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| 01:18.31 | juanmapalad | !ping |
| 01:18.49 | juanmapalad | hi, does all gsm gateway compatible with asterisk? thanks |
| 01:33.09 | juanmapalad | hi, does all gsm gateway compatible with asterisk? thanks |
| 01:49.24 | [TK]D-Fender | that is not a useful description |
| 01:50.32 | [TK]D-Fender | * speaks several VoIP protocols, and very particualr kinds of hardware interfaces directly from PCI() bus and USB |
| 01:50.34 | MaliutaLap | [TK]D-Fender: not even the second time around? |
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| 02:33.59 | LostBoy | MaliutaLap: lol |
| 02:37.11 | MaliutaLap | LostBoy: "Despite all my rage I am still just a rat in a cage." - these days I wait for [TK]D-Fender to set people straight and then use sarcasm to back him up, surprisingly deaths in the * community have decreased since I started this method of coping ;) |
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| 05:15.06 | juanmapalad | hi, does all gsm gateway compatible with asterisk? thanks |
| 05:17.39 | [TK]D-Fender | We just went through this |
| 05:17.40 | [TK]D-Fender | NO |
| 05:17.42 | [TK]D-Fender | Not ALL |
| 05:17.49 | [TK]D-Fender | Because not all talk the same language |
| 05:18.12 | [TK]D-Fender | A gateway translates from ONE kind of thing to another. |
| 05:18.31 | [TK]D-Fender | So get SPECIFIC about models you're looking at. |
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| 05:46.41 | MaliutaLap | [TK]D-Fender: want a bigger cluebat? ;) |
| 05:47.08 | [TK]D-Fender | does all gas work in cars? |
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| 05:49.52 | MaliutaLap | [TK]D-Fender: "does all petroleum distillates compatible with volkswagon?" is a better fit |
| 05:51.09 | MaliutaLap | wonder if he tries to run his car on vaseline |
| 05:51.44 | [TK]D-Fender | Pure carbon-copy questions and no sign of anything registering with the other party.... |
| 05:51.54 | [TK]D-Fender | I'm off for more productive matters. |
| 05:51.59 | MaliutaLap | beer? |
| 05:51.59 | [TK]D-Fender | like sleep |
| 05:52.09 | [TK]D-Fender | heads to bed |
| 05:52.10 | MaliutaLap | and beer ;) |
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| 10:26.39 | bear__ | hi there |
| 10:26.52 | bear__ | вÑем пÑÐ¸Ð²ÐµÑ |
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| 10:32.54 | bear__ | кÑо-нибÑÐ´Ñ Ð¼Ð¾Ð¶ÐµÑ Ð¿Ð¾Ð´ÑказаÑÑ Ð¿Ð¾ ami agi? |
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| 10:35.41 | Jesterboxboy | i have the problem that asterisk detects inband dtmf even when dtmfmode=2833, which results in receiving double dtmf digits. is there a way to switch that off in asterisk? |
| 10:35.46 | Jesterboxboy | version 11.8 |
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| 11:56.23 | cbeyerlein | heya, is there any prefabricated stuff for doing load-balancing via PJSIP endpoint? like aggregating 2 or more remote SIP servers in one PJSIP endpoint, and just dialing the endpoint. |
| 11:57.03 | cbeyerlein | I found how to call all contacts of one endpoint _in parallel_, but thats not what I want :) |
| 11:57.16 | file | there's no load balancing built in at that level, so you'd have to do it at a higher level |
| 11:58.24 | file | underneath it will do SRV on a hostname in 13, and NAPTR+SRV in master |
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| 11:59.44 | cbeyerlein | k thx, I will write some dialplan that checks status codes and re-attempts via other endpoint then. |
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| 12:29.41 | proute | Hi everybody. |
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| 12:31.36 | proute | I use Asterisk 1.8.32 in call center. I need to record all communications. I use mixmonitor in QUEUE. But, when I have lot of call, communications are degraded. Is there other way to record call in QUEUE without mixmonitor? Or is there an option to avoid that mixmonitor degrade the voice quality? |
| 12:31.41 | proute | thanks for your help |
| 12:32.07 | WIMPy | Get faster disks. |
| 12:32.34 | proute | I use a RAMDISK for record all call.... |
| 12:33.43 | WIMPy | You never copy them to some mass storage device? |
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| 12:35.38 | proute | yes at the end of the day |
| 12:36.03 | [TK]D-Fender | mixmonitor should not degrade. Your first step should be using Monitor instead. Id still sooner bet that the recordings are simply done pre-jitterbuffer/PLC and that's just what you'll get |
| 12:37.00 | proute | How can I use "Monitor" in QUEUE? Because I think that I use only mixmonitor? |
| 12:37.45 | WIMPy | You need to find out where you run out of resources. |
| 12:37.50 | [TK]D-Fender | "think"? What doees "looking" tell you? |
| 12:38.24 | proute | In queue.conf I have this options: monitor-type = MixMonitor |
| 12:38.52 | WIMPy | What about the FS? Do you distribute files across directories? |
| 12:39.04 | [TK]D-Fender | [08:35][TK]D-Fendermixmonitor should not degrade. Your first step should be using Monitor instead. |
| 12:39.46 | [TK]D-Fender | WIMPy: he's going to RAM. storage isn't the issue, it's the source or perhaps CPU |
| 12:40.13 | proute | WIMPy: I use different directory (One by day) to record files |
| 12:40.29 | proute | I use 4 CPU |
| 12:40.40 | WIMPy | That might be your issues. |
| 12:40.54 | [TK]D-Fender | [08:35][TK]D-Fendermixmonitor should not degrade. Your first step should be using Monitor instead. Id still sooner bet that the recordings are simply done pre-jitterbuffer/PLC and that's just what you'll get <----- |
| 12:44.06 | proute | For information, my CPU are 0,2% of use |
| 12:45.33 | WIMPy | Well, then maybe you have a network issues. |
| 12:45.38 | WIMPy | -s |
| 12:46.03 | proute | I don't think, because when I disable mixmonitor, calls are good |
| 12:46.09 | [TK]D-Fender | I've given you just about the only thing you can do. Go try. If that doesn't have a noticeable impact (and it shouldn't) then that leaves the recording being pre JB/PLC which is what it is. Also you need to be getting off that version fast. |
| 12:46.17 | proute | when I active mixmonitor, calls are degraded |
| 12:47.12 | WIMPy | So you obviousely ARE running out of resources somewhere, even if you don't see it. |
| 12:52.24 | proute | [TK]D-Fender: I will try monitor instead of mixmonitor |
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| 13:21.34 | file | [TK]D-Fender, that individual never returned... how interesting |
| 13:22.05 | [TK]D-Fender | does all gsm gateway works with asterisks?!?! |
| 13:22.39 | file | no, the PJSIP doesn't work one |
| 13:24.58 | [TK]D-Fender | Thee one where you basically handeed what was like the answer... and no so much as any confirmation like "Yup, I was dumb and or lazy", or "Nope, didn't work" #ragequit |
| 13:26.22 | file | aye |
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| 13:32.10 | [TK]D-Fender | If you're complaining and not immediately following suggestions; not giving feedback on attempts; not actually showing what you're doing, etc then stop being a drain on society. People... sheesh. |
| 13:32.15 | A1F4 | Is it possible to create private mobile communications network using asterisk |
| 13:32.36 | [TK]D-Fender | A1F4: Asterisk won't run your antenna's. |
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| 13:32.45 | [TK]D-Fender | -' |
| 13:32.58 | [TK]D-Fender | "mobile" can be kinda vague as well |
| 13:33.23 | [TK]D-Fender | A1F4: You should probably refine taht question a bit |
| 13:33.49 | A1F4 | GSM/CDMA network? |
| 13:33.59 | A1F4 | Cellphone n/w |
| 13:34.00 | [TK]D-Fender | As to what techs are involved in what degree... people give an idea where you are looking to do this.... |
| 13:34.35 | [TK]D-Fender | Aterisk doesn't speak either of those protocols so all sorts of other gear is involved. At what point * is of any use is the question... |
| 13:35.49 | A1F4 | Is their anything cool to do with asterisk as home back etc. |
| 13:37.07 | [TK]D-Fender | Too vague. |
| 13:37.18 | [TK]D-Fender | What's your idea of "cool"? |
| 13:37.25 | [TK]D-Fender | what is "home back"? |
| 13:38.41 | A1F4 | Is their any way to use asterisk in home to setup something other than corporate pbx and phone system. |
| 13:39.10 | [TK]D-Fender | It's whatever you make out of it |
| 13:39.15 | [TK]D-Fender | it could be a PBX. |
| 13:39.23 | [TK]D-Fender | It could be a dumb answering machine. |
| 13:39.28 | file | I'm working on a game of hide and seek in Asterisk with a coworker, so yeah, it is what you make it |
| 13:39.31 | [TK]D-Fender | For me it's a jukebox and coffee-maker |
| 13:40.09 | [TK]D-Fender | Read the book. Learn what Asterisk is made up of |
| 13:40.18 | [TK]D-Fender | And then see what you want to do. |
| 13:41.04 | glNito | I watched a Defcon piece last night on using Asterisk in an IMSI catcher |
| 13:41.09 | A1F4 | Cool .thanks bro... |
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| 13:43.27 | madduck | Any clue why the Gigaset S685IP base station would reply 500 Int.Server Error to MWI NOTIFY messages received from asterisk? Other than Gigaset being crap? ;) |
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| 14:16.59 | mub | SO my boss found out that I never actually compiled that driver for that PCIe timer thingy because he did the dahdi_test saying "SEE? IT DROPPED BELOW 99.9% ONCE! AAAAAAAAA SUCH A BIG DEAL!" |
| 14:17.18 | mub | So I found out the manufacturer of the card and followed their instructions here: http://amfeltec.com/products/downloads/systemtimer/README.dat |
| 14:18.17 | mub | Except I didn't actually install the dahdi-linux-complete package, I only grabbed the source, then compiled the voicesync package pointing it to that source |
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| 14:19.19 | mub | dahdi_test now always says 99.999% or 100.00%, but will this method fuck up my system in the future? |
| 14:21.00 | mub | The version of dahdi preinstalled might be different than the sources I linked to while compiling the voicesync thingy from amfeltec |
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| 14:56.46 | mrmccrac | im trying to make calls with asterisk and to have it use RTP to specify the rport to use, but in the Contact portion of the INVITE it keeps using port 5060 |
| 14:57.05 | mrmccrac | tried specifying rtpstart and rtpend in rtp.conf but didn't seem to help |
| 14:57.36 | [TK]D-Fender | contact port != rtp port |
| 14:57.47 | [TK]D-Fender | contact = signalling |
| 14:58.36 | mrmccrac | hmm a working X-Lite instance is sending Contact URI parameter: rintsance=<hexstring> |
| 14:58.56 | mrmccrac | asterisk is sending Contact URI Host Port: 5060 |
| 14:59.26 | mrmccrac | you know what signaling config this is? |
| 15:01.37 | [TK]D-Fender | SIP is the signaling |
| 15:01.41 | [TK]D-Fender | you're missing the point |
| 15:03.02 | mrmccrac | i guess i dont understand what rport / rinstance is |
| 15:03.15 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 15:03.50 | mrmccrac | and what xlite config tells it to use that and what asterisk config isn't telling it to use that |
| 15:04.16 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 15:06.09 | mrmccrac | has something to do with nat / RFC 3581 :) |
| 15:09.43 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 15:11.03 | mrmccrac | nat=yes still has contact with port 5060 |
| 15:11.29 | [TK]D-Fender | Which still has nothing to do with RTP. |
| 15:11.34 | [TK]D-Fender | never does |
| 15:11.38 | mrmccrac | yeah ive gathered that :) |
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| 15:14.26 | mrmccrac | has to do with nat support |
| 15:16.02 | [TK]D-Fender | mrmccracnat=yes still has contact with port 5060 <- this tells * to IGNORE what the client is sending for contact I |
| 15:16.18 | [TK]D-Fender | Which sounds like it's against your goals |
| 15:16.29 | mrmccrac | nat=yes|no|never|route |
| 15:16.58 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-sjpaiupmtszzqntd) |
| 15:18.18 | mrmccrac | reading http://www.voip-info.org/wiki/view/Asterisk+sip+nat |
| 15:21.33 | [TK]D-Fender | Last modification: Tue 15 of May, 2012 (14:18 UTC) by brownian |
| 15:21.38 | [TK]D-Fender | 3 years old for the latest change |
| 15:21.45 | [TK]D-Fender | most of that wiki is decrepit at best |
| 15:29.37 | mrmccrac | yeah i cant find the config option for this behavior |
| 15:30.22 | mrmccrac | http://doxygen.asterisk.org/trunk/Config_sip.html |
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| 15:37.18 | mrmccrac | ok its sending contact port 5060 for the invite |
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| 15:48.16 | mrmccrac | yeah i dont understand :( |
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| 17:16.48 | file | meep |
| 17:17.10 | mrmccrac | the To and From look correct now its just the Contact its sending is sending the Port and i dont want it to |
| 17:17.32 | file | raises eyebrow |
| 17:20.49 | mrmccrac | gah |
| 17:23.24 | *** join/#asterisk Ulrar (~Ulrar@luwin.ulrar.net) |
| 17:23.51 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
| 17:23.52 | Ulrar | Hi, just configured an Asterisk and I noticed that all the outgoing call end up with an unknown callerid |
| 17:24.27 | Ulrar | What am I supposed to define before using the trunk to get the correct callerid on the final phone ? |
| 17:26.39 | [TK]D-Fender | Before you dial .... set the callerid |
| 17:27.44 | *** join/#asterisk airjump (~Thunderbi@p20030070CE36303901153F1DAFC9B60F.dip0.t-ipconnect.de) |
| 17:28.12 | Ulrar | Just CallerID ? Not CallerID(num) or something ? |
| 17:28.28 | [TK]D-Fender | the FUNCTION |
| 17:28.35 | [TK]D-Fender | same one you should be using since 1.2 |
| 17:28.42 | [TK]D-Fender | "core show function CALLERID" |
| 17:29.11 | Ulrar | Ha well, that seem simple |
| 17:29.23 | Ulrar | I haven't used asterisk in a long long time :) |
| 17:29.25 | Ulrar | Thanks |
| 17:30.06 | [TK]D-Fender | 1.2 was about 8-9 years ago |
| 17:32.29 | file | 2005-11-21 |
| 17:32.41 | file | almost 10 years |
| 17:35.39 | [TK]D-Fender | wow... |
| 17:35.42 | [TK]D-Fender | where does the time go... |
| 17:36.55 | file | North Korea. |
| 17:41.24 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 17:42.38 | mrmccrac | well this finally works and i think the Contact Port wasn't what the problem was anyway |
| 17:43.03 | [TK]D-Fender | could have sworn he said that hours ago... |
| 17:43.36 | mrmccrac | well thats the first thing i saw that was different between what X-Lite does versus Asterisk |
| 17:43.44 | mrmccrac | i was comparing the two in wireshark |
| 17:45.51 | cusco | hi folks |
| 17:46.16 | cusco | I'm looking for hardware with solution made based on asterisk or something |
| 17:46.45 | cusco | something cheap... any ideas? |
| 17:46.45 | [TK]D-Fender | "something" |
| 17:46.50 | cusco | small |
| 17:46.59 | cusco | for a small company with 6 voip phones |
| 17:46.59 | [TK]D-Fender | try fixing that into a proper description including your actual needs |
| 17:47.23 | [TK]D-Fender | cuscoI'm looking for hardware with solution made based on asterisk or something <- </yoda> |
| 17:47.31 | WIMPy | And what kind of hardware are you looking for? |
| 17:47.41 | cusco | something small, set up one, and forgey |
| 17:47.43 | cusco | something small, set up one, and forget |
| 17:47.50 | cusco | ah, not sure if it needs to have a pstn entry |
| 17:48.07 | [TK]D-Fender | cusco: Any dumb SFF refurb machine + FreePBX ISO. |
| 17:48.11 | [TK]D-Fender | DONE |
| 17:48.31 | WIMPy | SFF? |
| 17:48.35 | [TK]D-Fender | Small Form Factor |
| 17:48.38 | cusco | yea.. |
| 17:48.48 | cusco | yea ok thanks :) |
| 17:48.50 | WIMPy | ah |
| 17:49.04 | [TK]D-Fender | like any of those compact Dell/HP/Lenovo desktops they use all the time |
| 17:49.11 | [TK]D-Fender | Same cheap junk as always. |
| 17:49.21 | [TK]D-Fender | get them for < $100 for a spec that's more than enough |
| 17:49.25 | *** part/#asterisk madduck (~madduck@debian/developer/madduck) |
| 17:49.37 | [TK]D-Fender | Because so far "we don't care" |
| 17:50.11 | cusco | yep |
| 17:50.17 | cusco | ok thx |
| 17:58.31 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:08.26 | cusco | looking for a card to connect to a analogic phone line, I'm looking for a fxo or bri card, right? |
| 18:08.45 | WIMPy | No. FXO only. |
| 18:09.53 | cusco | ow ok |
| 18:10.47 | mrmccrac | hum its trying find an extension in the outbound context when an incoming call comes in? |
| 18:11.04 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 18:11.19 | WIMPy | Whatever you configured. |
| 18:11.35 | cusco | define the context in the peer config in sip.conf |
| 18:11.42 | cusco | for incoming |
| 18:12.18 | mrmccrac | hmm yeah i have an inbound and outbound context in sip.conf |
| 18:13.20 | *** join/#asterisk bkruse (~Adium@236.sub-70-210-22.myvzw.com) |
| 18:13.40 | *** join/#asterisk airjump (~Thunderbi@p20030070CE36303901153F1DAFC9B60F.dip0.t-ipconnect.de) |
| 18:17.06 | [TK]D-Fender | There is no such thing as "outbound context" |
| 18:17.23 | [TK]D-Fender | sip.conf sends calls TO the dialplan. |
| 18:17.26 | [TK]D-Fender | Only |
| 18:20.15 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 18:21.15 | mrmccrac | i have two sections in sip.conf, one that defines context=inbound and one with context=outbound |
| 18:21.41 | *** join/#asterisk airjump (~Thunderbi@p20030070CE36303901153F1DAFC9B60F.dip0.t-ipconnect.de) |
| 18:22.17 | mrmccrac | when call comes in i see in debugging that its using the section which has context=outbound |
| 18:22.18 | [TK]D-Fender | ALL contexts are FROM those entries. |
| 18:22.55 | [TK]D-Fender | When a call comes in it is matched against your entries and whichever one that is.. that's the one whose context the call is sent into |
| 18:37.12 | mrmccrac | yeah im baffled as to why its not matching my inbound entry i guess |
| 18:37.29 | *** join/#asterisk twanny796 (2e0b54ae@gateway/web/freenode/ip.46.11.84.174) |
| 18:39.40 | [TK]D-Fender | What "it" was we talking about here? |
| 18:40.12 | [TK]D-Fender | Nouns : Proper > Personal pronouns |
| 18:42.43 | mrmccrac | [2015-10-05 18:42:18] NOTICE[21017]: chan_sip.c:23613 handle_request_invite: Call from 'username' (x.x.x.x:8060) to extension '5555555555' rejected because extension not found in context 'outbound'. |
| 18:43.32 | mrmccrac | before that it says it found peer 'outbound' |
| 18:43.38 | mrmccrac | it = "asterisk" ? heh |
| 18:44.59 | [TK]D-Fender | What DEVICE or SERVICE is sendig you THAT CALL? |
| 18:45.06 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 18:45.18 | [TK]D-Fender | <PROTECTED> |
| 18:45.32 | [TK]D-Fender | That is the dialplan context the section it did match sent it to |
| 18:45.36 | mrmccrac | im dialing from my desk phone to a phone number that gets routed to my asterisk instance |
| 18:45.51 | [TK]D-Fender | What is a "desk phone"? |
| 18:46.03 | [TK]D-Fender | I asked exactly what device or service is sending the SIP request to Asterisk |
| 18:46.06 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 18:48.31 | mrmccrac | User-Agent: ININ-TsServer/15.2.15.22 |
| 18:48.52 | mrmccrac | which is part of Customer Interaction Center |
| 18:49.01 | [TK]D-Fender | Got a normal basic name/description instead of a UA name? |
| 18:49.10 | [TK]D-Fender | LIke... is this from an ITSP you signed up with? |
| 18:50.28 | mrmccrac | http://www.inin.com/solutions/Pages/Contact-Center-Software.aspx |
| 18:50.32 | mrmccrac | its that basically, i dont run it |
| 18:50.39 | mrmccrac | another group on our campus does |
| 18:50.57 | WIMPy | 3 |
| 18:51.02 | WIMPy | oops |
| 18:51.02 | [TK]D-Fender | Ok. |
| 18:51.29 | [TK]D-Fender | "sip set debug on" <- |
| 18:51.37 | mrmccrac | yeah ive got that on |
| 18:51.44 | [TK]D-Fender | look at the call again and it will confirm which entry in sip.conf it matched |
| 18:52.02 | [TK]D-Fender | "found XXXXX" |
| 18:52.12 | [TK]D-Fender | mrmccrac[2015-10-05 18:42:18] NOTICE[21017]: chan_sip.c:23613 handle_request_invite: Call from 'username' (x.x.x.x:8060) to extension '5555555555' rejected because extension not found in context 'outbound'. <- NOT this |
| 18:52.20 | mrmccrac | Found peer 'cic-outbound' for '5555555555' from 1.2.3.4:8060 |
| 18:52.24 | [TK]D-Fender | There you go |
| 18:52.26 | mrmccrac | replaced real phone number and IP with fake above |
| 18:52.29 | mrmccrac | :) |
| 18:52.33 | [TK]D-Fender | that section successfully matched the inbound call |
| 18:52.34 | *** part/#asterisk Ulrar (~Ulrar@luwin.ulrar.net) |
| 18:52.45 | [TK]D-Fender | You almost never need a separate inbound VS outbound entry |
| 18:52.50 | mrmccrac | right so i also have a cic-inbound section, hmm ok |
| 18:52.57 | [TK]D-Fender | Typically a single peer entry is capable of doing BOTH duties |
| 18:53.20 | mrmccrac | i borrowed this syntax from another instance i setup to vitelity (http://www.vitelity.com) |
| 18:53.30 | mrmccrac | ill try not having a separate inbound/outbound |
| 18:54.13 | [TK]D-Fender | Also even if you had 2 sections... the context mentioned in those would be for calls FROM taht service |
| 18:54.20 | [TK]D-Fender | so it'd make no sense for them to be different |
| 18:55.29 | mrmccrac | i guess vitelity used two different hostnames |
| 18:55.33 | mrmccrac | for inbound vs. outbound |
| 18:56.47 | [TK]D-Fender | that context means nothing there |
| 18:57.01 | [TK]D-Fender | that is where calls get sent on INBOUND |
| 18:57.11 | [TK]D-Fender | THere is no such thing as "context" for outbound |
| 18:59.14 | mrmccrac | host=inbound35.vitelity.net instead of host=outbound.vitelity.net |
| 18:59.49 | mrmccrac | for registration it used the inbound35 host |
| 19:00.07 | mrmccrac | guess i shouldn't have used their example for this other use case :) |
| 19:00.12 | *** join/#asterisk bkruse (~Adium@236.sub-70-210-22.myvzw.com) |
| 19:00.25 | [TK]D-Fender | well the context they'd point to shouldn't be different. |
| 19:00.32 | [TK]D-Fender | they all handles FROM the provider |
| 19:00.36 | [TK]D-Fender | they have nothing to do with outbound |
| 19:00.44 | [TK]D-Fender | taht is FROM that source |
| 19:04.35 | *** join/#asterisk ruied (~ruied@bl12-55-151.dsl.telepac.pt) |
| 19:06.31 | mub | I don't use registration strings. Static assign an external IP and just tell your provider! |
| 19:06.41 | mub | And then give me that IP >:P |
| 19:15.29 | mrmccrac | all i know is it was a hell of a lot easier getting asterisk working w/ vitelity compared to my internal SIP provider |
| 19:15.53 | mrmccrac | "put these exact lines in your asterisk config" versus "you figure it out" |
| 19:15.56 | mrmccrac | :) |
| 19:16.22 | [TK]D-Fender | If you know your * basics... tnone of this is hard |
| 19:18.43 | glNito | I'm going to AstriCon so I can go through Asterisk From Scratch |
| 19:19.08 | [TK]D-Fender | You could also .... "Just Do it". |
| 19:19.18 | [TK]D-Fender | There is this wonderful book..... |
| 19:19.22 | mrmccrac | i prefer the whine-on-irc approach |
| 19:28.17 | mub | Maaan, I wish I could go to AstriCon... I'd get so fucked up and belligerent |
| 19:29.24 | mub | Look at that venue! Look at that hotel! I would cosplay as liquor-man |
| 19:31.27 | [TK]D-Fender | It isn't cosplay ... if that's your actual reality... |
| 19:32.01 | glNito | Don't let your dreams be dreams. |
| 19:32.41 | mrmccrac | i wish wearing a cape wasn't frowned upon |
| 19:33.18 | mrmccrac | i think im now finally dialing and receiving calls automatically |
| 19:34.19 | [TK]D-Fender | If my dreams weren't dreams I'd be running out of places for the bodies REALLY fast.... |
| 19:40.08 | *** join/#asterisk Trioxin (~Trioxin@24-145-38-195-dhcp.aik.sc.atlanticbb.net) |
| 19:40.18 | mub | [TK]D-Fender: I stole that. That's mine now. My coworker finds me funny |
| 19:43.23 | Trioxin | question, so many years ago I used to use Trixbox and became quite good at configuring it and freepbx and all that. Now, in 2015, there are a lot more mature and new options that I can see have grown past their old annoyances. I have a company that I'd like to build a new PBX server for.. IVR, extensions, call monitoring, call groups, maybe even voice recognition and text to speech. What would be a choice software to build |
| 19:43.23 | Trioxin | that on these days? I'm not expecting the ease of Trixbox but I don't want to have to spend weeks troubleshooting either. |
| 19:44.53 | Trioxin | and yeah I know trixbox is deprecated and with all these solutions out there now I wouldn't want to use it anyway. |
| 19:46.41 | [TK]D-Fender | Trioxin: You're already here... |
| 19:46.54 | [TK]D-Fender | as for "weeks troubleshooting either", that's up to you |
| 19:48.04 | [TK]D-Fender | Some people learn nothing, have no clue, can't follow instructions, and aren't patient enough to wait for the answers. |
| 19:48.33 | [TK]D-Fender | Voice recognition is something that will always take more personal effort. |
| 19:49.17 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 19:56.25 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 19:57.14 | Trioxin | well I'm looking at a list here. I'm going to put it on a VPS or dedi. I see: Elastix, PBX in a Flash, FreePBX Distro, AsteriskNow, FusionPBX, Blue.box, SipXecs, and Kamailio. So these are PBX solutions and I'm assuming that many of them are using asterisk under the hood. Any thoughts on these? |
| 19:57.48 | WIMPy | And there are a lot more. |
| 19:58.46 | Trioxin | I want the most feature rich with relative ease of use/setup. I'm very comfortable with admining Linux. |
| 19:59.22 | WIMPy | That re contradictory requests. |
| 19:59.31 | WIMPy | are |
| 19:59.36 | glNito | We use Asterisk, FreePBX and Kamailio |
| 19:59.39 | Trioxin | that's why I said "relative" |
| 20:00.34 | [TK]D-Fender | FusionPBX, Blue.box, SipXecs, and Kamailio. <- none of these are Asterisk |
| 20:01.10 | [TK]D-Fender | Elastix is an abomination. PIAF .... meh |
| 20:01.26 | Trioxin | I'm also in need of an auto dialer so I see there's this thing I've never heard of before called GoAutodial. In the past I've always used subscription autodialer companies. Was also wondering if anyone has had experience with GoAutodial |
| 20:01.30 | [TK]D-Fender | AsteriskNOW is basically the same as the FreePBX distro, but necessarily a little behind in its way |
| 20:01.48 | glNito | Elastix dropped the ball a while ago |
| 20:01.57 | [TK]D-Fender | Only time I've heard GoAutodial around here is people with no clue trying to get their systems working |
| 20:02.45 | Trioxin | hmm |
| 20:03.01 | [TK]D-Fender | We don't tend to get or keep too many GUI people here. |
| 20:03.10 | [TK]D-Fender | Mostly because this is not a GUI support channel.... |
| 20:03.10 | Trioxin | Oh right so I see about half of what I mentioned use Freeswitch |
| 20:03.27 | [TK]D-Fender | 1/4 actually |
| 20:03.34 | [TK]D-Fender | the other 2 are completely separate |
| 20:03.59 | Trioxin | oh. right fusion and blue.box are freeswitch |
| 20:04.37 | Trioxin | what about PBX in a Flash? |
| 20:05.04 | [TK]D-Fender | MEH |
| 20:05.13 | [TK]D-Fender | Avoid. More bundled crap |
| 20:05.36 | [TK]D-Fender | if you're going to take any FreePBX base then take the official FreePBX ISO unless you're going to DIY |
| 20:07.21 | *** join/#asterisk Demon_VoIP (~demon@ip253.net222.n37.ru) |
| 20:07.30 | Trioxin | well I could just go for a managed setup so then I would be left customizing. I just don't want to roll something and then find out it doesn't support a feature I want like the ones I mentioned. |
| 20:08.18 | Demon_VoIP | Hello. Do I understand correctly that the current implementation res_pjsip does not support ZRTP? |
| 20:08.22 | Trioxin | GoAutodial features look nice for b2b operations. I've run 2 call centers before and it would be nice to have an easy setup for that sort of thing |
| 20:08.37 | [TK]D-Fender | Trioxin: None do EVERYTHING. |
| 20:09.30 | [TK]D-Fender | Trioxin: Some you can add a few manual bits or bolt-on's and get to work. Some require more effort. Some combinations will become a futile mess |
| 20:10.27 | Trioxin | okay so out of the solutions I listed which to for sure stay away from? |
| 20:10.48 | Trioxin | that's a better question to ask |
| 20:11.01 | [TK]D-Fender | elastix & PIAF |
| 20:11.10 | [TK]D-Fender | Can't directly speak for the others |
| 20:11.19 | [TK]D-Fender | The FS stuf... you should be asking about elsewhere |
| 20:11.43 | Trioxin | k, forgetting FS, anyone else care to weigh in on what not to use? |
| 20:20.27 | glNito | [TK]D-Fender: What books would you recommend for learning Asterisk? |
| 20:20.34 | [TK]D-Fender | ~book |
| 20:20.39 | 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 |
| 20:20.45 | glNito | thanks |
| 20:21.06 | glNito | That's the one I was looking at so I'll go ahead and pick it up |
| 20:21.38 | [TK]D-Fender | Everything else is just a little specific reading on any given piece. This helpes with the foundation |
| 20:23.19 | *** join/#asterisk Uranio-235 (c837bdac@gateway/web/freenode/ip.200.55.189.172) |
| 20:24.09 | Uranio-235 | hi there, after setup jabber and motif, when make a call, asterisk show the following error |
| 20:24.12 | Uranio-235 | Error loading module 'chan_motif.so': /usr/lib/asterisk/modules/chan_motif.so: cannot open shared object file: No such file or directory |
| 20:24.33 | Uranio-235 | of curse, the file does not exist, in a compiled asterisk |
| 20:24.57 | Uranio-235 | what options shall I use in order to compile those modules? or what I doing wrong |
| 20:26.12 | [TK]D-Fender | You'd use the SOURCE to compile |
| 20:26.33 | [TK]D-Fender | And by "wrong" ... that'd be "not getting the source and compiling it" AKA (doing the job". |
| 20:26.50 | Uranio-235 | I compiled via AUR |
| 20:27.09 | Uranio-235 | not eve xmpp module seem to be working |
| 20:27.21 | Uranio-235 | sorry for my patethic english |
| 20:27.26 | [TK]D-Fender | Then you are likely missing pre-requisites. |
| 20:27.32 | [TK]D-Fender | menuconfg should tell you... |
| 20:28.10 | Uranio-235 | menuconfg??? |
| 20:28.20 | [TK]D-Fender | menuconfg |
| 20:28.32 | [TK]D-Fender | the thing you should be using to set the compile options. |
| 20:28.34 | [TK]D-Fender | +i |
| 20:29.48 | Uranio-235 | ah... aur compiling seem to be no execunting that... |
| 20:30.10 | [TK]D-Fender | AUR doesn't sound like the normal process at all... |
| 20:30.17 | [TK]D-Fender | and on that note... |
| 20:30.19 | [TK]D-Fender | heads home |
| 20:31.25 | *** join/#asterisk jeffspeff (~jeff.clay@12.49.160.131) |
| 20:31.47 | Uranio-235 | thanks |
| 20:32.32 | *** join/#asterisk ghost75 (~quassel@p5DE547C5.dip0.t-ipconnect.de) |
| 20:33.17 | ghost75 | somebody has knowledge why an sip device does an unregister during call without reason? |
| 20:34.14 | Uranio-235 | ghost75: often happen me, when the server is overload |
| 20:37.40 | ghost75 | when device cant reach server? |
| 20:47.05 | Uranio-235 | it obviusly fall down |
| 20:47.23 | Uranio-235 | but when the server is full for me, sometimes, some user get disconected |
| 20:47.53 | ghost75 | was only 1 call during that time |
| 20:48.09 | ghost75 | maybe just cable :< |
| 20:55.39 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 21:06.49 | Uranio-235 | what I need to compile res_motif |
| 21:06.51 | Uranio-235 | ?? |
| 21:07.36 | Uranio-235 | I have no ikseme, can I use openssl only instead? |
| 21:09.23 | file | iksemel is required |
| 21:09.58 | ghost75 | just install all packages ;) |
| 21:17.37 | Uranio-235 | ok, iksemel is not on the arch repo :-/ not even in AUR |
| 21:17.52 | Uranio-235 | ghost75: :-/ not enough disk space for the operation |
| 21:18.27 | Uranio-235 | have nica day/after/night |
| 21:18.33 | Uranio-235 | nice* |
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