| 00:00.23 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:01.56 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 00:13.38 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 00:34.31 | supersoaker | So i figured out my issue btw |
| 00:36.28 | supersoaker | the dial app had a gosub routine and during dialing its transfered the call to the local channel which invokes a second dial command with the same gosub routine |
| 00:37.10 | supersoaker | so, aprently, when dial is redirected it keeps the arguments from before? So i had stacked two call screening menues on top of eachother |
| 00:37.23 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 00:41.24 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 00:47.40 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 00:57.08 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 01:09.16 | *** join/#asterisk NoobSaibot (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 01:14.51 | *** join/#asterisk mfauzir (~m.fauzi.r@125.166.145.233) |
| 01:16.56 | *** join/#asterisk anthm (~anthm@static-198.22.64.229.freeswitch.org) |
| 01:37.01 | *** join/#asterisk aness (~aness@2a02:fe0:c310:3d0:217b:f60d:1ae0:e284) |
| 01:39.13 | *** join/#asterisk areski (~areski@80.174.128.116.dyn.user.ono.com) |
| 01:46.04 | *** join/#asterisk cunningpike (~cunningpi@2602:304:6eaf:6280:d1c0:2509:41d5:e8f1) |
| 01:53.29 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:02.19 | *** join/#asterisk cunningpike (~cunningpi@2602:304:6eaf:6280:d1c0:2509:41d5:e8f1) |
| 02:11.33 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 02:43.56 | mfauzir | hi all, need help please, i have installed asterisk+dahdi + TDM410P (2 FXS, 2 FXO) |
| 02:46.24 | WIMPy | The average psychic capabilities in here are low. So you will have to tell what you need help with. |
| 02:47.13 | mfauzir | one of my FXO connect to CO (Telco line), if i try to make outbound call from the extension phone , the called party has answered but the extension (calling party) always ringing (not hear sound or something from called party) |
| 02:49.25 | mfauzir | so, its like asterisk can not send answerd signal to extension phone |
| 02:49.36 | WIMPy | What about incomming calls? |
| 02:50.02 | WIMPy | Usually that comes right after dialling on pots, as there's no further indication. |
| 02:50.24 | mfauzir | incoming call, i don't test it yet. |
| 02:50.26 | WIMPy | But I'm not that much in to the analog stuff. |
| 02:51.55 | mfauzir | should i try to make incoming call first ? |
| 02:52.26 | WIMPy | It's always good to collect evidence. |
| 02:53.04 | mfauzir | ok wait.. |
| 02:57.45 | mfauzir | well, there is no problem with incoming call |
| 02:59.29 | WIMPy | So at least your hardware must be ok. |
| 03:05.23 | *** join/#asterisk jiku (~jiku@1.22.38.254) |
| 03:23.11 | *** join/#asterisk rumpler (b842a8af@gateway/web/freenode/ip.184.66.168.175) |
| 03:24.53 | rumpler | Hey guys, quick q; I'm reading up on OpenWRT/DD-WRT, and keep reading about people that are setting up Asterisk on their routers to make VoIP calls. I'm just wondering what the reasons/benefits behind it are (aside from it just being fun to play around with your own PBX), most voip phones can just connect to any arbitrary SIP server anyway, so why funnel everything through a router/asterisk server? |
| 03:25.40 | WIMPy | The same reason people use PBXs. |
| 03:27.44 | rumpler | WIMPy: Could you elaborate? In my (limited) experience, a PBX is essentially a router for voip calls, why bother with it when there are only one or two phones on the network, and they connect to an external PBX anyway. My best guess is that it's for funsies, but it's a project that I see come up over and over again, so maybe I'm missing some of the benefits |
| 03:28.17 | WIMPy | Well, in that case there is no point. |
| 03:28.49 | WIMPy | But you you only got an ITSP or a classic line and multiple phones, it does. |
| 03:29.01 | WIMPy | .. make sense, that is. |
| 03:30.10 | rumpler | As in, Asterisk acts as a mediator to route calls from multiple phones to a single itsp? I guess I can see the benefit there, thanks. |
| 03:30.34 | WIMPy | As I said: Just as with a PBX. |
| 03:30.50 | rumpler | thanks |
| 03:30.53 | WIMPy | Off course Asterisk is better at IVR stuff, if you need that. |
| 03:32.49 | rumpler | WIMPy: Yeah, IVR was the one real application that I could think of, but who needs IVR on a home network? (aside from tinkering, but I'm asking in more real-world terms) |
| 03:33.14 | WIMPy | I use it for a CAPTCHA. |
| 03:34.19 | rumpler | haha, for unknown numbers to have to solve before it rings your phone? (to kill bots?) |
| 03:34.32 | WIMPy | Exactely. |
| 04:06.22 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 04:06.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 04:42.31 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 04:56.22 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 04:56.31 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:1dd0:221:6aff:feb8:e0b2) |
| 05:07.08 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 05:11.23 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 05:12.42 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 05:25.51 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-ngizduhyeazbwzcv) |
| 05:45.34 | *** join/#asterisk Cynagen (~cynagen@ip72-208-60-104.ph.ph.cox.net) |
| 06:02.58 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 06:25.52 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 06:30.32 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:46.48 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 06:46.56 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 06:53.21 | babak | Hi, is there a variable |
| 06:53.38 | babak | i know call come from which span ? |
| 06:55.47 | *** join/#asterisk LiohAu (~LiohAu@bac69-8-88-165-4-152.fbx.proxad.net) |
| 06:56.37 | *** part/#asterisk alexw (~textual@unaffiliated/alexw) |
| 07:03.16 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 07:17.01 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 07:24.16 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:c542:f452:793:3714) |
| 07:29.59 | *** join/#asterisk W0rmDr1nk (~wormdrink@unaffiliated/wormdrink) |
| 07:37.21 | *** join/#asterisk jhlavacek (~jirka@avencall-gw-vbo.alionis.net) |
| 07:45.38 | *** join/#asterisk pryshebliad (~aab@2a00:b1c0:0:3:e2cb:4eff:feb5:132b) |
| 07:45.57 | pryshebliad | Hello! |
| 07:46.16 | pryshebliad | Why you don't making debian packages? |
| 07:46.29 | *** join/#asterisk klootzak (~maxi@46.45.23.37) |
| 07:46.35 | eirirs_ | pryshebliad: Why don't you? |
| 07:46.49 | pryshebliad | eirirs_: I can't |
| 08:08.55 | *** join/#asterisk skrusty (~skrusty@168.63.14.171) |
| 08:12.34 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:c542:f452:793:3714) |
| 08:19.09 | *** join/#asterisk CeBe (~CeBe@p5488BD40.dip0.t-ipconnect.de) |
| 08:20.34 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 08:26.49 | *** join/#asterisk wdoekes (~walter@wjd.osso.nl) |
| 08:32.30 | *** join/#asterisk vNistelroot (~secure@unaffiliated/vnistelroot) |
| 08:32.32 | vNistelroot | hi people |
| 08:32.43 | vNistelroot | any experience clustering asterisk with drbd? |
| 08:38.53 | zpotoloom | vNistelroot: did some testing a while back based on http://www.theserverexpert.com/elastix_2.4_ha_cluster-updated.pdf |
| 08:43.51 | vNistelroot | ohh, thank you zpotoloom, let me check it ! |
| 08:47.59 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 08:57.37 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 08:59.52 | vNistelroot | zpotoloom: took you a considered timeto set-up ? |
| 09:00.09 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:00.12 | vNistelroot | or is straight-forward |
| 09:02.52 | vNistelroot | for a little feedback |
| 09:11.17 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 09:19.23 | *** join/#asterisk calum_ (~calum_@host31-52-207-4.range31-52.btcentralplus.com) |
| 09:36.57 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 09:55.30 | Stefan27 | does make samples only add files in /etc/asterisk ? |
| 09:56.17 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 10:03.16 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 10:04.25 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 10:12.57 | *** part/#asterisk klootzak (~maxi@46.45.23.37) |
| 10:25.07 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-fqdzmtyoablmaifa) |
| 10:27.15 | vNistelroot | some feedback enabling srtp ? |
| 10:27.33 | *** part/#asterisk vNistelroot (~secure@unaffiliated/vnistelroot) |
| 10:36.24 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 10:40.12 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 10:45.10 | *** join/#asterisk areski (~areski@80.174.255.24.dyn.user.ono.com) |
| 10:53.35 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 10:56.22 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 11:09.54 | *** join/#asterisk hecatae (~hecatae@host-78-151-211-251.as13285.net) |
| 11:32.57 | *** join/#asterisk LiohAu_ (~LiohAu@bac69-8-88-165-4-152.fbx.proxad.net) |
| 11:38.00 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 11:52.16 | *** join/#asterisk mbowie (~mbowie@162.212.36.9) |
| 11:52.17 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 11:53.37 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.82.248) |
| 11:55.46 | ruben23 | hi guys any help and suggestion....which is better setup... 1. office site ----> local asteriks gateway -------peer ---->asterisk/US (voice/data separation) 2. office site -------> remote asterisk same country------->peer ------->US asterisk -- which will genarte best result with audio quality guys ..please any suggestion |
| 11:59.51 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 12:00.22 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:01.55 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 12:18.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:18.59 | *** join/#asterisk Draecos (~Draecos@106-68-45-158.dyn.iinet.net.au) |
| 12:23.21 | *** join/#asterisk cyborg-one (~cyborg-on@212-178-10-188.broadband.tenet.odessa.ua) |
| 12:24.41 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 12:29.40 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:29.47 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:36.46 | *** join/#asterisk aross42 (~aross@CPE000e84807d02-CMbcc8100f7698.cpe.net.cable.rogers.com) |
| 12:41.57 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 12:41.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:46.26 | *** join/#asterisk mbowie (~mbowie@162.212.36.9) |
| 12:48.49 | *** join/#asterisk jhlavacek (~jirka@80.215.229.102) |
| 12:53.46 | sekil | hello |
| 12:57.25 | file | hi |
| 12:59.15 | sekil | I have a situation where my * 11.12.1 sends 180 ringing to b-leg uppon receiving 183 w/o sdp |
| 12:59.30 | sekil | makes my b-leg fake rings |
| 13:00.03 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:03.21 | *** join/#asterisk jhlavacek (~jirka@91.211.167.225) |
| 13:04.59 | *** join/#asterisk theron (~theron@173.252.71.2) |
| 13:05.46 | *** join/#asterisk theron (~theron@173.252.71.2) |
| 13:07.01 | *** join/#asterisk kb3ien (~kb3ien@gazelle.ppm-vpn.databit7.com) |
| 13:07.33 | kb3ien | I'm getting some errors with func_odbc: I'm trying to debug backward from Unable to execute query [SELECT location FROM presence WHERE id='s'] |
| 13:15.41 | kb3ien | cli reports that `odbc read ODBC_PRESENCE s exec ` Failed to execute query. [SELECT location FROM presence WHERE id='s'] |
| 13:20.23 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-imbqmpwyadvtwgdp) |
| 13:20.23 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:20.23 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 13:22.43 | *** part/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 13:23.04 | kb3ien | is there a way to make odbc more verbose ? |
| 13:24.48 | *** join/#asterisk blee (~blee@50-88-4-82.res.bhn.net) |
| 13:29.21 | *** join/#asterisk shadej (~bzegeju@213.55.73.146) |
| 13:31.44 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 13:33.17 | jwr__ | i'm thinking about moving an on-site pbx to amazon. which means the connection between people's desk phones and the pbx would go over the internet. what are the best practices for that? SIP encryption? |
| 13:35.57 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:39.13 | kb3ien | isql can connect as the asterisk user, but not asterisk.... grrr. |
| 13:42.18 | kb3ien | SIP and RTP encryption are good, but with some codecs or with VAD, you want to be careful. The size of the packets are not constant. and the information leaks that way. |
| 13:42.57 | kb3ien | you probably want to consider running IPSEC in addition to other precautions. |
| 13:44.27 | *** join/#asterisk vNistelroot (~secure@unaffiliated/vnistelroot) |
| 13:44.49 | vNistelroot | Hi all |
| 13:45.12 | vNistelroot | zpotoloom: I have a question about drbd, could you please point me in the right way? |
| 13:47.21 | vNistelroot | For my servers, I have a 160 GB of SAS hd for SO and 1TB SATA for recordings and so on |
| 13:48.02 | vNistelroot | as far as I´ve read, when partitioning you must create two partitions for the SO, and let one unmounted for drbd |
| 13:48.22 | *** join/#asterisk jepperl (d5963da2@gateway/web/cgi-irc/kiwiirc.com/ip.213.150.61.162) |
| 13:49.21 | *** join/#asterisk kb3ien (~kb3ien@gazelle.ppm-vpn.databit7.com) |
| 13:49.36 | jepperl | Hey, im currently debugging an issue where the asterisk is sending RTP packets to another port than it announces to the client. This causes the server to sometimes not respect the range set in rtp.conf. There is no NAT (or PAT) involved in the setup on the server side |
| 13:50.26 | jepperl | Cant seem to find anything about it on the web, any chance some of you guys might have seen it before? |
| 13:50.27 | *** join/#asterisk cpugenius (cpugenius@unaffiliated/cpugeniusmv) |
| 13:50.46 | cpugenius | does anybody know of a way to detect that a recall (disconnect) has occurred in the atxfer feature? |
| 13:50.55 | Penguin | I'm fairly sure the end point device is responsible for telling asterisk which RTP ports it wants to use. |
| 13:51.21 | *** join/#asterisk jsjc (~Adium@5.9.67.27) |
| 13:51.32 | jepperl | In that case, asterisk does not comply with the client request |
| 13:51.35 | zpotoloom | vNistelroot: i'd recommend that you should really get familiar with drbd in general at first ( http://www.drbd.org/ ) before going near to any live systems |
| 13:52.08 | Penguin | What range are you setting on the phones, and what range are you setting on asterisk's rtp.conf? |
| 13:52.08 | wdoekes | jepperl: nat=auto/yes causes the client-set port to get ignored if the client sends from a different port first |
| 13:52.23 | zpotoloom | i'm not at home with drbd yet, thus can't give any good advice |
| 13:52.39 | wdoekes | symmetric rtp |
| 13:53.01 | zpotoloom | but the setup logic seems promising, and using the howto it worked |
| 13:53.06 | kb3ien | odbc seems to have no debugging at any level. |
| 13:53.16 | zpotoloom | just need to dig deeper |
| 13:53.26 | wdoekes | kb3ien: core set debug 8 |
| 13:53.30 | jepperl | penguin: i have tried with port > 1024 and 6000:30000 . In the rtp.conf i have 13000:65003 |
| 13:53.48 | wdoekes | and otherwise add tracing/debugging to unixodbc itself |
| 13:54.11 | jepperl | wdoekes: my serverside nat setting is force_rport;comedia |
| 13:54.24 | Penguin | That's the same as 'yes'. |
| 13:54.33 | kb3ien | yep it goinng to be a problem for another day.... |
| 13:54.44 | Penguin | If no nat is involved, set nat to 'no'. |
| 13:54.56 | wdoekes | jepperl: comedia == symmetric rtp |
| 13:57.45 | vNistelroot | thanks zpotoloom ! Im reading it deeply, im unsure how to manage partitions |
| 13:57.59 | vNistelroot | not the process but thinking about size |
| 14:06.51 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:09.25 | *** join/#asterisk c0rnoTa (~c0rnoTa@kobratel.starlink.ru) |
| 14:09.48 | jepperl | Thanks for the answers :) Is there any place on the webz where i can find information on all the dtls settings? (dtlssetup mostly).. Can't seem to find anything using google besides stackoverflow links with no real explanation |
| 14:12.18 | *** join/#asterisk superrafal (~superrafa@80.50.125.74) |
| 14:12.48 | superrafal | hi guys |
| 14:14.04 | superrafal | can anyone help me in getting WebRtc to work with asterisk 11.12? I'm using SipML5 library and I can't take it anymore. Too much problems ;) |
| 14:14.27 | jepperl | superrafal: what problems do you have? |
| 14:15.21 | superrafal | <PROTECTED> |
| 14:15.33 | superrafal | on menuselect in asterisk i don't see a module for that however |
| 14:15.43 | newtonr | https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 14:15.54 | file | there is no module, it's automatically compiled in if your OpenSSL has DTLS-SRTP support |
| 14:16.07 | file | the configure script does a check for the support |
| 14:16.50 | superrafal | great idea, gonna check it out thanks :) |
| 14:16.54 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 14:17.05 | newtonr | superrafal, see the dependencies in that tutorial |
| 14:18.27 | newtonr | jepperl, what about the documentation in the sample sip.conf ? |
| 14:18.37 | newtonr | jepperl, regarding your comment on dtlssetup |
| 14:19.25 | newtonr | jepperl, http://hastebin.com/luhuzopoka.vhdl |
| 14:21.44 | *** join/#asterisk g_r_eek (~g_r_eek@78-109-50.adsl.cyta.gr) |
| 14:21.52 | jepperl | newtonr: found it :) thanks a ton |
| 14:22.37 | newtonr | jepperl, if there is ever documentation seemingly missing from either the sample config files or wiki.asterisk.org (the two official places) then please let me know so I can create an issue to go write it. |
| 14:22.49 | newtonr | jepperl, np |
| 14:24.19 | jepperl | newtonr: i certainly will do that :) neat |
| 14:29.00 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 14:30.14 | *** part/#asterisk c0rnoTa (~c0rnoTa@kobratel.starlink.ru) |
| 14:35.14 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-qdkhjyyhjsdsqxjp) |
| 14:35.14 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:41.54 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:42.17 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:43.07 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-usexelbsjetyorls) |
| 14:45.53 | *** join/#asterisk CeBe (~CeBe@brsg-4dbbadf6.pool.mediaWays.net) |
| 14:55.21 | *** join/#asterisk petris (~petris@198.251.81.123) |
| 14:58.51 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-xarevkncogfybdxb) |
| 15:12.55 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:29.46 | babak | rmudgett: Hi, it it possible to use AOC in ECT service ? |
| 15:30.12 | babak | for tranfered calls |
| 15:30.44 | drmessano | With this recent BASH bug, I would like to move all my PBX'es to the more secure Windows platform. Who can help me install Asterisk in Windows? |
| 15:33.02 | Qwell | drmessano: Please to be trying http://www.asteriskwin32.com/ |
| 15:33.47 | drmessano | Most thank you |
| 15:34.17 | drmessano | You require support me?? |
| 15:44.22 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:1dd0:221:6aff:feb8:e0b2) |
| 15:45.29 | WIMPy | babak: Not from Asterisk. |
| 15:48.17 | WIMPy | Actually Apart from passing it on, there's nothing you can do with AOC AFAIK. |
| 15:52.26 | babak | WIMPy: I mean on normal behavior Local exchange will send AOC for calls that Asterisk transfered completely to it with ECT service ? |
| 15:53.38 | WIMPy | It can do, yes. |
| 15:57.08 | gusto | what is AOC? that monitor company? |
| 15:57.35 | [TK]D-Fender | ~aoc |
| 15:57.51 | Qwell | gusto: Advice of Charge |
| 15:57.51 | [TK]D-Fender | Advice Of Charge : telcom billing signalling |
| 15:57.57 | gusto | ok |
| 16:00.47 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 16:01.19 | WIMPy | imagines your telco sending you a TV or monitor each time you transfer a call. LOL |
| 16:01.30 | gusto | no |
| 16:07.15 | pryshebliad | is someone installed from this source https://build.opensuse.org/package/binaries/network:telephony:asterisk-12/asterisk?repository=Debian_7 ? |
| 16:08.03 | pryshebliad | Hey |
| 16:08.25 | *** join/#asterisk klootzak (~maxi@46.45.23.37) |
| 16:12.06 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 16:23.37 | *** join/#asterisk CeBe (~CeBe@brsg-4dbbadf6.pool.mediaWays.net) |
| 16:24.51 | *** join/#asterisk sdjernes (~sdjernes@184.178.41.136) |
| 16:30.11 | Stefan27 | is http://packages.asterisk.org/centos/6/asterisk-12/i386/RPMS/ likely to be updated with an rpm for 12.6.0 at some time in the future? |
| 16:53.33 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:03.36 | *** join/#asterisk smellis_werk (~smellis@2001:470:c452:2::9) |
| 17:08.21 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 17:08.22 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:20.06 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 17:28.37 | babak | Hi, I want to use Asterisk just for playing costume music on hold(crbt), may be have more than 1000 different users , with musiconhold.conf 1000 classes is very hard ,how we can define dynamic music file names ? |
| 17:29.17 | *** join/#asterisk dvorkbjel (~viskestel@li607-220.members.linode.com) |
| 17:30.17 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 17:38.55 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 17:43.49 | babak | or is it possible to pass a parameter form dialplan to music on hold class to choose specific file to play ? |
| 17:46.40 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:46.44 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:48.15 | *** part/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:48.41 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:48.41 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:54.46 | *** join/#asterisk Aamit (~amit@1.39.13.221) |
| 18:07.02 | *** part/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 18:22.05 | *** join/#asterisk novaflash (~novaflash@openvpn/corp/support/novaflash) |
| 18:27.30 | *** join/#asterisk jhlavacek (~jirka@80.215.203.3) |
| 18:27.49 | *** part/#asterisk jhlavacek (~jirka@80.215.203.3) |
| 18:29.25 | *** join/#asterisk jhlavacek (~jirka@80.215.203.3) |
| 18:39.50 | *** join/#asterisk Aamit (~amit@1.39.13.221) |
| 18:49.12 | cpugenius | babak: you can use the realtime backend for musiconhold to have it backed by a database table instead |
| 18:49.31 | cpugenius | maybe create a view based on a users table with or without defaults |
| 18:52.54 | babak | cpugenius: thx, is it possible to simply put music file path in Dial(...m(/path/music.wav)) ? |
| 18:59.02 | rmudgett | babak: no The system works with musicclass names because a musicclass could be a media stream. |
| 18:59.50 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 19:02.51 | babak | or maybe it was easyer to use simultaneous call with Dial(destination&musicExtention) |
| 19:03.40 | babak | and play music without Amswer on MusicExtention |
| 19:07.28 | *** join/#asterisk LiohAu (~LiohAu@bac69-8-88-165-4-152.fbx.proxad.net) |
| 19:11.05 | babak | <PROTECTED> |
| 19:11.23 | rmudgett | I don't know |
| 19:11.27 | *** join/#asterisk bkruse (~Adium@162.17.78.101) |
| 19:12.48 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:13.11 | *** join/#asterisk bkruse (~Adium@162.17.78.101) |
| 19:15.49 | babak | rmudgett : thank you very much for your helps, I test successfully ECT , and now I am thinking to use Asterisk+ECT for providing CRBT for LocalExchanges subscribers |
| 19:19.39 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:38.54 | zamba | where can i find the script for the different voice prompts for asterisk? i want to create a new language for it |
| 19:41.23 | *** join/#asterisk AkkerKid (~AkkerKid@12.12.180.226) |
| 19:41.47 | AkkerKid | Hi All! |
| 19:42.51 | zamba | hi you! |
| 19:43.16 | AkkerKid | is there a vicidial channel or does everyone hang out here? |
| 19:47.36 | AkkerKid | I'll take that as a "Meh" |
| 20:06.04 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 20:13.11 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:19.26 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 20:23.16 | marceloamorim | should I keep safe_asterisk running if I receive the message unable to connect to remote asterisk ( does /var/run/asterisk/asterisk.ctl exist? I remove all modules run asterisk with -vvvc and didn`t get any error |
| 20:23.59 | marceloamorim | the permission to that pid file is the same to the /etc/asterisk |
| 20:24.26 | marceloamorim | and appears just if I run asterisk -c or safe_asterisk |
| 20:40.09 | newtonr | sounds like you might have another funky asterisk process running at the same time |
| 20:41.30 | marceloamorim | well, I uninstall the version 11 and install this ast 12.6 to test the cel |
| 20:42.04 | newtonr | so what does "ps aux | grep -i asterisk" show? |
| 20:42.26 | marceloamorim | nothing |
| 20:42.32 | marceloamorim | just the grep |
| 20:43.21 | newtonr | yeah then probably still a permission issue. triple check everything! |
| 20:45.56 | marceloamorim | I didn`t find any asterisk.service to run with this systemctl |
| 20:46.22 | marceloamorim | but the [ OK ] appears, but I`ll try, ty newtonr |
| 20:47.51 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:53.23 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:54.17 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:56.53 | *** join/#asterisk sdjernes (~sdjernes@184.178.41.136) |
| 21:02.29 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:14.34 | Katty | drmessano: YOU |
| 21:14.44 | Katty | drmessano: txt me. |
| 21:14.50 | Katty | drmessano: cause i'm afk now. |
| 21:16.15 | *** join/#asterisk imox (~imox@p548434E8.dip0.t-ipconnect.de) |
| 21:19.14 | WIMPy | feels reminded of late night advertising. |
| 21:29.57 | *** join/#asterisk theron_ (~theron@173.252.71.2) |
| 21:42.09 | *** join/#asterisk areski (~areski@80.174.255.79.dyn.user.ono.com) |
| 21:43.21 | *** join/#asterisk aross42 (~aross@192-0-133-151.cpe.teksavvy.com) |
| 21:50.39 | *** join/#asterisk Skten (~Sktenau@monitor.ic.ipera.net.au) |
| 22:03.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:18.11 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-qdkhjyyhjsdsqxjp) |
| 22:24.48 | *** join/#asterisk ovoshlook (55158cfd@gateway/web/freenode/ip.85.21.140.253) |
| 22:27.50 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 22:28.37 | ovoshlook | Hello, does anyone know about default call limits at asterisk? I try to do load test between 2 asterisks via originate, but when I send 100 calls, I see maximum 38 channels at asterisk that recieved calls, another calls queued as I think. Am I right with at my guess? |
| 22:29.35 | WIMPy | There's no limit. |
| 22:29.49 | WIMPy | But chan_sip is good at blocking itself. |
| 22:29.49 | Skten | Unless there is some limit / issue in originate, there are no out of the box "limits" |
| 22:30.45 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 22:41.20 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 22:46.23 | *** join/#asterisk fornax (~fornax@85.183.43.221) |
| 22:47.19 | fornax | Hi I'm using asterisk 12.5 and need a hint how I can listen for a devstate change via ari websocket, can anyone help me? |
| 22:56.31 | *** join/#asterisk blee (~blee@50-88-4-82.res.bhn.net) |
| 23:01.00 | *** join/#asterisk wrxed (~rrittgarn@162.17.102.177) |
| 23:01.54 | wrxed | trying to enable dnsmgr in asterisk 11. I have set the enable=yes in dnsmgr.conf yet when I do dnsmgr status it says disabled... Am I missing something obvious (i've restarted asterisk since modifying the file as well as reloaded the module) |
| 23:05.33 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 23:05.33 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:08.06 | *** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf) |
| 23:10.01 | *** join/#asterisk aross42 (~aross@192-0-133-151.cpe.teksavvy.com) |
| 23:18.39 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 23:22.34 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:24.36 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 23:30.37 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |