| 00:17.44 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 00:18.34 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 00:44.09 | *** join/#asterisk u0m3 (~u0m3@92.80.85.161) |
| 00:47.07 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 01:00.49 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:01.43 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 01:14.55 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 01:17.59 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 01:25.31 | *** join/#asterisk FreezingAlt (~FreezingC@135.0.41.14) |
| 01:28.55 | *** join/#asterisk FreezingAlt (~FreezingC@135.0.41.14) |
| 01:30.57 | *** join/#asterisk FreezingAlt (~FreezingC@135.0.41.14) |
| 01:44.19 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 01:45.04 | zekoZeko | what's the difference between automon and automixmon? It seems to me they both create the same thing, mixing both sides of the conversation into a mono wav file... |
| 01:46.29 | zekoZeko | i'd love it if there was a way to record into a "stereo" wav (caller left, callee right) |
| 01:47.08 | zamba | i believe i've seen something like that somewhere |
| 01:47.20 | zamba | maybe entered the conversation too late, though |
| 01:50.12 | file | the difference is the underlying application used, and thus implementation |
| 01:50.43 | file | automon uses Monitor() which is MUCH older and implemented very differently, automixmon uses MixMonitor() which is newer |
| 02:10.04 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 02:10.34 | zekoZeko | thanks |
| 02:23.00 | zekoZeko | yay, managed to setup on demand call recording and sending the file to e-mail... the journalists will love this, they really hate the hybrid they use now :) |
| 02:25.57 | zekoZeko | that's why i'd like to separate the recording into two channels so if someone barges in the room in the middle of an interview it would only be audible on the part of the recording that we don't even need. |
| 02:26.34 | zamba | cool |
| 02:26.36 | zamba | i like the feature |
| 02:27.39 | zekoZeko | it should be possible. My incron shows this when the recording finishes: |
| 02:27.42 | zekoZeko | CMD (/usr/local/sbin/send-asterisk-recording.sh /var/spool/asterisk/monitor auto-1419819591-819-817-in.wav) |
| 02:27.51 | zekoZeko | CMD (/usr/local/sbin/send-asterisk-recording.sh /var/spool/asterisk/monitor auto-1419819591-819-817-out.wav) |
| 02:27.53 | zekoZeko | CMD (/usr/local/sbin/send-asterisk-recording.sh /var/spool/asterisk/monitor auto-1419819591-819-817.wav) |
| 02:28.11 | zekoZeko | so it's obvious it has the separate recordings, but it mixes them together |
| 02:28.30 | zekoZeko | have to dive into the docs |
| 02:35.35 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 02:39.57 | *** join/#asterisk goseeped (~goseeped@190.207.75.74) |
| 02:43.27 | *** join/#asterisk DivideBy0x0 (~DivideBy0@70-127-249-249.res.bhn.net) |
| 03:03.24 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 03:04.54 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 03:08.04 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:19.35 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
| 03:21.29 | *** join/#asterisk cvbnm (~chatzilla@pool-173-76-79-95.bstnma.fios.verizon.net) |
| 03:21.57 | cvbnm | Is there a place I can go to hire an asterisk programmer? |
| 03:22.41 | *** join/#asterisk rhineheart_m (~rhinehear@unaffiliated/rhineheartm/x-283746) |
| 03:23.25 | rhineheart_m | hello.. in hardware detector is says "Not detected by Asterisk" all calls now are "all circuits are busy" |
| 03:29.30 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 03:31.54 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 03:47.17 | *** join/#asterisk u0m3_ (~u0m3@92.80.103.197) |
| 04:00.06 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 04:26.15 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 04:43.20 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.83.187) |
| 04:45.25 | ruben23 | hi guys anyone can help with this error somehow ---> [Dec 29 15:42:14] WARNING[5392]: chan_sip.c:13482 handle_response_invite: Received response: "Forbidden" from '"DC827986W0000003982W" <sip:diallerxfer@192.168.7.125>;tag=as28c7e285' |
| 04:46.02 | ruben23 | when i dial a local extension on peered asterisk i got this error always |
| 04:51.08 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 04:55.53 | ChannelZ | I'm guessing your peer is misconfigured and isn't matching |
| 05:05.42 | ruben23 | ChannelZ: but both i can see from each other...asterisk 1 ---> diallertrans/diallerxfer 192.168.7.123 N 5060 OK (1 ms) |
| 05:06.00 | ChannelZ | that doesn't mean anything |
| 05:06.09 | ruben23 | asterisk2 ----> diallertrans/diallerxfer 192.168.7.125 Auto (No) No 5060 OK (1 ms) |
| 05:06.33 | ruben23 | does it not implies that both are peered correctly..? |
| 05:07.27 | ChannelZ | No, it means you got them to register to each other |
| 05:07.55 | ChannelZ | But something is different about you sending calls between them. Need to see more console output and/or SIP debug to go much further though |
| 05:13.05 | ruben23 | where do i execute sip debug on asterisk 1 - sending calls or asterisk 2 receiving calls.? |
| 05:14.02 | ChannelZ | either side would be a place to start but the receiving end since it's the one rejecting |
| 05:17.00 | ChannelZ | core set verbose 3 |
| 05:17.02 | ChannelZ | sip set debug on |
| 05:17.07 | ChannelZ | <PROTECTED> |
| 05:17.57 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 05:24.36 | ruben23 | ok doing it now |
| 05:28.52 | ChannelZ | plays the Jeopardy theme |
| 05:38.37 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 05:44.00 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 06:09.59 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 06:33.24 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 06:33.56 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 06:58.17 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 07:00.07 | *** join/#asterisk zblk (~andrey@92.53.115.234) |
| 07:13.04 | ChannelZ | shrugs |
| 07:18.16 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 07:22.46 | [TK]D-Fender | ChannelZ, He also asked a question in #freepbx and didn't respond to questions there either. |
| 07:24.02 | ChannelZ | He does that a lot, comes in all in a panic about something.. then never responds |
| 07:28.05 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 07:32.44 | *** join/#asterisk linuxgeek (~linuxgeek@2a03:b0c0:2:d0::5f:2001) |
| 07:35.37 | [TK]D-Fender | heads to bed... |
| 07:51.49 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 07:56.42 | *** join/#asterisk u0m3__ (~u0m3@92.80.103.197) |
| 08:02.38 | *** join/#asterisk ghoti (~paul@hq.experiencepoint.com) |
| 08:02.57 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 08:04.09 | *** join/#asterisk tm1000 (sid6728@gateway/web/irccloud.com/x-bvlkukxqulwbwctx) |
| 08:04.22 | *** join/#asterisk zamba (marius@flage.org) |
| 08:07.51 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:11.20 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:18.27 | ChannelZ | Color testing.. ignore me |
| 08:18.31 | ChannelZ | barfs |
| 08:23.02 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 08:23.20 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
| 08:23.41 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 08:24.43 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 08:44.59 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 08:47.38 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 08:58.14 | *** join/#asterisk areski (~areski@80.174.128.116.dyn.user.ono.com) |
| 09:11.29 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 09:35.37 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 09:42.47 | *** join/#asterisk zblk (~andrey@92.53.115.234) |
| 09:59.35 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 10:01.16 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 10:07.55 | *** join/#asterisk CeBe (~CeBe@brsg-d9beab0a.pool.mediaWays.net) |
| 10:16.17 | *** join/#asterisk stefan27 (~stefan27@212.247.4.149) |
| 10:24.51 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 10:30.13 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:40.43 | *** join/#asterisk ghoti (~paul@hq.experiencepoint.com) |
| 10:49.15 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 11:03.41 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:11.55 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 11:18.06 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 11:18.56 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 11:37.36 | *** join/#asterisk kamika (5f5c7676@gateway/web/freenode/ip.95.92.118.118) |
| 11:40.57 | kamika | Hi, what is like the "normal" hardware setup for a asterisk machine... with 20 simultaneous calls? |
| 11:41.38 | kamika | I'm looking at some server in xcase.co.uk... |
| 11:42.02 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:42.05 | WIMPy | No such thing as "normal". |
| 11:50.12 | kamika | WIMPy, yes I know... but an I3 with 2GB is a typical configuration? |
| 11:50.48 | WIMPy | Saying "typical" instead of "normal" doesn't change things. |
| 11:51.08 | WIMPy | The number of calls is rather meaningless. It depends on what you want to do. |
| 11:54.35 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 11:56.11 | *** join/#asterisk riess82 (~riessma@91-115-67-14.adsl.highway.telekom.at) |
| 11:56.36 | stefan27 | [2014-12-29 12:56:00] WARNING[5333]: app_queue.c:3041 queue_set_param: Changing to the linear strategy currently requires asterisk to be restarted. |
| 11:56.40 | stefan27 | why is that? |
| 11:57.02 | *** join/#asterisk goseeped (~goseeped@190.207.75.74) |
| 11:57.34 | kamika | typical, well I just would like to know want is the "minimal" configuration for a simple asterisk server.... just don't want to spend a lot of money in something that will be overkill :/ |
| 11:57.39 | WIMPy | You peeled off the shiney veneer. |
| 11:57.44 | WIMPy | Put it back on. |
| 11:58.13 | WIMPy | kamika: Install it on your plastic router. |
| 11:58.59 | goseeped | hi guys , im having problems webrtc VP8 Passthrough support for the video codec , asterisk 13.1 , ubuntu 14.04 , pjsip from asterisk github repo, librtp from cisco git repo , endpoints FF 30 and last chome too ,audio is fine ! debug is fine , asterisk 13 just not accept vp8 offer from web browser . |
| 11:59.06 | goseeped | look my dtls config : media_encryption=dtls , dtls_verify=fingerprint, dtls_cert_file=/etc/asterisk/keys/asterisk.pem , dtls_ca_file=/etc/asterisk/keys/ca.crt , dtls_setup=actpass , use_avpf=yes , ice_support=yes , media_use_received_transport=yes |
| 11:59.21 | goseeped | Has anyone tried video with webrtc and pjsip . In my case voice works well but video does not work even when i allow vp8 codec in pjsip.conf ?????? |
| 12:01.31 | *** join/#asterisk kizoku (~albus@cm-84.215.23.97.getinternet.no) |
| 12:02.23 | stefan27 | I use webrtc with chan_sip not pjsip, but haven't tried video yet (13.1 too) |
| 12:05.34 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 12:06.00 | kizoku | I feel like a twat, but can anyone show me a simple way to enable sip debuging output? My register messages are arriving at the machine, but it doesnt look like they are beeing processed by asterisk. |
| 12:06.19 | kizoku | (testbed setup, so load is of no concern) |
| 12:06.32 | *** join/#asterisk mirela666 (~mirko@212.200.146.247) |
| 12:08.39 | stefan27 | in the command line interface of asterisk, have you done "sip debug" or "sip set debug on" (depending on version) |
| 12:11.09 | kizoku | Thanks! |
| 12:11.19 | stefan27 | also is the correct port bound at your machine? e.g. "netstat | grep asterisk" shows udp 0 0 0.0.0.0:8060 0.0.0.0:* 9118/asterisk for me, because i use port 8060 for sip |
| 12:12.13 | kizoku | Should it show even if Im only using udp? |
| 12:12.39 | kizoku | This was working with the version installed by yum, but not with the one I compiled from source (13.0.1) |
| 12:16.30 | kizoku | Oh my. It was my bad. |
| 12:16.34 | kizoku | iptables >_< |
| 12:16.35 | kizoku | Working now. |
| 12:18.35 | *** join/#asterisk calum_ (~calum_@cpc70817-harg4-2-0-cust754.7-1.cable.virginm.net) |
| 12:20.26 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 12:20.42 | *** join/#asterisk riess82 (~riessma@91-115-67-14.adsl.highway.telekom.at) |
| 12:24.54 | *** join/#asterisk kleszcz (tick@linuxmafia.pl) |
| 12:26.41 | goseeped | stefan27 : i also tried with chan_sip but also video doesn't work ; when you see signalling asterisk is rejecting the VP8 offer ! |
| 12:33.16 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 12:41.31 | *** part/#asterisk mirela666 (~mirko@212.200.146.247) |
| 12:48.04 | stefan27 | does the CLI give you any information when it rejects the offer |
| 12:48.23 | stefan27 | any warning messages in console? |
| 12:50.49 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 12:50.49 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:00.53 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:07.01 | *** join/#asterisk bob_cheesey (~shw@support.memset.com) |
| 13:07.19 | mjordan | stefan27: you're just inserting your queue members into your database? |
| 13:07.44 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 13:08.31 | bob_cheesey | hi guys, sorry if this is the wrong place for this. i've got a hunt group to some snoms and when people are working from home they use the snom gui to forward calls to their mobiles, however this also forwards the hunt group. if someone is on the phone the hunt group always goes straight to their voicemail as that effectively picks up the call |
| 13:08.36 | bob_cheesey | is there any way around this? |
| 13:10.37 | mjordan | stefan27: anyway... to answer your question from #asterisk-dev about how dynamic members are loaded - they are loaded in the order that a SELECT statement on the queue_members table returns them, whenever something needs to interact with that queue. That could be the next caller, etc. |
| 13:10.50 | mjordan | stefan27: how your backend returns those rows is dependent on your realtime backend |
| 13:10.52 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 13:13.52 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-xkkbvmjypfkwsrfz) |
| 13:15.43 | stefan27 | thanks mjordan |
| 13:16.43 | stefan27 | yes, i'm just inserting them into the database |
| 13:18.14 | stefan27 | but I'll take a closer look since I didn't write all the scripts myself. Also, is there really no way to change a queue from rrmemory to linear without restarting asterisk? |
| 13:18.50 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 13:21.21 | mjordan | that's what the warning message says. |
| 13:38.20 | stefan27 | right -- I did find that warning, so not having to restart is more of a feature request. |
| 13:44.38 | stefan27 | the code surrounding that warning in app_queue.c (line 3040) in app_queue.c contains no explanations why full restart is needed. maybe there's a work-around by reloading a bunch of modules |
| 14:05.49 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 14:18.00 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:18.24 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 14:18.50 | *** join/#asterisk Vince-0 (~vincent@41.162.57.226) |
| 14:21.43 | *** join/#asterisk Milenco (~Milenco@home.milenco.net) |
| 14:29.00 | *** join/#asterisk jeffspeff (~Jeff@12.49.160.131) |
| 14:31.17 | *** join/#asterisk areski (~areski@18.212.133.37.dynamic.jazztel.es) |
| 14:40.56 | davlefou | hi, |
| 14:41.20 | davlefou | did you have use spychan with aterisk 11? |
| 14:42.03 | *** join/#asterisk frederikroes1 (~fro@195-130-157-248.ifiber.telenet-ops.be) |
| 14:44.21 | frederikroes1 | hello |
| 14:44.44 | frederikroes1 | I have a question |
| 14:44.50 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 14:44.52 | frederikroes1 | I'm running Asterisk 11.14.1 via AsteriskNow 3 |
| 14:45.04 | frederikroes1 | I already updated FreePBX to version 12 |
| 14:45.20 | frederikroes1 | but how can I upgrade Asterisk to the latest version 13 (LTS) ? |
| 14:47.58 | frederikroes1 | anyone who can clarify what the best way to upgrade is? |
| 14:48.56 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-yqhqrholzatennyk) |
| 14:48.56 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:52.00 | *** part/#asterisk bob_cheesey (~shw@support.memset.com) |
| 14:54.46 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 14:55.21 | frederikroes1 | nobody? |
| 14:55.52 | tomodachi | frederikroes1: people here i believe will recomend you to talk to your distribution |
| 14:56.22 | tomodachi | if they dont have the "latest" asterisk , you can download and insatll it yourself from sources. Its of course not the "way" that most distributions work |
| 15:05.03 | frederikroes1 | ok... |
| 15:05.23 | frederikroes1 | so if it's not found under yum.. I need to install it from source ? |
| 15:09.07 | *** join/#asterisk generalhan_ (~tester@about/windows/staff/generalhan) |
| 15:17.42 | *** join/#asterisk newtonr (~newtonr@173-21-90-203.client.mchsi.com) |
| 15:17.43 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:20.59 | *** join/#asterisk pmd` (~pmd`@2001:8b0:ca70:32b2:3cde:f4ba:ea45:27b) |
| 15:23.23 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 15:26.57 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-uvgefudzntxqrqpa) |
| 15:36.37 | stefan27 | frederikroes1 whats your OS? http://packages.asterisk.org/centos/6/asterisk-13/ helped me |
| 15:47.37 | frederikroes1 | stefan27 Centos 6.4 |
| 15:51.39 | stefan27 | try the above rpms |
| 15:52.12 | stefan27 | (i havent tested them) |
| 15:53.52 | *** join/#asterisk riess82 (~riessma@88-117-62-119.adsl.highway.telekom.at) |
| 15:58.37 | frederikroes1 | ok :) thank you |
| 15:58.59 | frederikroes1 | but I'm not sure if it will work with AsteriskNow :) |
| 16:00.20 | *** join/#asterisk Johnny5_ (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 16:10.48 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 16:12.30 | stefan27 | regarding my queue-question from this morning: I would like to be able to administrate the ring-order using linear queue-strategy where the queue uses realtime from mysql backend. That is, I want the in-memory-list (as shown by "queue show") to be synced with the queuemembers mysql-db. Currently if the output of asterisk's query [pid 2693] read(112, "\3SELECT * FROM queuemembers WHERE |
| 16:12.30 | stefan27 | interface LIKE '%' AND queue_name = 'queue-01' ORDER BY interface", 99) = 99 ... returns the same result-set as last-time only with a different order, asterisk does not update its in-memory-list at all. That's by design I guess but I want to work around it somehow |
| 16:13.29 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 16:13.42 | stefan27 | to change ring-order without patching asterisk it seems i have to perform a sequence of agent-logout+agent-logins |
| 16:14.32 | stefan27 | Maybe there's another way to accomplish what I want |
| 16:19.20 | stefan27 | has anyone else requested this kind of feature? ringing members in the order in which they were added seems arbitrary, but maybe not a lot of people use realtime+linear and want to control order |
| 16:29.47 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 16:36.52 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 16:39.53 | *** join/#asterisk KNERD (~KNERD@64.31.22.134) |
| 16:48.50 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:50.00 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 17:02.13 | Cuzner | Anyone in here use sippy_cup for testing? I can get scenarios working fine if I craft a peer without authentication details, but I can't seem to figure out how to create a scenario that can utilize sip auth |
| 17:02.37 | Cuzner | i tried specifying the login creds with - register <user> <pass>, but I still get 401s |
| 17:13.25 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:13.26 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 17:14.37 | Cuzner | or if someone can point me to the appropriate channel to be asking such a question, that'd be great too :) |
| 17:19.44 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 17:20.03 | azerus | Any nice guides for setting up a call-center with Asterisk? |
| 17:27.46 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 17:34.07 | Penguin | ~book |
| 17:34.07 | 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 |
| 17:34.29 | azerus | <3 |
| 17:55.08 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |