| 00:04.18 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
| 00:09.15 | *** join/#asterisk lorsungcu_ (~lorsungcu@204.17.231.116) |
| 00:12.28 | *** join/#asterisk Baylink (~Baylink@107.144.146.22) |
| 00:27.44 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 00:31.54 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 00:35.47 | *** join/#asterisk linuxgeek (~linuxgeek@2a03:b0c0:2:d0::5f:2001) |
| 00:59.00 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 01:00.50 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:10.39 | *** join/#asterisk doop (~doop@colostomy.club) |
| 01:39.37 | *** join/#asterisk kalz (~kalz@2602:fff6:f:1::d979:58e8) |
| 01:43.42 | *** join/#asterisk pc-coholic (~martin@bundesnerdrichtendienst.de) |
| 01:43.54 | pc-coholic | good evening everyone! |
| 01:44.10 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 01:44.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:44.48 | *** join/#asterisk Navion (~navion@rivendell/users/navion) |
| 01:46.30 | pc-coholic | I've got a problem - perhaps you can tell me where I did go wrong... I have multiple "register => user:pass@server.tld/user" for different users, but with the same server. But all incoming calls appear on the first context that is found for that server in sip.conf - although each register has it's own context... |
| 01:46.36 | pc-coholic | Any clues, what black magic I need to apply, that the correct contexts are used? |
| 01:49.03 | *** join/#asterisk Navion (~navion@rivendell/users/navion) |
| 01:50.30 | *** join/#asterisk Navion (~navion@rivendell/users/navion) |
| 01:52.53 | *** join/#asterisk Navion (~navion@rivendell/users/navion) |
| 01:54.37 | *** join/#asterisk Navion (~navion@rivendell/users/navion) |
| 01:58.25 | *** join/#asterisk Navion (~navion@rivendell/users/navion) |
| 02:01.22 | *** join/#asterisk Navion (~navion@rivendell/users/navion) |
| 02:05.44 | *** join/#asterisk Navion (~navion@rivendell/users/navion) |
| 02:08.39 | *** join/#asterisk lorsungcu_ (~lorsungcu@204.17.231.116) |
| 02:27.38 | *** join/#asterisk Dovid (~Dovid@ool-2f113961.dyn.optonline.net) |
| 03:06.13 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:09.10 | *** part/#asterisk akv1 (~akv@gw.cego.dk) |
| 03:13.16 | *** join/#asterisk TSM (~the_softw@89.213.15.104) |
| 03:14.16 | *** join/#asterisk TSM (~the_softw@fw-lon2.wenn.com) |
| 03:24.54 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.77.45) |
| 03:33.25 | *** join/#asterisk TSM (~the_softw@89.213.15.104) |
| 03:34.02 | *** join/#asterisk TSM (~the_softw@fw-lon2.wenn.com) |
| 03:43.05 | *** join/#asterisk TSM (~the_softw@89.213.15.104) |
| 03:43.46 | *** join/#asterisk TSM (~the_softw@fw-lon2.wenn.com) |
| 03:44.26 | *** join/#asterisk Dovid (~Dovid@ool-2f113961.dyn.optonline.net) |
| 03:45.22 | *** join/#asterisk TSM (~the_softw@89.213.15.104) |
| 03:46.15 | *** join/#asterisk TSM (~the_softw@fw-lon2.wenn.com) |
| 03:47.13 | *** join/#asterisk TSM (~the_softw@89.213.15.104) |
| 03:47.45 | *** join/#asterisk TSM (~the_softw@fw-lon2.wenn.com) |
| 03:53.10 | *** join/#asterisk aruntomar (~aruntomar@49.248.195.29) |
| 03:58.35 | *** join/#asterisk TSM (~the_softw@89.213.15.104) |
| 03:59.26 | *** join/#asterisk TSM (~the_softw@fw-lon2.wenn.com) |
| 04:00.22 | ChannelZ | OMG TSM WTF |
| 04:05.20 | *** join/#asterisk lorsungcu_ (~lorsungcu@50.137.202.86) |
| 04:10.24 | *** join/#asterisk TSM (~the_softw@89.213.15.104) |
| 04:11.05 | *** join/#asterisk TSM (~the_softw@fw-lon2.wenn.com) |
| 04:17.35 | *** join/#asterisk TSM (~the_softw@89.213.15.104) |
| 04:18.15 | *** join/#asterisk TSM (~the_softw@fw-lon2.wenn.com) |
| 04:26.32 | *** join/#asterisk TSM (~the_softw@89.213.15.104) |
| 04:27.05 | *** join/#asterisk TSM (~the_softw@fw-lon2.wenn.com) |
| 04:59.01 | *** join/#asterisk lorsungcu__ (~lorsungcu@209-173-236-30.usfamily.net) |
| 05:13.30 | Penguin | newtonr, pabelanger, malcolmd, qwell, file |
| 05:37.58 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 05:49.35 | *** join/#asterisk angryuser (~angryuser@LCaen-656-1-198-99.w193-251.abo.wanadoo.fr) |
| 05:56.33 | *** join/#asterisk CrashHD (~na@104.220.23.22) |
| 06:20.36 | [TK]D-Fender | <pc-coholic> I've got a problem - perhaps you can tell me where I did go wrong... I have multiple "register => user:pass@server.tld/user" for different users, but with the same server. But all incoming calls appear on the first context that is found for that server in sip.conf - although each register has it's own context... <- nope |
| 06:20.36 | Penguin | [TK]D-Fender: its |
| 06:21.06 | [TK]D-Fender | Penguin, Nice script... |
| 06:21.30 | [TK]D-Fender | pc-coholic, REGISTER has no such thing as "context" |
| 06:21.58 | [TK]D-Fender | pc-coholic, The existance of a PEER entry has no direct relation to any register statement. |
| 06:22.11 | [TK]D-Fender | pc-coholic, they are completely seperate entities |
| 06:22.53 | [TK]D-Fender | pc-coholic, * will take the first peer that matches the incoming call. That same one successfully matches all of those calls based on how you configured it. that is the problem. |
| 06:27.50 | *** join/#asterisk mokmeister (~quassel@95.45.40.109) |
| 06:31.22 | *** join/#asterisk CrashHD (~na@104.220.23.22) |
| 06:58.53 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 07:05.05 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 07:05.25 | *** join/#asterisk zblk (~andrey@92.53.115.234) |
| 07:12.02 | *** join/#asterisk zblk (~andrey@92.53.115.234) |
| 07:18.10 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:19.42 | *** join/#asterisk RagBal (~RagBal@54694E34.cm-12-2b.dynamic.ziggo.nl) |
| 07:29.44 | *** join/#asterisk ghostlines (~ghostline@92.109.90.61) |
| 07:37.22 | RagBal | When forwarding a call to an external number I want to sent the phone number of the caller as a p asserted id, this is because our sip provider doesn't allow foreign CIDs but does allow passing the p asserted id |
| 07:37.32 | RagBal | How would I set this up? |
| 07:46.46 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:47.19 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 08:10.27 | *** part/#asterisk zblk (~andrey@92.53.115.234) |
| 08:19.50 | ChannelZ | See https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information |
| 08:20.04 | *** join/#asterisk R4v3n (~R4v3n@il267-1-78-206-224-3.fbx.proxad.net) |
| 08:24.34 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 08:24.59 | *** join/#asterisk Cynagen (~cynagen@ip72-208-223-60.ph.ph.cox.net) |
| 08:31.55 | *** join/#asterisk sommarnatt (~sommarnat@kungsbacka.oderland.com) |
| 08:37.07 | *** join/#asterisk zblk (~andrey@92.53.115.234) |
| 08:43.46 | *** join/#asterisk CustosL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 08:47.40 | *** join/#asterisk CeBe (~CeBe@port-92-200-19-252.dynamic.qsc.de) |
| 08:52.02 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:55.55 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-xodmnhdqyedrluhs) |
| 09:03.08 | *** join/#asterisk jhlavacek (~jirka@ata35-1-78-208-220-3.fbx.proxad.net) |
| 09:11.51 | *** join/#asterisk Akuma (~Akuma@apn-5-60-2-81.dynamic.gprs.plus.pl) |
| 09:28.20 | *** join/#asterisk gwenloh (c1efc0b2@gateway/web/freenode/ip.193.239.192.178) |
| 09:28.25 | gwenloh | Hello |
| 09:51.36 | *** join/#asterisk areski (~areski@239.158.14.37.dynamic.jazztel.es) |
| 09:57.19 | *** join/#asterisk pawiecki (~pawiecki@217.97.180.1) |
| 10:03.30 | pawiecki | good morning! :) I have a question - if asterisk sees my phone's IP address, but status is unreachable - what can be the problem? |
| 10:04.14 | WIMPy | Someting in between. |
| 10:04.21 | sommarnatt | Usually NAT |
| 10:04.44 | sommarnatt | or firewall |
| 10:04.45 | sommarnatt | :) |
| 10:08.42 | pawiecki | that's what i was thinking of, but why does it display the correct ip of phone? So they can see each other, but can't communicate through SIP? |
| 10:13.57 | pawiecki | sorry for noobs questions, i'm new to this and trying to figure it out step by step. |
| 10:14.37 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 10:41.29 | R4v3n | hello, I retest today :D Somebody use or already used h323plus ? Thank you ! :) |
| 10:44.16 | *** join/#asterisk akv1 (~akv@195.88.36.18) |
| 10:49.02 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 10:52.13 | akv1 | Hey, I've just setup an Asterisk 13 server to replace an old 1.8 - but I have a problem as the AGENT() function has password deprecated - I use the agent.conf to define agents for our support-phones but all logins we do manually in the dialplan where an agent is added to the queue with correct priority after a manual login with Read(AGENT_USERID,agent-user,,,,5) and authenticate(${AGENT(${AGENT_USERID}:password)},j)... are there any way of doing this with Asteris |
| 11:00.26 | *** join/#asterisk workingcats (~workingca@212.122.48.77) |
| 11:03.45 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:16.40 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:20.19 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 11:28.08 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 11:59.16 | *** join/#asterisk areski (~areski@81.Red-83-53-30.dynamicIP.rima-tde.net) |
| 12:02.24 | *** join/#asterisk R4v3n (~R4v3n@il267-1-78-206-224-3.fbx.proxad.net) |
| 12:08.31 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 12:11.05 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 12:11.42 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 12:12.23 | dan_j | Hi. Is there a way to have tenanted call parking? To ensure users from one company can't answer parked calls from other companies. |
| 12:36.26 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 12:37.56 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-19-252.dynamic.qsc.de) |
| 12:45.12 | marceloamorim | hello, I`m trying to find way to call to a customer and after he answer I put some audio for him, so far I`m using the file.call if I use application Dial works pretty well to call to the customer, but didn`t put audio after that |
| 12:45.33 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 12:46.10 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:157:3142:bcbc:6bc9) |
| 12:47.44 | *** join/#asterisk calum_ (~calum_@host81-154-106-148.range81-154.btcentralplus.com) |
| 12:48.21 | marceloamorim | ps: I forgot to specified that I`m calling the customer and to my extension, but I wish the customer listen the audio before he talk to me |
| 12:55.56 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 13:00.48 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:22.04 | *** join/#asterisk Draecos (~Draecos@106-69-20-37.dyn.iinet.net.au) |
| 13:30.14 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 13:32.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:38.29 | *** join/#asterisk zafu (~pif@217-162-81-21.dynamic.hispeed.ch) |
| 13:38.47 | zafu | hi, how can I use readline with asterisk at the command line? |
| 13:44.08 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:44.29 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:47.50 | [TK]D-Fender | zafAs in? |
| 13:48.10 | [TK]D-Fender | zafu: As in? |
| 13:49.40 | zafu | [TK]D-Fender: I just installed 13.0.0 in my debian system and backward search from the CLI is gone (CTRL-R) |
| 13:49.55 | zafu | or any kind of line editing feature |
| 13:50.16 | [TK]D-Fender | Search from where? |
| 13:50.38 | zafu | you know like at the shell, backward history search |
| 13:50.54 | [TK]D-Fender | that is not an Asterisk issue then. |
| 13:51.10 | zafu | well an asterisk CLI issue then |
| 13:51.21 | [TK]D-Fender | No, OS shell isn't asterisk. |
| 13:51.43 | [TK]D-Fender | are you talking about LINUX CLI, or ASTERISK CLI? |
| 13:51.49 | zafu | * CLI of course |
| 13:52.09 | [TK]D-Fender | * doesn't seem to have this |
| 13:52.24 | [TK]D-Fender | that is compile-in library |
| 13:52.31 | zafu | * used to have editline, no? |
| 13:52.41 | [TK]D-Fender | so far ... nope |
| 13:53.09 | [TK]D-Fender | Just checked 2 different major OS versions, and 2 versions of * |
| 13:53.22 | [TK]D-Fender | 1.8 vs 13 |
| 13:53.32 | zafu | never mind then |
| 13:58.57 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 13:58.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:59.48 | *** join/#asterisk CeBe1 (~CeBe@wlan-141-23-90-62.tubit.tu-berlin.de) |
| 14:01.25 | wdoekes | zafu: yes, editline used to built in, but is now taken from libeditline if available |
| 14:02.05 | wdoekes | don't know about backwards search though |
| 14:04.03 | zafu | I see, libedit2 is installed yet no more CTRL-R |
| 14:05.17 | tzafrir | zafu, is it used? Try ldd /usr/sbin/asterisk |
| 14:05.50 | zafu | tzafrir: it is |
| 14:06.05 | zafu | maybe it needs a configuration now |
| 14:06.10 | zafu | key mapping of sorts |
| 14:06.48 | zafu | in biggie, just curious about the change |
| 14:06.52 | zafu | s/in/no/ |
| 14:14.25 | *** join/#asterisk Draecos (~Draecos@106-68-101-185.dyn.iinet.net.au) |
| 14:16.16 | *** join/#asterisk g_r_eek (~g_r_eek@46.227.60.214) |
| 14:19.29 | bulkorok | ${ANSWEREDTIME} = billsec or duration ?! |
| 14:21.13 | hrnt | i am curious, how common it is to use wideband codecs for internal calls? |
| 14:22.44 | tomodachi | hrnt: well depends on your hardware I guess |
| 14:22.51 | [TK]D-Fender | Overall not much, but for those specifically ordering those kinds of phones, it isn't that rare either |
| 14:23.02 | *** join/#asterisk CeBe1 (~CeBe@wlan-141-23-90-62.tubit.tu-berlin.de) |
| 14:23.23 | tomodachi | in general i think g711 used internally offers quite good quality honestly. Better than any cell phone call i ever make |
| 14:23.58 | tomodachi | and i work 500 Meters from Ericsson, |
| 14:24.23 | hrnt | we have been using internally G.722 for quite some time. i thought HD voice was a norm nowadays but when i google for random voip stuff it seems that people are still using G.711 or G.729. |
| 14:25.00 | tomodachi | oldies but goldies |
| 14:25.01 | hrnt | (we are 99% polycom, mostly IP550 and IP650 and now VVX500) |
| 14:25.29 | tomodachi | we use mostly ata converters for our ip telephony with some Thomson sip phones, but the codec support is quite sparse |
| 14:26.02 | tomodachi | but i think we are going to use a lot more vide conferencing soon , im implementing a jitsi video bridge |
| 14:26.10 | tomodachi | its just so simple |
| 14:26.18 | [TK]D-Fender | hrnt: Using G.729 on internal calls makes no sense whatsoever. |
| 14:26.26 | tomodachi | runs from a browsers, allows both video and audio and uses the opus codec |
| 14:26.40 | hrnt | [TK]D-Fender: well, our "internal" calls can be between sites |
| 14:26.50 | hrnt | e.g. finland and north carolina |
| 14:26.52 | [TK]D-Fender | hrnt: Get a bigger pipe :) |
| 14:27.13 | hrnt | [TK]D-Fender: yeah well actually we are not using G.729 for anything :) but it seems that some other people are. |
| 14:27.17 | [TK]D-Fender | hrnt: How are you passing calls between the sites? |
| 14:27.32 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:27.32 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:29.23 | *** join/#asterisk danjenkins__ (~dan@nat/digium/x-mkjaddgnslpippmq) |
| 14:29.38 | hrnt | [TK]D-Fender: right now are just running one asterisk in the states and our phones talk "directly" with it |
| 14:30.47 | *** join/#asterisk CeBe1 (~CeBe@wlan-141-23-90-62.tubit.tu-berlin.de) |
| 14:31.44 | hrnt | tomodachi: have you tried the wideband codecs? in our opinion the different between G.722 and G.711 is very noticeable. |
| 14:31.49 | hrnt | difference |
| 14:32.53 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 14:34.15 | tomodachi | hrnt: our phones dont support so not fully |
| 14:34.25 | tomodachi | but ive tried it with a soft phone into the asterisk echo application |
| 14:34.35 | tomodachi | but generally speaking i think g711 is "good enough" |
| 14:35.51 | [TK]D-Fender | hrnt: yup, that does cost more BW doing it that way |
| 14:36.49 | hrnt | [TK]D-Fender: true, but right now we are not limited by BW so why not use it all :) |
| 14:37.09 | hrnt | tomodachi: yeah, i guess the phones are usually the bottleneck |
| 14:37.30 | hrnt | i have a linksys SPA942 and polycom IP550 next to each other and the voice quality difference between them is HUGE |
| 14:37.57 | [TK]D-Fender | hrnt: I was referring to your mention of G.729 :) |
| 14:38.19 | hrnt | a wideband codec would be completely wasted on the linksys |
| 14:38.20 | [TK]D-Fender | hrnt: I skipped over the part when you followed that you aren't actually using it... |
| 14:38.41 | [TK]D-Fender | SPA vs Polycom at G.711 is huge |
| 14:38.49 | [TK]D-Fender | has nothing to do with the codec.... |
| 14:38.54 | hrnt | [TK]D-Fender: indeed |
| 14:38.55 | [TK]D-Fender | Polycom > All |
| 14:39.24 | hrnt | SPA on G.722 (even if it supported it) would still be much worse than polycom on G.711. |
| 14:39.39 | file | yeah... |
| 14:40.29 | hrnt | it is really weird reading some reviews for the SPA942. "The voice quality of the SPA-942 is very good". no, it is not :P |
| 14:40.35 | [TK]D-Fender | Well... maybe not "huge", but truly noticable. |
| 14:41.00 | [TK]D-Fender | That might be fair. The SPA's are "decent". Polycom just jump to "really good" |
| 14:42.11 | hrnt | well, i usually use the speakerphone and there at least i'd say the difference is huge |
| 14:42.15 | *** join/#asterisk Draecos (~Draecos@106-68-101-185.dyn.iinet.net.au) |
| 14:43.10 | [TK]D-Fender | Oh THAT is another game altogether |
| 14:43.20 | [TK]D-Fender | I was at least restricting myself to handset quality |
| 14:43.37 | [TK]D-Fender | [09:38][TK]D-FenderPolycom > All |
| 14:46.08 | *** join/#asterisk zerohalo (~zerohalo@2601:6:f80:224:d5e7:bdaa:54e3:f6a) |
| 14:47.28 | Cuzner | you can pry my old clunker of a 1500 from my cold dead hands... |
| 14:52.30 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 14:53.24 | TSM | any UK peeps looking to buy a sangoma A101D PCI-E card? |
| 14:55.20 | *** join/#asterisk burnbrighter (~kurt@50-79-211-169-static.hfc.comcastbusiness.net) |
| 14:56.12 | [TK]D-Fender | Cuzner: Nah, the incinerator can handle the both of you ;) |
| 14:57.20 | wdoekes | zafu: $ echo 'bind ^R em-inc-search-prev' > ~/.editrc; EDITRC=~/.editrc asterisk -rv # although the search does not behave at all over here (old asterisk though) |
| 14:57.43 | Cuzner | i pretty much have every model of poly... SPIP, SSIP, VVX |
| 14:57.55 | Cuzner | I still like my old 1500 over the new VVXs |
| 15:01.13 | Cuzner | oh actually, i don't have any HDX's |
| 15:01.18 | Cuzner | I should get my hands on one |
| 15:01.53 | [TK]D-Fender | Cuzner: 1500 what exactly? |
| 15:02.02 | zafu | wdoekes: thanks this works :) |
| 15:02.13 | Cuzner | oh i'm mixxing the Voicestations too... damn |
| 15:02.18 | Cuzner | missing even |
| 15:02.24 | Cuzner | [TK]D-Fender: VVX 1500 |
| 15:02.30 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 15:02.37 | zafu | strange that one has to EDITRC= |
| 15:03.02 | Cuzner | it's pretty ancient compared to the new VVX phones |
| 15:03.16 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 15:03.18 | [TK]D-Fender | Cuzner: I'd heard there were a lot of quirks on teh VVX vs the SPIP. What is your take on this? |
| 15:04.04 | Cuzner | Well, the SPIPs are rock solid, but they're old... People like colourful touchy screens. |
| 15:04.39 | Cuzner | and I don't think Poly's doing anything besides maintenance releases for SPIP phones now, so I consider them EOL. |
| 15:04.47 | Cuzner | (firmware that is) |
| 15:04.51 | [TK]D-Fender | Cuzner: I means in terms of VVX being problematic by comparison. I have had SPIP's 301,320,335,430,501,600,601 here |
| 15:05.09 | Cuzner | oh, 601 w/ sidecar is a beast of a phone |
| 15:05.12 | Cuzner | love it |
| 15:05.23 | [TK]D-Fender | Cuzner: Yup, after SIP 1.6.2 FW taht is :) |
| 15:05.45 | [TK]D-Fender | Cuzner: My 6XX here are all on 3.1.8 mind you. |
| 15:05.48 | [TK]D-Fender | (now) |
| 15:05.58 | wdoekes | zafu: new since 11 (?). in 1.8 it didn't el_source (read editrc) at all |
| 15:06.34 | zafu | wdoekes: hmm, in 11 it used to work without any editrc conf at all |
| 15:06.42 | zafu | since CTRL-R is a default |
| 15:06.47 | Cuzner | our most popular SPIP around here are the 450 and 550, i have a 550 right next to me running SIP 4.0.7 |
| 15:08.02 | Cuzner | I'm actually putting the new 5.1.3 (we already have 5.2.0 released) on our firmware repo right now. |
| 15:08.12 | wdoekes | zafu: just telling you what the source says, and it says that el_source is only called if the mentioned env is set |
| 15:08.19 | [TK]D-Fender | 550 was really a waste. Not much cheaper than a 650 and no sidecar, USB, only 4 line keys.... value loser model |
| 15:08.19 | *** join/#asterisk CeBe (~CeBe@port-92-200-19-252.dynamic.qsc.de) |
| 15:08.30 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-nyowgupanzapturt) |
| 15:09.53 | *** join/#asterisk Baylink (~Baylink@66.87.148.183) |
| 15:10.25 | [TK]D-Fender | Only reason to pick it is to shut up someone from middle management you really don't want to give a 650 to and save a "few" bucks on. |
| 15:10.40 | *** join/#asterisk angryuser (~angryuser@LCaen-656-1-198-99.w193-251.abo.wanadoo.fr) |
| 15:11.09 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-19-252.dynamic.qsc.de) |
| 15:11.30 | [TK]D-Fender | I used to use a 600 at my desk, but switched to a 335 a few years ago because the font & backlight were actually readable at a distance and it saved some space on my desk. Call handling isn't as night though.... |
| 15:12.01 | [TK]D-Fender | 3+line 4-softkey Polycom = AWESOME call handling. |
| 15:12.25 | [TK]D-Fender | Split/Join adhoc conferencing = the bestest. |
| 15:12.38 | Cuzner | hehe, we use RMXs |
| 15:13.25 | [TK]D-Fender | I have Aastra 6739i's for my CSR's because of the very visible colour screen for BLF & their IDLE XML notifications for live Queue status. |
| 15:13.36 | [TK]D-Fender | RMX? |
| 15:14.33 | Cuzner | Weird, I'm trying to find the product page on poly's site, and I can't... |
| 15:14.50 | Cuzner | http://www.polycom.com/products-services/realpresence-platform/realpresence-collaboration-servers.html |
| 15:16.05 | [TK]D-Fender | Ah, I have a decrepit Polycom VSX 7000 here.... |
| 15:16.28 | Cuzner | http://www.polycom.com/products-services/hd-telepresence-video-conferencing/realpresence-room/realpresence-group-series.html |
| 15:16.37 | Cuzner | we use those as well, mostly the 300 and 500 |
| 15:17.20 | Qwell | What's that cost, like $16M? |
| 15:17.52 | [TK]D-Fender | Qwell: the 500 looks like an amped up version of my VSX7000. that thing cost about $7 in 2007 |
| 15:17.56 | [TK]D-Fender | $7G |
| 15:18.15 | Qwell | <surprise type="feigned"/> |
| 15:18.17 | Cuzner | yeah, i have no idea what these things cost, i don't do the purchasing :) |
| 15:18.17 | [TK]D-Fender | So I imagine it's cost much less than $16M ;) |
| 15:18.30 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 15:18.34 | Cuzner | I'm just the monkey that turns the wrench |
| 15:18.52 | file | money? that's just a number on paper |
| 15:18.57 | file | or plastic, depending on country |
| 15:19.05 | Qwell | file: beat me to it... |
| 15:19.06 | Cuzner | yeah, it's polymer here :) |
| 15:19.09 | [TK]D-Fender | Qwell: $11000 @ CDW |
| 15:19.33 | Qwell | [TK]D-Fender: a webcam costs me like $24 at Walmart :p |
| 15:19.37 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:157:3142:bcbc:6bc9) |
| 15:19.39 | Cuzner | [TK]D-Fender: what's $11k? the RPGS 700? |
| 15:19.56 | [TK]D-Fender | Cuzner: RP 500 |
| 15:20.03 | Cuzner | oh wow, pricey |
| 15:20.08 | [TK]D-Fender | http://www.cdw.ca/shop/products/Polycom-RealPresence-Group-500-720p-video-conferencing-kit/2858595.aspx |
| 15:20.13 | [TK]D-Fender | And sorry, that $CAD |
| 15:20.32 | [TK]D-Fender | So probably around $9000 USD |
| 15:20.43 | Cuzner | I'm pretty sure we have a deal with Polycom to test out new stuff w/ free demos and stuff |
| 15:21.12 | file | I just get to have Digium phones ;( |
| 15:21.41 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 15:21.55 | [TK]D-Fender | file: Oh you poor poor sod :( |
| 15:21.59 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 15:23.11 | Cuzner | consoles file |
| 15:23.12 | Qwell | file: Want my Snom? Or my Polycom? Or my Grandstream? Or one of my Aastras? Or my Obihai? |
| 15:23.17 | file | Qwell, not really. |
| 15:23.17 | Cuzner | there there |
| 15:23.21 | Qwell | ooo, or my Cisco?! |
| 15:23.32 | Qwell | You know you want a Selsius badged phone! |
| 15:23.32 | Cuzner | Hey, grandstream makes an alright ATA iirc |
| 15:23.40 | file | I actually keep meaning to pack one of mine away, I only need my D70 |
| 15:23.47 | [TK]D-Fender | remembers to the golden years of Qwell's chan_skinny.so botnet :) |
| 15:23.50 | [TK]D-Fender | the* |
| 15:24.01 | file | uses softphones |
| 15:24.15 | Qwell | file: I've never shown you pictures of my phone wall. |
| 15:24.26 | file | Qwell, nope! |
| 15:24.26 | [TK]D-Fender | Cuzner: GS's had no real dialplan, etc. Lots of shortcomings |
| 15:24.28 | [TK]D-Fender | ~gs |
| 15:24.28 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:24.30 | Qwell | I should take one some time. |
| 15:25.34 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 15:26.02 | Cuzner | I haven't touched anything GS in nearly a decade... I've been all poly or cisco since I joined TPN. |
| 15:26.11 | Cuzner | and Ciscos make me rage |
| 15:26.22 | Cuzner | but I can make 'em work :) |
| 15:26.44 | *** join/#asterisk War_Bear (~War_Bear@shiftingshadow.warbear.co.uk) |
| 15:30.05 | [TK]D-Fender | Cisco never really cared about "standards" much that way. "CCM or Bust". The SPA's still exhibit their Sipura heritage which is a good thing. Those are at least compliant, and decent for the price (which is hard to beat in many markets) |
| 15:31.04 | file | Cisco actually has many SIP stacks |
| 15:31.16 | file | some are more standard than others |
| 15:32.46 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-19-252.dynamic.qsc.de) |
| 15:33.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 15:34.01 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-kddmtibdqbuykrog) |
| 15:34.01 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:37.25 | Cuzner | it's not so much the stack that messes me up, it's the configuration sans CUCM |
| 15:37.55 | Cuzner | there is no real secure way to deliver phone configs... stuck with TFTP without a CUCM. |
| 15:38.47 | eirirs | a cheap cisco 3825 router with CME solved all my cisco-rage with phones |
| 15:38.53 | eirirs | and put it on trunk to asterisk |
| 15:38.55 | eirirs | win |
| 15:39.06 | eirirs | (ebayd a 3825) |
| 15:39.52 | Cuzner | eirirs: yeah, we provision phones without any on-premise devices (besides the phones themselves) |
| 15:40.13 | Cuzner | we also have to account for "hotelled" users. |
| 15:40.46 | Cuzner | secure off-site phone provisioning is our biggest hangup with Ciscos. |
| 15:41.27 | hrnt | Cuzner: have you had any problems with the 550 running 4.0.7? |
| 15:41.41 | hrnt | Cuzner: specifically random "lag" spikes where the phone interface freezes. shows up as a spike in the CPU graph. |
| 15:42.18 | Cuzner | I don't really test the phones that much, but I know we have a ton of 450s and 550s in the wild running it without complaints. |
| 15:42.40 | Cuzner | i have a 450 next to me running 4.0.7 right now. |
| 15:44.06 | hrnt | Cuzner: ok. we had a problem where the 550s and 650s had these 1 second "pauses" when it stopped updating the screen. sometimes they even crashed and rebooted themselves. we finally downgraded to 3.x which solved the problem. |
| 15:44.57 | rrittgarn | hrnt: I had that issue but worse, i had accidentally set my qualifyfreq to 0 and it effectively dos'd the phone |
| 15:45.04 | rrittgarn | aastras were effected but polycoms were |
| 15:45.52 | Cuzner | Yeah, to be honest I wouldn't know of any of these problems if our users were experiencing them. Our support department probably does exactly what you did if they have complaints after a SIP upgrade... Our portal allows us to configure firmware versions on a per-device basis. |
| 15:46.22 | Cuzner | I personally haven't had the problem you're descibing when i'm testing stuff. |
| 15:46.46 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:46.59 | hrnt | rrittgarn: ok. i have my qualifyfreq set to 60... |
| 15:47.29 | hrnt | Cuzner: we saw the CPU spikes on all phones but only like 10% of them crashed |
| 15:47.49 | Cuzner | weird, that would suck to troubleshoot |
| 15:48.26 | hrnt | Cuzner: yep, it did :) oh well, 3.x does everything we need to we didn't really lose anything by downgrading. |
| 15:49.21 | hrnt | -to+so |
| 15:52.04 | Katty | hello my asterisk does not work at all how to fix pls |
| 15:52.08 | Katty | is urgent thx |
| 15:52.16 | Cuzner | 407 is the latest SPIP firmware i believe, no? |
| 15:53.24 | [TK]D-Fender | Cuzner: For standard SIP, yes |
| 15:54.00 | Cuzner | wait, no, i think 411 is |
| 15:56.38 | [TK]D-Fender | 411 lists as Lync, not standard |
| 15:56.44 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:da4:8219:34ff:fecf:17f0) |
| 15:57.11 | Cuzner | I know 4.1.8, 5.1.3 and 5.2.0 are all VVX-only |
| 15:57.31 | Cuzner | I'm downloading 411 split now to see what's in it. |
| 16:02.47 | [TK]D-Fender | Cuzner: 411 = Lync-only, not SIP |
| 16:04.44 | Cuzner | from the readme - The UCS 4.1.1 distributable files contain both the BootROM (now named |
| 16:04.44 | Cuzner | Updater) and SIP software. |
| 16:05.56 | Cuzner | it's probably just the 4.0.7 release + lync support |
| 16:10.03 | Cuzner | buuut from the release notes - Polycom UC Software 4.1.1 is for use with Microsoft® |
| 16:10.03 | Cuzner | LyncTM Server and using a single registered line |
| 16:10.03 | Cuzner | only |
| 16:10.43 | Cuzner | Going to see what happens when i try to reg on a * box, just for giggles. |
| 16:12.15 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 16:12.29 | Katty | rearranges all of file's cisco trunk routing |
| 16:12.37 | file | hi |
| 16:12.37 | Katty | randomly creates new vlan id numbers |
| 16:12.41 | *** join/#asterisk Akuma (~Akuma@apn-5-60-2-81.dynamic.gprs.plus.pl) |
| 16:12.47 | Katty | removes the management IP from file's vlan |
| 16:12.56 | Chainsaw | disables spanning tree |
| 16:13.00 | Katty | ohhhhh |
| 16:13.05 | Katty | Chainsaw: clever boy ^_^ |
| 16:13.57 | hrnt | speaking of lync, has anybody actually used it as a PBX? |
| 16:14.06 | Katty | Chainsaw: watch cisco auto detect loop and disable the port :/ |
| 16:14.16 | WIMPy | Exchange spanning trees for christmas trees? |
| 16:14.43 | Katty | hrnt: no, but i got one from toys r us that says fisher price on the side. |
| 16:14.48 | Katty | hrnt: that's rpobably similar ;> |
| 16:15.05 | Cuzner | hrnt: we have customers that wanted to use our UC software with their existing lync boxes. I believe we just through a opensips box in front of it to proxy. |
| 16:15.50 | Chainsaw | hrnt: It's like Exchange. A bit naive and in need of a gatekeeper box between it and the big bad internet |
| 16:16.29 | hrnt | we use lync purely for instant messaging (well, the office 365 version). considering how bad that is... |
| 16:16.43 | hrnt | i can only imagine how fun it is as a PBX |
| 16:20.47 | *** join/#asterisk lorsungcu_ (~lorsungcu@c-50-137-202-86.hsd1.mn.comcast.net) |
| 16:32.26 | *** join/#asterisk lorsungcu_ (~lorsungcu@209-173-236-30.usfamily.net) |
| 16:41.49 | *** join/#asterisk theron (~theron@199.201.65.133) |
| 16:52.34 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-osoknkghxqkmhbhi) |
| 17:00.32 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 17:07.19 | *** join/#asterisk kayfox (~kayfox@orca.zerda.net) |
| 17:09.44 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 17:11.27 | *** join/#asterisk newtonr (~newtonr@173-21-90-203.client.mchsi.com) |
| 17:11.28 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 17:11.40 | *** join/#asterisk lorsungcu_ (~lorsungcu@c-50-137-202-86.hsd1.mn.comcast.net) |
| 17:19.43 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:22.03 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:25.39 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-ugfnirznzjeqmifo) |
| 17:26.27 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-bwrykqtiwfumxwnu) |
| 17:32.31 | *** join/#asterisk bluesoul|work_l (~bluesoul@unaffiliated/bluesoul) |
| 17:46.36 | *** join/#asterisk danjenkins__ (~dan@nat/digium/x-rfqwjmnrjzanqngx) |
| 17:50.26 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 18:04.05 | pc-coholic | [TK]D-Fender: Oh, I see... I never realized, that this was "by design"... Well - gotta change my dialplan then :) Thanks for pointing this out! |
| 18:06.29 | *** join/#asterisk NirS (~NirS@5.28.148.71) |
| 18:14.07 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:157:3142:bcbc:6bc9) |
| 18:22.03 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 18:40.22 | *** join/#asterisk riess82 (~riessma@88-117-113-96.adsl.highway.telekom.at) |
| 18:40.39 | *** join/#asterisk sdjernes (~sdjernes@184.178.41.136) |
| 18:43.35 | *** join/#asterisk jepperl (026efe6a@gateway/web/cgi-irc/kiwiirc.com/ip.2.110.254.106) |
| 18:46.05 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 18:46.19 | jepperl | Hello :) I am having some issues dialing outbound. What is happening is that after i send my INVITE to the sip-provider, i get "Proxy Authentication Required" back. I then resend my INVITE with the Proxy-Authorization header. Usually, this does the trick, but this time i just get "Proxy Authentication Required" back. This loops a few times before t |
| 18:46.19 | jepperl | he call is failing. I assume my authorization has failed since i keep getting the "Proxy Authentication Required" back, but i don't know how |
| 18:48.28 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 19:01.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:06.44 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 19:10.42 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:13.22 | *** join/#asterisk bergw (~bergw@173-163-61-46-cpennsylvania.hfc.comcastbusiness.net) |
| 19:14.49 | bergw | hey everyone. I'm having a little trouble gathering rtpqos stats. rtt is always zero. p2p is off. anything obvious i might be missing? |
| 19:15.31 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-suzvqaikhjmwfmov) |
| 19:19.13 | *** join/#asterisk fadumpt (~fadumpt@66-169-39-177.dhcp.gnvl.sc.charter.com) |
| 19:20.34 | fadumpt | Looking for a point in the right direction, because Google hasn't helped so far. I added a time condition that rings a different set of numbers between 7am and 930am (includes a cell phone number) and it seems to have disabled call recordings for that time period. |
| 19:21.06 | [TK]D-Fender | ~freepbx |
| 19:21.07 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:21.09 | [TK]D-Fender | fadumpt: ^^^^^ |
| 19:21.09 | Penguin | You probably started recording after that point. |
| 19:21.42 | fadumpt | Penguin, what do you mean? |
| 19:22.03 | Penguin | I mean you had GotoIfTime() before MixMonitor() |
| 19:22.12 | Penguin | So it went somewhere else before you started recording. |
| 19:22.27 | [TK]D-Fender | Penguin: FreePBX... |
| 19:22.34 | *** join/#asterisk lorsungcu_ (~lorsungcu@c-50-137-202-86.hsd1.mn.comcast.net) |
| 19:22.49 | Penguin | He asked in an asterisk channel, so I gave an asterisk answer. |
| 19:22.57 | Penguin | If he's not using asterisk, that's on him. |
| 19:23.23 | fadumpt | I'm fine with those conditions. |
| 19:23.49 | fadumpt | So I need to add a line to the dialplan of the time condition? (Since I'm using Freepbx and it created it's own entry) |
| 19:24.11 | Penguin | It's just my guess that your recording application comes after the time condition. When the time condition is true, it never hits the recording app. |
| 19:25.58 | Penguin | You can either move the recording method before the time condition so both dialing routines are included, or add a recording method in the portion of the dial plan routine which is used conditionally. |
| 19:26.21 | Penguin | I would move my MixMonitor() to before my ExecIfTime() or GotoIfTime(). But that's just me. |
| 19:28.32 | fadumpt | So the Time Conditions entries and the Call Recording entries are all in the same file? (I really need to learn more Asterisk....FreePBX has made me soft :) ) |
| 19:28.40 | Penguin | no clue |
| 19:28.50 | fadumpt | oh okay |
| 19:28.55 | fadumpt | well that gives me something to look for. |
| 19:28.57 | Penguin | uses asterisk |
| 19:29.15 | fadumpt | FreePBX is so much easier...until something doesn't work.... |
| 19:29.24 | Penguin | In asterisk, it's a flat file. |
| 19:29.25 | [TK]D-Fender | fadumpt: Yes, extensions_additional.conf. Which you are never supposed to touch. You need to use FreePBX properly to have it implement what you want it to do. |
| 19:29.31 | Penguin | You can optionally include other files. |
| 19:30.09 | Penguin | In FreePBX, I think it does nothing to the base file and includes about 20 other files that it does its work in. |
| 19:31.29 | Penguin | I think asterisk is easier, you think FreePBX is easier... |
| 19:31.37 | Penguin | One of us is wrong. |
| 19:32.01 | *** join/#asterisk roswell (roswell@home.roswellsite.ru) |
| 19:32.40 | fadumpt | it's not a wrong, it's just that I was lazier in the beginning. I tried to learn Asterisk but FreePBX got me working faster. |
| 19:33.00 | Penguin | I suppose I can see that. |
| 19:35.07 | fadumpt | Okay, I see time conditions listed after macro-systemrecording |
| 19:35.39 | roswell | freebpx, elastix, etc suck. they are simply too limited |
| 19:35.57 | roswell | or let's say, limiting |
| 19:36.45 | fadumpt | yeah, I kind of always knew that but I never thought I needed "those advanced features" ... but not a not so advanced feature kinda went missing on me. |
| 19:36.53 | fadumpt | not=now |
| 19:37.19 | roswell | well actually you're now trying to learn freepbx instead of asterisk ) |
| 19:38.58 | *** join/#asterisk lorsungcu__ (~lorsungcu@209-173-236-30.usfamily.net) |
| 19:39.04 | fadumpt | yeah |
| 19:39.14 | roswell | sorry guys i got myself in the middle, would anyone enlighten me on what was the question? ) |
| 19:39.21 | fadumpt | :) |
| 19:39.47 | fadumpt | I created a time condition that included a cell phone (if that matters) and now during that period of time, call recordings don't show up |
| 19:40.22 | fadumpt | so 7am to 930am there will be no call recording icon and then after 930am when the time condition turns off, they appear |
| 19:40.34 | fadumpt | time condition to a ring group I mean |
| 19:41.02 | roswell | uhm, so calls in that period of time are simply being forwarded onto a cellphone? |
| 19:42.23 | fadumpt | no, the ring group rings voip phones as well as a cell phone and after 930 it goes to a dial plan that doesn't ring the cell phone. |
| 19:42.53 | fadumpt | I assumed that it possibly couldn't record the conversation if the cell phone picked up the call but I'm not getting recordings from any phone. |
| 19:43.43 | roswell | mhm got that. actually, speaking of freepbx, time ranges are applied to the whole queue |
| 19:45.21 | fadumpt | I see in the extensions_additional.conf that system recordings comes right before time conditions |
| 19:46.06 | roswell | btw also try checking out whether you have enabled call monitoring for the truck which performs outgoing calls onto the cellphone |
| 19:46.15 | roswell | err truck.. trunk* of course |
| 19:46.41 | fadumpt | well separated by an entry called [to-maint] and the second entry of it is exten=> s,1,NoCDR() |
| 19:47.40 | roswell | let's keep it out, including my thoughts on what freepbx does to asterisk dialplan ) |
| 19:47.41 | fadumpt | and then two lines referring to a time condition |
| 19:48.16 | roswell | would you paste somewhere the output of >dialplan show |
| 19:52.51 | Penguin | Oh for crying out sakes... |
| 19:52.56 | Penguin | I've totally lost it now. |
| 19:52.57 | Penguin | ssh: Could not resolve hostname vim: Name or service not known |
| 19:53.27 | Chainsaw | Caffeine IV for the Penguin. |
| 19:54.13 | Penguin | I have a liter of Mtn Dew. |
| 19:54.25 | Chainsaw | Close enough :) |
| 19:54.29 | Penguin | But it's not doing anything for me. |
| 19:54.39 | Cuzner | Mountain Dew still has 0 caffeine in Canada :( |
| 19:54.51 | Chainsaw | Penguin: This is going to take something radical. Like actual sleep. |
| 19:55.09 | Chainsaw | Back later. |
| 19:56.26 | Penguin | Canadian Mountain Dew has no caffeine? I have never heard of this craziness. |
| 20:00.14 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 20:05.16 | Cuzner | Penguin: we have laws regarding citrus drinks containing caffeine, i know, it's silly. |
| 20:05.32 | Penguin | Do you happen to know why? |
| 20:06.30 | Cuzner | oh shit, looks like they changed it in 2012 |
| 20:07.14 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-19-252.dynamic.qsc.de) |
| 20:08.06 | Cuzner | yeah, i think that's about the time I saw Barqs hitting the shelves... We had some weird law about Cola being the only thing that could contain caffeine |
| 20:08.51 | roswell | boo bots |
| 20:09.34 | roswell | fadumpt, you there, mate? |
| 20:09.50 | Cuzner | dr. pepper was/is considered a cola, and was probably the highest caffeinated drink you could get up here... Any caffeinated drinks that weren't colas (ala energy drinks like red bull) were sold as "Natural Health Food Products", which skirted the laws as they didn't call themselves soft drinks or sodas. |
| 20:12.08 | fadumpt | roswell, yeah, I went to copy the text and realized I had no clue how to do it. |
| 20:12.27 | fadumpt | so I tried to just copy the extensions_additional because of that crazy maint entry but that came out garbled |
| 20:12.38 | roswell | sighs ) |
| 20:12.43 | fadumpt | :) |
| 20:13.41 | roswell | actually here's my friendly advice -- while it's not too late, use pure asterisk |
| 20:13.41 | *** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com) |
| 20:14.56 | fadumpt | it's http://pastebin.com/t2CnQ3wx |
| 20:15.07 | fadumpt | okay, that's *just* that odd looking entry |
| 20:15.55 | [TK]D-Fender | exten => s,n,System(/var/lib/asterisk/bin/schedtc.php 60 /var/spool/asterisk/o$ |
| 20:16.01 | [TK]D-Fender | incomplete line |
| 20:16.14 | roswell | when i see exten => A,n+1, i want to kill it with fire ) |
| 20:16.46 | roswell | it's same => n,... for i don't know 7 years |
| 20:16.47 | fadumpt | exten => s,n,System(/var/lib/asterisk/bin/schedtc.php 60 /var/spool/asterisk/outgoing ${CALLERID(number)}) |
| 20:16.50 | Cuzner | same => n,... looks nicer |
| 20:17.27 | roswell | ^ yup listen to that dad |
| 20:17.42 | Cuzner | roswell: yeah, i write all my dialplan like that now, w/ as few GoTo()s as possble, i try to replace them all with GoSub()s & Return()s |
| 20:18.33 | fadumpt | so does the NoCDR and references to time conditions seem suspect? |
| 20:19.03 | Cuzner | roswell: it helps that I have probably the best teachers I can get... |
| 20:20.57 | roswell | fadumpt, what's on the missing line? |
| 20:21.02 | fadumpt | exten => s,n,System(/var/lib/asterisk/bin/schedtc.php 60 /var/spool/asterisk/outgoing ${CALLERID(number)}) |
| 20:22.33 | roswell | what's on timeconditions ? |
| 20:22.50 | roswell | i mean in* ) |
| 20:23.19 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 20:28.45 | fadumpt | okay |
| 20:28.46 | fadumpt | http://pastebin.com/GXfMwYGg |
| 20:31.10 | roswell | fadumpt, also anything which contains "ext-local*" in it |
| 20:33.56 | roswell | hence i hate freepbx, it's unpredictable ) |
| 20:34.15 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:34.53 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 20:35.05 | fadumpt | okay, so NoCDR == No Call Detail Records |
| 20:35.17 | fadumpt | so can I remove or comment out that line and see what happens tomorrow morning? |
| 20:36.02 | roswell | it doesn''t involve voice recoring anyhoo |
| 20:36.20 | roswell | involve == affect* |
| 20:37.19 | fadumpt | okay. |
| 20:40.40 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 20:40.46 | fadumpt | I found an option on the Inbound Routes page called Call Recording and the options are Allow, Record on answer, record immediately, never - I'm changing from allow to record immediately |
| 20:42.09 | fadumpt | I've gotta go but thanks for all of the help |
| 20:45.36 | *** join/#asterisk Jacoby6000 (~jbarber@static-71-252-134-63.dllstx.fios.verizon.net) |
| 20:46.02 | Jacoby6000 | What happens if I run queue "remove member x from y" if queue member X is not logged in to queue Y? |
| 20:50.00 | Penguin | Remember, you were told that you should not edit the extensions file yourself. And you were told to go to #freepbx for help fixing your problem. |
| 20:50.39 | fadumpt | I didn't touch any files and I did *join* freepbx.... |
| 20:50.49 | fadumpt | I changed the setting in freepbx itself |
| 20:50.52 | Penguin | One reason for this is that when you edit the thing by hand, the next time you hit the apply button in FreePBX, your manual changes will be gone. |
| 20:50.59 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 20:51.08 | Penguin | good |
| 20:51.35 | *** join/#asterisk calum_ (~calum_@cpc70817-harg4-2-0-cust754.7-1.cable.virginm.net) |
| 20:58.07 | calum_ | I have a client with an asterisk 1.8 based system (freepbx) and they are experiencing strange behaviour with the system. There have been reports of not being able to call internally and external calls (dahdi) dropping out. I turned on call recording and have found a couple of calls where the audio drops out and it appears to the extension user as if the external call has been disconnected. The thing is that the dahdi cha |
| 20:58.07 | calum_ | nnel is still open as the psnt caller can still be heard on the recording. The logs show the extension hangs up and in turn the dahdi channel is closed. It is almost as if the RTP stream between the handset and the PBX is lost. Where should I start / go from here????? |
| 21:03.52 | Penguin | ~freepbx |
| 21:03.52 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:06.50 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 21:14.36 | *** join/#asterisk lorsungcu_ (~lorsungcu@c-50-137-202-86.hsd1.mn.comcast.net) |
| 21:21.53 | *** join/#asterisk calum_ (~calum_@cpc70817-harg4-2-0-cust754.7-1.cable.virginm.net) |
| 21:26.12 | calum_ | I have a client with an asterisk 1.8 based system (freepbx) and they are experiencing strange behaviour with the system. There have been reports of not being able to call internally and external calls (dahdi) dropping out. I turned on call recording and have found a couple of calls where the audio drops out and it appears to the extension user as if the external call has been disconnected. The thing is that the dahdi cha |
| 21:26.15 | calum_ | nnel is still open as the psnt caller can still be heard on the recording. The logs show the extension hangs up and in turn the dahdi channel is closed. It is almost as if the RTP stream between the handset and the PBX is lost. Where should I start / go from here????? |
| 21:29.32 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pnnnxokbpmzdsomd) |
| 21:29.32 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:31.05 | Cuzner | ~freepbx |
| 21:31.05 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:32.54 | *** join/#asterisk king1337-2 (~king1337@S0106c8fb2641d848.vs.shawcable.net) |
| 21:35.03 | calum_ | Cuzner: I don't think it is a freepbx issue though! I think it is todo with asterisk or the RTP stream!!! |
| 21:36.00 | Cuzner | personally i'd start with a pcap. |
| 21:36.05 | KNERD | Under asterisk 11 and 13 I am getting this error after running "asterisk c-vvvv" --> asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory . This continues to show up even after doing "-disable-asteriskssl " |
| 21:36.15 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.31.158) |
| 21:37.46 | *** join/#asterisk lorsungcu_ (~lorsungcu@209-173-236-30.usfamily.net) |
| 21:42.20 | calum_ | Cuzner: Ok so I can use wireshark to capture the packets but is there a common error to look for? |
| 21:42.34 | KNERD | i got it |
| 21:48.26 | Cuzner | calum_: http://en.wikipedia.org/wiki/List_of_SIP_response_codes#4xx.E2.80.94Client_Failure_Responses |
| 22:06.28 | *** join/#asterisk lorsungcu__ (~lorsungcu@74-95-143-117-Minnesota.hfc.comcastbusiness.net) |
| 22:07.01 | *** join/#asterisk lvlinux (~ruel@unaffiliated/lvlinux) |
| 22:19.44 | *** join/#asterisk lorsungcu_ (~lorsungcu@209-173-236-30.usfamily.net) |
| 22:20.57 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-qwqqjwttydogzwhh) |
| 22:20.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:46.53 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-19-252.dynamic.qsc.de) |
| 23:01.50 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 23:05.33 | *** join/#asterisk areski (~areski@219.6.17.95.dynamic.jazztel.es) |
| 23:06.43 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-bttxajfyewsfisvh) |
| 23:06.52 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 23:06.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 23:06.53 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 23:07.02 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 23:07.04 | *** join/#asterisk leffelad (~leffelad@shiftingshadow.warbear.co.uk) |
| 23:07.34 | *** join/#asterisk War_Bear (~War_Bear@shiftingshadow.warbear.co.uk) |
| 23:07.44 | *** join/#asterisk elbriga (~elbriga@177.220.175.234) |
| 23:08.27 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-hynxwqhhlbzegjfa) |
| 23:08.27 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:08.51 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-ezqcqaldbkjjvqju) |
| 23:12.36 | *** join/#asterisk overyander (~Jeff@12.49.160.131) |
| 23:16.14 | *** join/#asterisk Hexxis (~Hexxis@107.150.31.94) |
| 23:38.23 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-19-252.dynamic.qsc.de) |