 07:52.11 | zildjian | hmm |
| 07:52.22 | zildjian | something must be broken on the vitelity side then. I can't explain this |
| 07:54.01 | ChannelZ | haha I just re-enabled my xmpp in asterisk and I had 6 texts that had been queued up at vitelity for god knows how long |
| 07:54.21 | zildjian | lol |
| 07:54.46 | zildjian | what do you tell asterisk to do with the messages? |
| 07:55.00 | ChannelZ | I just had them NoOp() to the console |
| 07:55.38 | zildjian | ah ok |
| 07:55.44 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 07:55.45 | zildjian | I'm probably gonna insert them into a database |
| 07:55.54 | ChannelZ | I don't really have a use for it, I just set it up to see how it worked when Vitelity first started offering it a few years ago |
| 07:55.57 | zildjian | then get them via custom rest api |
| 07:56.23 | zildjian | maybe I should upgrade my asterisk |
| 07:56.57 | zildjian | oh gosh I'm running 1.8.15.1 |
| 07:56.59 | zildjian | :( |
| 07:57.03 | ChannelZ | you can try turning on jabber debug to see if it's receiving messages it doesn't understand or something |
| 07:57.18 | ChannelZ | see if they're even making it to asterisk in the first place |
| 07:59.59 | zildjian | yeah even with their own web tool on the GUI it's not even working |
| 08:00.02 | zildjian | Just gonna file a ticket |
| 08:00.51 | ChannelZ | hmm. I logged in and it said "Offlines messages: 6" and sure enough when I fired up asterisk it barfed them all at me |
| 08:00.56 | *** join/#asterisk areski (~areski@80.174.128.54.dyn.user.ono.com) |
| 08:01.34 | zildjian | yeah mine says 0 |
| 08:01.45 | zildjian | even though I've sent the DID like 25 messages |
| 08:03.59 | zildjian | hehe intresting |
| 08:04.18 | zildjian | it's so broken |
| 08:04.49 | ChannelZ | hmm. You said you have pidgin running, at the same time as jabber on asterisk? Maybe it's not delivering to multiple destinations properly |
| 08:05.05 | zildjian | disabled the jabber in asterisk |
| 08:05.17 | zildjian | i'm connected directly via pidgin now |
| 08:05.27 | zildjian | i can connect |
| 08:05.34 | zildjian | but can't receieve/send anything |
| 08:10.27 | ChannelZ | hmm, interesting. I just logged into mine via pidgin and it added 3 buddies; one of my cell phone (which presumably I'd sent an SMS to a long time ago as a test), one of the number I'd had those 6 queued messages from, and one called 'S.MS Help' |
| 08:10.43 | zildjian | yeah |
| 08:10.56 | zildjian | S.MS help just tells you how to add a new number or short code |
| 08:11.41 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:14.01 | *** join/#asterisk areski (~areski@80.174.128.114.dyn.user.ono.com) |
| 08:19.06 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 08:22.45 | zildjian | ChannelZ: SMS is working with another DID of mine. I just think this DID I was using before is screwed up |
| 08:23.09 | ChannelZ | Wacky. |
| 08:23.11 | zildjian | ChannelZ: how do you differentiate between a regular call and a SMS based call? |
| 08:23.16 | zildjian | in asterisk |
| 08:23.45 | ChannelZ | well I have jabber/xmpp going into a different context |
| 08:24.08 | ChannelZ | (it's not related to your SIP DID anyway) |
| 08:24.42 | ChannelZ | IE I have [xmpp-in] and in it exten => s,1,NoOp(${MESSAGE(body)}) |
| 08:24.47 | zildjian | oh that's right |
| 08:24.58 | ChannelZ | or in your case NoOp(${JABBER_RECEIVE(accountname)}) (or something like that |
| 08:24.59 | zildjian | asterisk never sees it other than the jabber/xmpp config |
| 08:25.16 | zildjian | so it's essentially just a polling mechnism |
| 08:26.03 | ChannelZ | well not exactly polled, the XMPP message gets sent from wherever to asterisk (which is logged into the server) and then when it receives a message, it passes it along to the dialplan |
| 08:26.29 | zildjian | right |
| 08:26.47 | zildjian | and then you create a bunch of custom stuff to deal with it there. |
| 08:26.53 | ChannelZ | yeah |
| 08:27.02 | zildjian | like call all your phones in your house and play michael jackson music on answer |
| 08:27.06 | zildjian | etc |
| 08:27.19 | ChannelZ | If that seems like a useful thing to do, sure :) |
| 08:27.24 | zildjian | haha ;) |
| 08:27.58 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 08:30.11 | zildjian | ChannelZ: Is it possible to track down who owns a DID? My cell phone gets spammed continually by autodialers (most likely running asterisk or freepbx). |
| 08:30.23 | zildjian | is there some kind of registry for who owns specific DIDs |
| 08:30.35 | zildjian | I'm guessing not |
| 08:31.39 | ChannelZ | I'm sure there is on the telco side but nothing public-facing that I personally know of, besides punching it into Google and seeing if it shows up on one of the sites like 800notes.com etc |
| 08:31.52 | zildjian | right |
| 08:31.57 | zildjian | figured I'd ask |
| 08:37.56 | *** join/#asterisk bulkorok (~Benjamin_@p5099b1e7.dip0.t-ipconnect.de) |
| 08:39.21 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 08:43.03 | linocisco | hia ll |
| 08:43.16 | ChannelZ | oy |
| 08:43.17 | linocisco | i have problem with background noise on softphone |
| 08:43.35 | linocisco | Blink and zoiper also |
| 08:44.33 | ChannelZ | hard to diagnose. Maybe the audio settings on the computer? Or what codec is being used? Can you define "background noise" better? |
| 08:46.10 | linocisco | i can hear other side but it is like talking outside in the wind even though we are talking in office. |
| 08:52.09 | ChannelZ | Does it sound digital in nature? Do you have a recording? |
| 08:52.16 | *** join/#asterisk ghostlines (~ghostline@i191002.upc-i.chello.nl) |
| 08:52.16 | *** join/#asterisk dustinm (~dustinm@105.ip-167-114-152.net) |
| 08:58.05 | linocisco | ChannelZ, what do you mean? |
| 08:59.40 | ChannelZ | Well, does it literally sound like wind or does it sound like a distortion.. popping, clicking, phasing, etc. Do you only hear it when talking or is it there even when you are silent? |
| 09:00.23 | ChannelZ | do prompts from asterisk (voicemail menu, anything) sound bad as well or only when you're connected between softphones? |
| 09:00.34 | ChannelZ | And again, what codec(s) are being used in the call? |
| 09:08.52 | *** join/#asterisk infobot (ibot@rikers.org) |
| 09:08.52 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); 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 |
| 09:13.52 | *** join/#asterisk Cust0sL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 09:20.01 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 09:33.15 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 09:35.19 | *** join/#asterisk ghostlines (~ghostline@82.201.93.50) |
| 09:36.44 | *** join/#asterisk ghostlines (~ghostline@82.201.93.50) |
| 09:37.00 | linocisco | ChannelZ, i guess it is double echo cancellation. I know how to turn off echo cancellation from softphone but I dont know how to from Asterisk side |
| 09:40.45 | *** join/#asterisk areski (~areski@221.Red-88-5-47.dynamicIP.rima-tde.net) |
| 10:00.26 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 10:04.05 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 10:22.46 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 10:23.28 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 10:36.16 | drathir | its normal obtain [Apr 30 12:30:39] NOTICE[20051]: res_pjsip/pjsip_distributor.c:256 log_unidentified_request: Request from '"1801" <sip:1801@server_external_ip>' failed for 'random_ip_adresses:5092' (callid: 2e11ec5-f27160a-XXXXXXX@server_external_ip) - No matching endpoint found |
| 10:37.27 | drathir | the 1801 isnt used in any config file... |
| 10:39.39 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 10:45.15 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-pdowlacyotinqgeb) |
| 10:58.52 | *** join/#asterisk RoyK (~roy@77.88.71.251) |
| 11:02.30 | RoyK | hi all. I have a little thing I'd love to make, preferably small using a raspberry pi, for instance. I want to connect an old, analogue phone to this thing (to an FXO?) and connect the "thing" to my cell phone with USB |
| 11:02.55 | RoyK | meaning when someone call my cell phone, then I can pick up the rotary phone, using that as a "handsfree"-like unit |
| 11:03.08 | RoyK | anyone seen something like this? |
| 11:11.09 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 11:15.27 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 11:29.44 | *** join/#asterisk sysop2 (~sysop@c-24-98-217-140.hsd1.ga.comcast.net) |
| 11:29.49 | sysop2 | hi |
| 11:32.10 | sysop2 | question is there any way I can use asterisk to make a normal phone into one of those dial a number automatically phones when it goes off hook? usually they are for 911 but I have seen other uses. normally its in the phone. but can asterisk detect when an old school ananlog phone hooked up to a analog to sip converter goes off hook and act accordingly |
| 11:33.23 | sysop2 | also if I got one of those automatic dial 911 phones could I reroute 911 just for that one extension? |
| 11:34.04 | sysop2 | I want to put a phone on the front of my house that when picked up it will connect to my sip phone. |
| 11:34.38 | r00f | i don't know about hook (personally i never saw any events in dahdi, relate to that), but as for different routing for one extension - yes |
| 11:35.06 | r00f | exten => _911/101,1,Dial(SIP/phone) |
| 11:35.15 | r00f | where 101 is your extension number |
| 11:35.49 | sysop2 | sweet! |
| 11:51.12 | *** join/#asterisk hehol (~hehol@locanet.dyndns.org) |
| 11:53.25 | Acilim | Hello all, I am newbie in asterisk. When I am testing outbound calls I realized that, when I make outbound call, even its not answered(just ringed) I see answered status in my cdr. Anyone has any recommendation with this? |
| 11:59.37 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:12.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:14.30 | r00f | Acilim, there can be different causes. it may be something in your dialplan telling asterisk to Answer() before dialing out |
| 12:15.09 | r00f | or, some sort of FAS on receiving end |
| 12:15.29 | r00f | [TK]D-Fender for sure knows better |
| 12:15.46 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:17.15 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-vhxtvxalugnlwyxf) |
| 12:21.53 | drathir | lol |
| 12:22.07 | drathir | all opinions are welcomed i guessing... |
| 12:22.56 | Greenlight | In "sip show channelstats" how does it calculate the lost% for send? And is it always accurate? |
| 12:23.38 | eirirs | dunno if you got some sort of numbering of UDP packets |
| 12:23.49 | [TK]D-Fender | drathir: What's to guess at? You are getting random connection attempts from hackers across the net. |
| 12:25.31 | Greenlight | So the remote side sends details of the packets it didn't get based on the number? |
| 12:25.56 | Greenlight | I could understand the receive side, based on the seq number, but was unsure about the send |
| 12:26.03 | eirirs | Greenlight: the remote side have no idea of which udp packets you were receiving |
| 12:26.27 | Greenlight | So how does channelstats calculate the "send" stats? |
| 12:26.44 | eirirs | maybe by how much it's actually sending? |
| 12:26.55 | Greenlight | It shows lost% |
| 12:27.15 | Greenlight | I'm seeing some lost% of 15% and trying to see if it's a real figure or not |
| 12:27.42 | eirirs | maybe something at any layers above udp layer that calculates this |
| 12:29.40 | drathir | [TK]D-Fender: thanks, but that opinion was about 14:15 < r00f> [TK]D-Fender for sure knows better |
| 12:30.57 | [TK]D-Fender | drathir : its normal obtain [Apr 30 12:30:39] NOTICE[20051]: res_pjsip/pjsip_distributor.c:256 log_unidentified_request: Request from '"1801" <sip:1801@server_external_ip>' failed for 'random_ip_adresses:5092' (callid: 2e11ec5-f27160a-XXXXXXX@server_external_ip) - No matching endpoint found |
| 12:31.01 | [TK]D-Fender | drathir: ^^^^ |
| 12:31.16 | [TK]D-Fender | drathir: I was answering YOUR question from earlier |
| 12:31.59 | [TK]D-Fender | [08:22]drathirall opinions are welcomed i guessing... <- thios was not worded in a way that is suggestive that you're talking about someone else's situation |
| 12:35.49 | drathir | [TK]D-Fender: oh thanks a lot too im appreciated, that kind of activity are often happen or is fare thing? |
| 12:36.00 | drathir | fare/rare* |
| 12:36.20 | [TK]D-Fender | drathir: At any point your system may get contacted by scanners out there looking for boxes to exploit |
| 12:38.09 | War_Bear | Is it possible to get nano seconds in asterisk? |
| 12:38.41 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:38.45 | War_Bear | like date +%N |
| 12:40.36 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 12:48.48 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 12:50.24 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 12:55.35 | *** join/#asterisk mdhas (~mdhas@76.9.206.238) |
| 12:56.10 | *** join/#asterisk mjordan (~mjordan@104-145-12-182.dedicated.allstream.net) |
| 12:56.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:17.02 | *** join/#asterisk Draecos (~Draecos@106-69-148-154.dyn.iinet.net.au) |
| 13:21.23 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 13:21.33 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 13:21.46 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:24.29 | *** join/#asterisk camerin (~camerin@elite.bshellz.net) |
| 13:31.32 | *** join/#asterisk aross42 (~aross@dsl-eclipsefnd.storm.ca) |
| 13:35.13 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:41.39 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 13:43.16 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 13:51.26 | *** join/#asterisk theron (~theron@2620:10d:c0a4:1304:e4d:e9ff:fe9a:dab6) |
| 13:54.40 | *** join/#asterisk zapata (~zapata@2001:470:1f0b:11bc:e4ee:495b:d042:a421) |
| 13:55.09 | drathir | someone sayin somethin like 195.154.41.182 ? it isnt any "default" config pointing? beter ask before report abusive behaviour... |
| 13:59.46 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 14:01.20 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 14:02.00 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-pvhksjehsyusasjm) |
| 14:07.08 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 14:12.15 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:12.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:13.00 | *** join/#asterisk kharwell (kharwell@nat/digium/x-idjfmupjjengeadj) |
| 14:14.00 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:15.14 | X-Rob | http://www.reddit.com/r/PHP/comments/34ebz7/were_the_freepbx_dev_team_we_have_our_own_php/ |
| 14:15.20 | X-Rob | ^^^ for anyone who's interested. |
| 14:21.37 | *** join/#asterisk redrook (488f38d2@gateway/web/freenode/ip.72.143.56.210) |
| 14:21.39 | *** part/#asterisk Ove_ (Ninj4@ar.en.snuskgub.be) |
| 14:21.57 | redrook | hi, everybody |
| 14:22.33 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 14:22.51 | redrook | question: does asterisk 13 works with webrtc for video calls (like sipml5). spend days and can not make ut work. |
| 14:23.31 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-cfeqvnooporvksmc) |
| 14:23.31 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:33.58 | *** part/#asterisk redrook (488f38d2@gateway/web/freenode/ip.72.143.56.210) |
| 14:35.19 | *** join/#asterisk gugaua (5e66624d@gateway/web/freenode/ip.94.102.98.77) |
| 14:36.39 | gugaua | Hello, I have got a "problem" with asterisknow... call recordings page is showing a blank page since my clean installation... does anyone know about this problem? |
| 14:38.24 | *** join/#asterisk redrook (488f38d2@gateway/web/freenode/ip.72.143.56.210) |
| 14:39.23 | redrook | anybody here with info about asterisk 13+ webrtc + video ? |
| 14:42.15 | *** join/#asterisk K0HAX (~K0HAX@c-75-72-143-131.hsd1.mn.comcast.net) |
| 14:47.41 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 14:50.38 | *** join/#asterisk vader- (~Adium@204.183.88.4) |
| 14:51.26 | jeffspeff | in my dialplan, i have exten=fax,n,Set(FAXOPT(maxrate)=9600) and exten=fax,n,Set(FAXOPT(minrate)=9600) yet when you look at the packet capture, it shows a bitrate of 14400 |
| 14:55.05 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 15:03.08 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 15:07.12 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-121-144.dynamic.qsc.de) |
| 15:11.39 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-vifwhshbjobwtnxj) |
| 15:14.12 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:14.29 | *** join/#asterisk KNERD (~KNERD@198.12.96.132) |
| 15:20.44 | *** join/#asterisk camerin (~camerin@elite.bshellz.net) |
| 15:28.53 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 15:36.11 | *** join/#asterisk OurRoyalGabe (~quassel@cpe-104-162-60-254.nyc.res.rr.com) |
| 15:58.27 | *** join/#asterisk mbowie (~mbowie@162.212.36.9) |
| 16:00.32 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 16:20.50 | *** join/#asterisk afournier1 (~admin@46.255.181.29) |
| 16:24.54 | afournier1 | I've been following the Patch Contribution Process on the wiki, and i cannot connect to Gerrit... it looks like my account was automatically created on, but with no password. When trying to reset the password with Gerrit, i receive a mail that links to http://yooden.digium.api:8095/crowd/console/resetpassword.action => unknown host. what should i do ? |
| 16:25.57 | mjordan | afournier1: did you sign a CLA? |
| 16:26.00 | afournier1 | yep |
| 16:26.13 | mjordan | afournier1: when you click sign in, does it take you to openid.asterisk.org? |
| 16:26.35 | afournier1 | yes |
| 16:26.45 | mjordan | your password should be the same as JIRA |
| 16:26.47 | afournier1 | i signed the CLA 10 minutes ago |
| 16:26.50 | mjordan | ah |
| 16:26.53 | mjordan | it hasn't been accepted yet |
| 16:26.55 | afournier1 | i see |
| 16:27.02 | mjordan | it has to be reviewed |
| 16:27.09 | afournier1 | i'll wait then, true i did not receive an e-mail after i signed the CLA |
| 16:27.14 | mjordan | yup |
| 16:27.17 | afournier1 | thanks |
| 16:27.20 | mjordan | soon as you get that, you should be able to login |
| 16:27.24 | mjordan | np, sorry for the confusion |
| 16:30.36 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:36.43 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:44.38 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 16:48.00 | *** join/#asterisk PHunter (~Phunter@wsip-68-14-213-118.ph.ph.cox.net) |
| 16:48.30 | PHunter | How well does Asterisk work with saving voicemails on a remote server? |
| 16:48.40 | PHunter | Does it 'stream' it to the file? |
| 16:48.47 | [TK]D-Fender | No |
| 16:48.52 | PHunter | (aka doesn't buffer then save..) |
| 16:49.01 | [TK]D-Fender | You should have read up on the backend =storage options. |
| 16:49.08 | [TK]D-Fender | Databse / IMAP / file |
| 16:49.20 | PHunter | Voicemails in DB? |
| 16:49.28 | PHunter | wat |
| 16:50.01 | [TK]D-Fender | ~book |
| 16:50.01 | 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 |
| 16:50.01 | PHunter | The default is /var/lib/asterisk/spool |
| 16:50.20 | [TK]D-Fender | There is no such thing as "default" |
| 16:50.21 | PHunter | If i Link that to another server, should I expect garbled/cutoff messages |
| 16:50.24 | [TK]D-Fender | They go where you configure. |
| 16:50.28 | PHunter | asterisk.conf |
| 16:50.30 | PHunter | comes out of box |
| 16:50.32 | [TK]D-Fender | What you have in your configs ....is your business... |
| 16:50.49 | PHunter | Right. |
| 16:50.50 | [TK]D-Fender | Asterisk make have SAMPLE configs... but those are SAMPLES, not "default" |
| 16:50.55 | [TK]D-Fender | may* |
| 16:51.01 | PHunter | okay. |
| 16:51.08 | PHunter | lets say its /folderA |
| 16:51.16 | PHunter | if folderA is actually a network share |
| 16:51.19 | [TK]D-Fender | Then files are files |
| 16:51.29 | PHunter | should I expect an issue with garbled/cutoff messages? |
| 16:51.33 | [TK]D-Fender | No storage metho involves transmitting raw audio that can get lost |
| 16:51.42 | [TK]D-Fender | there is no such thing as "garbled" at the point of storage |
| 16:51.51 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 16:52.04 | PHunter | Ok. So what could cause garbled/cutoff messages? |
| 16:52.13 | [TK]D-Fender | Crappy SOURCE |
| 16:52.28 | PHunter | Compiler Flags maybe? |
| 16:52.30 | [TK]D-Fender | The channel you are recording is bad |
| 16:52.33 | [TK]D-Fender | no |
| 16:52.35 | PHunter | oh |
| 16:52.36 | [TK]D-Fender | your CALL is bad |
| 16:52.38 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 16:52.45 | PHunter | Well. |
| 16:52.54 | [TK]D-Fender | </story> |
| 16:53.04 | PHunter | Then a significant amount of calls have been bad over the last few weeks |
| 16:53.25 | [TK]D-Fender | You will probably want to do something about that... |
| 16:53.46 | PHunter | A guy sitting 10 feet from me, can call my phone, leave a voicemail, and it sounds like fast forwarding, jumping, and ends early. |
| 16:53.52 | PHunter | Server is ~40ms away |
| 16:53.54 | PHunter | no NOT |
| 16:53.55 | PHunter | NAT* |
| 16:54.24 | PHunter | Server: no NAT, Endpoints: NAT. |
| 16:54.33 | PHunter | This is not the only scenario. |
| 16:56.29 | [TK]D-Fender | 40ms doesn't mean the packet flow isn't jittery garbage, or that CPU timing isn't bad |
| 16:56.49 | PHunter | True. |
| 16:56.50 | [TK]D-Fender | It has precisely zero to do with the storage however |
| 16:57.06 | PHunter | Okay, so its the connection to asterisk, and not asterisk to storage? |
| 16:57.19 | [TK]D-Fender | [12:56][TK]D-FenderIt has precisely zero to do with the storage however |
| 16:57.22 | *** join/#asterisk gusto (~gusto@2001:4c50:62f:2800:8219:34ff:fecf:17f0) |
| 16:57.34 | PHunter | I get it. Just verifying. |
| 16:57.51 | [TK]D-Fender | That didn't leave anything to interpretation |
| 16:58.00 | PHunter | I ask questions because obviously I don't know the answer. No need to be agressive. |
| 16:58.22 | [TK]D-Fender | just saying it should 100% from that |
| 16:58.26 | [TK]D-Fender | clear* |
| 16:59.05 | PHunter | Well the issue is that I have a boss who isn't all there and If I don't cross my T's and dot my I's on Everything, The ship sinks. |
| 16:59.38 | PHunter | So forgive me for verifying what your saying. |
| 16:59.45 | PHunter | bows down. |
| 16:59.57 | [TK]D-Fender | I wasn't yelling here... |
| 17:00.16 | [TK]D-Fender | ll good.... |
| 17:00.18 | [TK]D-Fender | a |
| 17:00.46 | PHunter | No but the way you 'said' it, shows a bit of snideness. |
| 17:01.45 | PHunter | I appreciate all the help I can get. But I didn't chose to the DCAA. I was gifted the system. Im doing the best I can, with what I have. |
| 17:02.30 | *** join/#asterisk philfry (~albertk@50-200-136-34-static.hfc.comcastbusiness.net) |
| 17:02.31 | PHunter | Overworked, underpaid, and nobody else has a clue whats going on. Im up until 1am troubleshooting crap. |
| 17:02.36 | PHunter | and back at work at 8. |
| 17:08.59 | Cyford33 | PHunter does the call sound fine if yall talking from exten to exten? |
| 17:11.14 | PHunter | Yes |
| 17:15.42 | PHunter | Cyford33: Yes the audio sounds fine when calling between extensions. Its purely voicemail thats the issue. |
| 17:16.53 | Cyford33 | i am guessing you didnt have any voicemail formt changing scripts right, like formatting it in mp3 of something like that |
| 17:19.56 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@186.6.246.236) |
| 17:21.04 | PHunter | No its gsm wav and ulaw. |
| 17:23.25 | PHunter | Well |
| 17:23.37 | PHunter | just standard gsm and wav and the text file are generated. |
| 17:24.19 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 17:24.43 | Cyford33 | im guessing codec is gsm and voicemails are wav |
| 17:24.55 | Cyford33 | i sec though |
| 17:25.32 | [TK]D-Fender | Voicemail records the raw stream, phones may be doing PLC to cover up the crappy connection |
| 17:25.50 | PHunter | PLC? |
| 17:26.00 | [TK]D-Fender | Packet Loss Concealement |
| 17:26.13 | [TK]D-Fender | Jitter Buffer can play in as well |
| 17:26.44 | PHunter | Okay. Constructive. Is there a way to check on that? |
| 17:26.52 | PHunter | When I get money Ill buy the book. |
| 17:27.10 | PHunter | But until then, I have 86 endpoints jumping up and down. |
| 17:29.53 | PHunter | Ill be right back I have to reboot my subpar workstation. |
| 17:31.44 | *** join/#asterisk riess82 (~riessma@62-46-210-110.adsl.highway.telekom.at) |
| 17:34.01 | Greenlight | I've an install where I'm seeing: "channel.c: Didn't get a frame from channel SIP/...." It then hangs up the channel. Why is this? |
| 17:35.29 | Greenlight | Seems odd that not receiving a single frame would be cause for hanging it up |
| 17:36.02 | file | that is one of the mechanisms used to signal hangup |
| 17:36.40 | Greenlight | Oh, from the remote endpoint? |
| 17:37.17 | file | it can be what a hangup received from the outside is translated into within the core |
| 17:37.37 | *** join/#asterisk overyander (~Jeff@12.49.160.131) |
| 17:37.39 | overyander | I'm getting this error FAX CNG detected but no fax extension but I have a fax extension. https://gist.github.com/jeffclay/557cc8355e6d938ce034 can someone point me in the right direction on this please? |
| 17:37.44 | *** join/#asterisk kayfox (~kayfox@orca.zerda.net) |
| 17:37.51 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 17:38.00 | Greenlight | Hmm.. okay. My clients are complaining of random call drops, and that's all I can spot of merit in the debug log ;/ |
| 17:40.36 | Cyford33 | [TK]D-Fender also heard garbled voicemail from wav audio players trying to play the WAV file without converting it.. plays good in phones, but not the acuall audio player, unless changing WAV into wav or mp3 |
| 17:42.13 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 17:42.41 | *** join/#asterisk beardy_ (~beardy@unaffiliated/beardy) |
| 17:45.40 | *** join/#asterisk AlHafoudh_ (~AlHafoudh@echo.freevision.sk) |
| 17:46.19 | *** join/#asterisk NightMonkey_ (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 17:47.21 | *** join/#asterisk yokel_ (~yokel@unaffiliated/contempt) |
| 17:47.46 | *** join/#asterisk Kilo`byte (~kilobyte@5.231.51.78) |
| 17:49.41 | *** join/#asterisk K0HAX (~K0HAX@c-75-72-143-131.hsd1.mn.comcast.net) |
| 17:49.41 | *** join/#asterisk ZerOlegend (~ZerOlegen@184-158-70-162.dyn.centurytel.net) |
| 17:49.41 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 17:49.41 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 17:49.41 | *** join/#asterisk murosai (~murokki@62-78-255-113.bb.dnainternet.fi) |
| 17:49.41 | *** join/#asterisk cryptic (~cryptic@pool-98-113-133-214.nycmny.fios.verizon.net) |
| 17:49.47 | *** join/#asterisk Eric-K (eric@188.226.225.197) |
| 17:53.07 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 17:53.52 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 17:55.19 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:55.27 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 17:55.30 | *** join/#asterisk vader- (~Adium@204.183.88.4) |
| 17:55.45 | *** join/#asterisk Asteriskuser42 (sking@nat/digium/x-gcnqwzvifrbyecor) |
| 17:55.58 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-rbqqpqbrqssavdra) |
| 17:58.32 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 18:05.59 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 18:09.04 | *** part/#asterisk riess82 (~riessma@62-46-210-110.adsl.highway.telekom.at) |
| 18:09.31 | overyander | what's the proper way to detect a fax from a voice call prior to dialing the sip peer/extension? |
| 18:28.19 | sgriepentrog | overyander: answer, play ringback for ~6 seconds, then dial. If your fax detect is enabled on the channel, it will jump to the fax. |
| 18:29.35 | overyander | sgriepentrog, thanks, i added the answer() and it started working. |
| 18:29.52 | overyander | sgriepentrog, are you familiar with faxing in asterisk? |
| 18:37.23 | overyander | is canreinvite deprectated in version 11.2? |
| 18:41.29 | sgriepentrog | overyander: fax: some. canreinvite still exists in chan_sip, but is different in pjsip (although that's 12+) |
| 18:44.03 | *** join/#asterisk Cust0sL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 18:45.49 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 18:47.54 | overyander | i'm trying to refer any incoming faxes to a new number with a different provider. The error i'm getting is SIP transfer to <sip:11235551234@MYPROVIDER.IP.ADDY> failed, REFER not allowed. |
| 18:48.58 | overyander | i have allowtransfer set to yes and promiscredir set to yes as well |
| 18:51.11 | *** join/#asterisk robmal (r@wporzo.pl) |
| 18:59.11 | Greenlight | I'm seeing a hangup with cause 64, but can't see that on https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings is there somewhere else these could be documented? |
| 19:01.04 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:04.08 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 19:06.56 | sgriepentrog | overyander: I'd do a sip packet capture to make sure, but it sounds like the provider you're trying to send REFER to is failing it - likely because they're not willing to support that. |
| 19:09.10 | sgriepentrog | Greenlight: sounds like somebody invented their own code. 64 is not valid afaik. |
| 19:33.50 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 19:37.14 | Greenlight | Hmm ... so my hangup issue looks like some odd race condition between bridging and mixmonitor. The cause 64 is AST_SOFTHANGUP_UNBRIDGE |
| 19:56.33 | mjordan | Greenlight: that is weird, as that isn't an actual hangup. That's a special "soft" hangup (which is an odd concept) that is just supposed to remove a channel from a bridge. |
| 19:56.41 | mjordan | Greenlight: in what circumstance are you seeing that? |
| 19:56.47 | *** join/#asterisk theron_ (~theron@199.201.64.131) |
| 19:57.03 | Greenlight | It seems to be when I request a bridge via AMI |
| 19:57.08 | Greenlight | They start to get all bridged up |
| 19:57.17 | Greenlight | And then I start a MixMonitor on one of the channels |
| 19:57.25 | Greenlight | Then I see that SoftHangup, and then a proper hangup |
| 19:57.37 | Greenlight | All in the same second ofc |
| 19:58.01 | Greenlight | I'm guessing it's trying to native bridge the two channels (same codec etc) ? |
| 20:00.51 | Greenlight | It's not occuring every time I do this, but enough that I've 15 emails listings from a customer, each one listing 5-10 customers that have been "disconnected" |
| 20:04.14 | Greenlight | I'm half wondering if I just comment out the block in channel.c that checks for no nomonitors or audiohooks before attempting a native bridge |
| 20:06.07 | Greenlight | Since I know that I *will* be attaching an audiohook to each bridge, even it's not quite been attached yet |
| 20:06.34 | Greenlight | I realise it's a bit of a hack, but might that help me? |
| 20:06.35 | *** join/#asterisk Cust0sL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 20:09.57 | mjordan | hm |
| 20:10.02 | mjordan | Greenlight: that should have been fixed |
| 20:10.05 | mjordan | Greenlight: what version? |
| 20:10.19 | mjordan | Greenlight: I know that was a bug in early 13, but I'm pretty sure we fixed that in 13.2 or 13.3. |
| 20:10.52 | Greenlight | 11.17 |
| 20:11.07 | Greenlight | I've not had the time to change my app for the AMI changes with 13 yet ;/ |
| 20:11.16 | Greenlight | Is the fix back-portable? |
| 20:12.42 | mjordan | hrm. |
| 20:12.53 | mjordan | I'm not sure why that would actually even be a bug in 11 |
| 20:13.32 | mjordan | really, the soft hangup is there to break a native bridge, otherwise your mixmonitor will get no audio |
| 20:13.36 | mjordan | native bridge => no media flowing |
| 20:13.51 | mjordan | why that would hangup the channel is beyond me. It shouldn't be doing that. Are you sure you aren't getting a BYE request? |
| 20:15.47 | *** join/#asterisk kharwell (kharwell@nat/digium/x-syvvcsaqbhjfpqut) |
| 20:17.43 | Greenlight | Oh, I thought the native bridge was the one where it avoids the core (packet to packet bridging is it?) |
| 20:17.59 | Greenlight | And yes, no BYE request from what I can see in DEBUG log |
| 20:19.07 | mjordan | Greenlight: yes, that is native bridging. If you need to record media, media has to flow through the core. That's what the soft hangup flag is attempting to do. |
| 20:19.28 | mjordan | Greenlight: it is a bug in 11 if nothing else. That soft hangup flag shouldn't cause a channel to be hung up. |
| 20:20.00 | Greenlight | Perhaps a race condition between the mixmonitor attaching the audiohook and the native bridge starting? |
| 20:20.26 | *** join/#asterisk j-fish (~Redneck@unaffiliated/j-fish) |
| 20:21.35 | mjordan | no idea without seeing a log. Please do file an issue with a log |
| 20:22.12 | Greenlight | http://pastebin.com/83mzaLHZ <-- That's the log, grep'd for just the two channels (otherwise it's massive) |
| 20:22.43 | Greenlight | It's strange; I've had other boxes running with the same setup for ages without this problem |
| 20:25.15 | Greenlight | Buuut if I just comment out that native bridge part, will it just always bridge through the core? |
| 20:34.30 | *** join/#asterisk gusto (~gusto@2001:4c50:62f:2800:8219:34ff:fecf:17f0) |
| 20:39.46 | *** join/#asterisk gusto (~gusto@2001:4c50:62f:2800:8219:34ff:fecf:17f0) |
| 20:53.55 | *** join/#asterisk MasterChen (~chen@chenb0x.net) |
| 20:53.58 | MasterChen | hi all |
| 20:56.16 | MasterChen | quick questiona bout sound files. I uploaded a custom greeting for an IVR into /var/lib/asterisk/sounds/custom/, and while the file is present and acknowledged in the CLI, it is not playing. I get dead air. I already chown'd asterisk:asterisk. The permissions were 644. I tried changing it to 755, but that didn't seem to work either |
| 20:56.31 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:04.54 | *** join/#asterisk theron (~theron@2620:10d:c0a4:1304:e4d:e9ff:fe9a:dab6) |
| 21:08.10 | *** join/#asterisk simonpatapon_out (~sime@unaffiliated/simonpatapon) |
| 21:09.01 | *** part/#asterisk simonpatapon_out (~sime@unaffiliated/simonpatapon) |
| 21:10.52 | RoyK | hm... seems finding something like a USB-FXS adapter may be hard. How hard would it be to rip out the guts from an old rotary phone and replace it with a pi and some electronics do the rotary dialing thing and then bridge between chan_alsa and chan_mobile? |
| 21:11.27 | RoyK | can't seem to find chan_alsa on this pi (just apt-get install of asterisk 1.8) |
| 21:16.40 | *** join/#asterisk areski (~areski@80.174.128.114.dyn.user.ono.com) |
| 21:17.26 | WIMPy | Ther are USB-FXS adaptors. But I have no idea how hared (or possible) it would be to get the driver working on raspbian. |
| 21:18.24 | RoyK | I guess that's another issue :P |
| 21:18.42 | RoyK | WIMPy: do you know where I can find those? I googled a bit and found almost nothing |
| 21:19.08 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-ysnikpdberfdbwxw) |
| 21:19.13 | RoyK | on the other hand - ripping out the guts of an old phone and replacing it with a pi and a small battery now seems like a rather good idea |
| 21:19.28 | WIMPy | I can't remember what the vendor was called. But maybe someone else remebers. |
| 21:20.59 | RoyK | I searched ebay and found just one USB FXS, and it turned out to be FXO |
| 21:21.04 | WIMPy | It took me a whole long day (night) to get an USB-BRI adapter working of Raspbian. It's the Bastard OS From Hell :-(( |
| 21:21.23 | RoyK | heh |
| 21:21.30 | RoyK | doesn't surprise me |
| 21:22.18 | WIMPy | It was not because the PI isn't the fastest. |
| 21:23.31 | RoyK | compared to what? |
| 21:23.49 | WIMPy | A standard PC. |
| 21:23.52 | [TK]D-Fender | ~savemoney |
| 21:23.52 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 21:24.02 | RoyK | WIMPy: well, it's not, obviously :P |
| 21:24.32 | RoyK | WIMPy: a PC wouldn't work too well on 5 watts of power |
| 21:24.54 | WIMPy | But I spent most of the time trying to find out what packages need to be installed in order to get anything going on the Raspbian shit. |
| 21:25.16 | WIMPy | I would definitely try Arch next time. |
| 21:25.59 | WIMPy | Not sure how much better it is, but it can't be worse. |
| 21:26.39 | RoyK | looks like Jessie might work directly on the pi2 |
| 21:26.42 | RoyK | hm... |
| 21:26.46 | RoyK | must try :D |
| 21:26.59 | WIMPy | Upgrading to Jessie was part of it, yes. |
| 21:27.09 | RoyK | not raspbian - debian |
| 21:37.39 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 21:39.14 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 21:40.51 | *** join/#asterisk sgriepentrog (~sgriepent@162-239-32-132.lightspeed.brhmal.sbcglobal.net) |
| 21:58.33 | *** join/#asterisk war9407 (~war9407-@static-72-73-18-14.clppva.fios.verizon.net) |
| 22:12.40 | zekoZeko | i recommend installing asterisk from raspbx on raspbian |
| 22:12.51 | zekoZeko | works like a charm for me |
| 22:13.11 | WIMPy | How is that going to help you with building drivers? |
| 22:13.21 | zekoZeko | but all it does is lookup the CID and pushes a button over GPIO if the number matches :) |
| 22:13.41 | zekoZeko | oh sorry, wasn't reading thoroughly enough |
| 22:20.44 | RoyK | debian jessie (not raspbian) on rpi2 - now let's see if it survives the initial dist-upgrade :) |
| 22:26.50 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 22:34.22 | Hsilamot | is there a channel for openvox cards or dahdi ? |
| 22:34.43 | WIMPy | No other than this one. |
| 22:37.18 | *** part/#asterisk mjordan (~mjordan@104-145-12-182.dedicated.allstream.net) |
| 23:06.00 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 23:12.56 | *** join/#asterisk xiddus (~xiddus@ns328768.ip-37-187-115.eu) |
| 23:17.00 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 23:17.53 | *** join/#asterisk pchero (~pchero@0x555140b5.adsl.cybercity.dk) |
| 23:24.00 | *** part/#asterisk kharwell (kharwell@nat/digium/x-syvvcsaqbhjfpqut) |
| 23:42.35 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 23:50.20 | sysop2 | hey found the answer to my question, how do you make an analog phone into a hotline/autodial phone. just use a $51 dollar converter and setup hotline in the web interface. pretty cool. http://www.mehrdust.com/archives/hotline-calls-plar-with-asterisk |
| 23:51.15 | sysop2 | hope someone finds it useful. I am putting an armored phone on the front of my house setup to ring my sip phone when someone picks it up. |
| 23:55.38 | [TK]D-Fender | sysop2, Cisco ATA 186's are rather rare to see out in the field. All of the Linksys/Siprura/nowCiscoAgain SPA series support this as well, and are available new and used for cheaper. |
| 23:57.36 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-arfrybanffkpmtkv) |
| 23:59.37 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |