| 00:05.26 | *** join/#asterisk Cubber (~ronny@mail.adirondackitsolutions.com) |
| 00:29.53 | *** join/#asterisk jaytagdamian (~jaytagdam@office.jaytag.co.uk) |
| 00:33.27 | *** join/#asterisk italorossi (~Adium@179.156.47.253) |
| 00:36.17 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 00:45.31 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 01:01.26 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:29.07 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 01:32.00 | *** join/#asterisk TandyUK (~admin@2a02:13a0:a006:1:0:dead:beef:cafe) |
| 01:49.31 | *** join/#asterisk bl4ckb1rd (~z_freenod@209.141.50.124) |
| 01:58.12 | *** join/#asterisk zway (474c9f97@gateway/web/freenode/ip.71.76.159.151) |
| 01:58.31 | zway | anyone have any experience getting Asterisk running on Azure VM? |
| 02:08.13 | *** join/#asterisk ks3 (~ks3@cpe-24-209-22-231.cinci.res.rr.com) |
| 02:09.53 | *** join/#asterisk resist0r (~resist0r@unaffiliated/mrresist0r) |
| 02:23.32 | *** join/#asterisk d00gster_ (~d00gster@unaffiliated/d00gster) |
| 02:33.05 | *** join/#asterisk saratogga (~saratogga@ip98-165-139-238.ph.ph.cox.net) |
| 02:45.42 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 03:07.14 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 03:19.28 | brieweb | I installed asterisk and I am looking at the "Linux Networking Cookbook" by Carla Schroder. |
| 03:19.44 | brieweb | She says that you start asterisk and type at the cli the following |
| 03:19.48 | brieweb | dial 1000 |
| 03:20.08 | brieweb | but, I tried that and it says "No such command 'dial 1000'" |
| 03:20.15 | brieweb | I imagine that things have changed? |
| 03:22.30 | *** join/#asterisk vinhdizzo (~vinh@cpe-45-48-34-12.socal.res.rr.com) |
| 03:25.34 | [TK]D-Fender | Ditch the concept of dialing from CLI |
| 03:25.42 | [TK]D-Fender | Get a softphone or some other device talking to * |
| 03:31.27 | brieweb | any tips for getting started? |
| 03:31.40 | brieweb | I am looking at this book and the README for Asterisk |
| 03:33.38 | [TK]D-Fender | Go for the officially recomended book. |
| 03:33.40 | [TK]D-Fender | ~book |
| 03:33.40 | 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 |
| 03:33.49 | [TK]D-Fender | Written by core devs for * |
| 03:34.08 | [TK]D-Fender | <brieweb> I installed asterisk and I am looking at the "Linux Networking Cookbook" by Carla Schroder. <- how does this even sound like it talks about Asterisk? |
| 03:48.57 | *** join/#asterisk saratogga_ (~saratogga@ip98-165-139-238.ph.ph.cox.net) |
| 03:53.19 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 04:01.23 | brieweb | the book has a whole section on Asterisk |
| 04:07.42 | brieweb | I am checking out the Asterisk book now |
| 04:21.18 | *** join/#asterisk bmg505 (~leon@196-215-47-220.dynamic.isadsl.co.za) |
| 04:34.05 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 04:36.21 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 04:44.03 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 05:09.40 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 05:13.00 | *** join/#asterisk coinbird (~coinbird@ip-178-203-251-11.hsi10.unitymediagroup.de) |
| 05:31.31 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 05:35.10 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 05:54.07 | *** join/#asterisk KerioMorgan (~Adium@mail.kerio.com) |
| 05:59.14 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-sshrihrfwbtzrlen) |
| 06:02.08 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 06:10.37 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 06:34.23 | Sophira | Hi. I'm having issues with NAT. I have a SIP user (using TLS) who is behind NAT and registered to my server on the Internet (which is not behind NAT). I have "nat=force_rport,comedia" in their friend definition, and have verified that the client is sending RTP to Asterisk and that Asterisk is receiving them, but Asterisk is trying to send its own RTP packets to the local (192.168.*.*) address of the clie |
| 06:34.29 | Sophira | nt. |
| 06:35.31 | Sophira | I'm using Asterisk v11.17.1. |
| 06:37.37 | Sophira | (Well, I should say that the client is sending RTP to the correct server IP and that they show up on a packet sniff of the IP, but I don't think Asterisk is accepting them.) |
| 06:38.43 | Sophira | (because a Record() doesn't produce anything more than a 44-byte WAV file) |
| 06:51.16 | *** join/#asterisk goddva (~goddva@77.40.154.242) |
| 06:56.53 | *** join/#asterisk tparcina (~tomo@cpe-109-60-19-121.st3.cable.xnet.hr) |
| 07:04.01 | *** join/#asterisk themayor (~themayor@unaffiliated/themayor) |
| 07:04.02 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 07:04.02 | *** join/#asterisk stephan48 (stephan@opennic/stephan) |
| 07:04.03 | *** join/#asterisk Gugge (gugge@92.246.2.105) |
| 07:04.04 | *** join/#asterisk wdoekes (~walter@wjd.osso.nl) |
| 07:04.12 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 07:04.22 | *** join/#asterisk pjm (~pjm@uhfsatcom.plus.com) |
| 07:04.24 | *** join/#asterisk Marquel (~Marquel@fuchsfanclub/allerdings/marquel) |
| 07:04.26 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 07:05.00 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 07:05.18 | *** join/#asterisk Qwell (~north@asterisk/developer/Qwell) |
| 07:05.19 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 07:06.01 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 07:06.31 | *** join/#asterisk ketas (~ketas@229-211-191-90.dyn.estpak.ee) |
| 07:07.44 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 07:10.14 | *** join/#asterisk doome_ (~doome@94-21-148-51.pool.digikabel.hu) |
| 07:12.03 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 07:33.10 | *** join/#asterisk lumasepa (~gestoip@193.145.124.30.local.ull.es) |
| 07:37.09 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:56.58 | *** join/#asterisk jaytagdamian (~jaytagdam@office.jaytag.co.uk) |
| 08:03.27 | *** join/#asterisk AviiNL (~AviiNL@185.21.52.255) |
| 08:06.02 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:06.55 | *** join/#asterisk bounceman (~bounceman@185.32.9.250) |
| 08:07.11 | bounceman | What do you think would be the best way to determine whenever asterisk is done handeling a .call file? |
| 08:07.21 | bounceman | Meaning, the call has been completed. |
| 08:09.59 | bounceman | polling cdr database? |
| 08:13.44 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
| 08:14.21 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
| 08:19.16 | WIMPy | Give yourself feedback from the dialplan. |
| 08:38.56 | *** join/#asterisk l0st-soul (~none@2001:920:4009:a::caca) |
| 08:39.52 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 08:39.52 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 08:48.31 | *** join/#asterisk CeBe1 (~CeBe@a81-14-224-229.net-htp.de) |
| 08:49.57 | *** join/#asterisk bounceman (~bounceman@185.32.9.250) |
| 08:50.02 | bounceman | damn the disconncet |
| 08:50.10 | bounceman | WIMPy: sounds complicated? |
| 08:50.21 | bounceman | I know, complicated are sometimes good. |
| 08:55.08 | brieweb | asterisk doesn't support pulseaudio? I had to run it using "padsp asterisk -vvvc" |
| 08:55.14 | brieweb | to simulate the /dev/dsp |
| 08:55.32 | brieweb | it crashed when I tried calling 500 |
| 08:56.08 | brieweb | chan_oss.c:786 oss_indicate: Don't know how to display condition 26 on Console/dsp |
| 08:56.26 | brieweb | or, when I dialed using the following: |
| 08:56.35 | brieweb | console dial 500 |
| 08:58.34 | brieweb | can I modify something is oss.conf? |
| 09:00.41 | *** join/#asterisk doome_ (~doome@82.150.48.146) |
| 09:03.28 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:10.04 | WIMPy | bounceman: It depends entirely on your application. |
| 09:22.24 | bounceman | it's kinda basic, just generates a call file and a tcpdump capture. I need to know when to end the tcpdump |
| 09:23.11 | bounceman | or I can just limit the call in the callfile |
| 09:23.16 | bounceman | and use that number, hmm |
| 09:26.47 | *** join/#asterisk bof23 (~Thunderbi@185.13.183.107) |
| 09:27.07 | *** join/#asterisk Iamnach0 (~Iamnacho@ip72-213-62-18.om.om.cox.net) |
| 09:29.00 | *** join/#asterisk defsdoor (~andy@cpc73037-sutt4-2-0-cust62.19-1.cable.virginm.net) |
| 09:45.13 | bounceman | WaitTime: <number> - How many seconds to wait for an answer before the call fails (ring cycle). Defaults to 45 seconds. |
| 09:45.21 | bounceman | Is there a way to limit the actual call length? |
| 09:45.42 | WIMPy | 'core show function TIMEOUT' |
| 09:49.28 | bounceman | are all functions available in the .call file? |
| 09:54.42 | WIMPy | You can use them via SetVar. |
| 09:57.43 | bounceman | Setvar: ${TIMEOUT}=10 |
| 09:57.47 | bounceman | something like that? |
| 09:57.54 | bounceman | or is TIMEOUT=10 enough |
| 09:58.37 | WIMPy | I guess you should take a look at the dialplan basics for tasic syntax stuff. |
| 09:58.42 | WIMPy | ~book |
| 09:58.42 | 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 |
| 10:03.10 | bounceman | Am not a book type. Anyway I've updated my dialplan now with Set(TIMEOUT(absolute)=45) |
| 10:03.15 | bounceman | so let us try it out |
| 10:07.33 | bounceman | god I love TT-monkeys |
| 10:10.42 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
| 10:17.54 | *** join/#asterisk rsanting (~rsanting@217.21.195.34) |
| 10:33.01 | *** join/#asterisk stefan27 (~stefan27@212.247.4.149) |
| 10:33.27 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5EA87.dip0.t-ipconnect.de) |
| 10:39.48 | stefan27 | unwanted invites to my asterisk goes to [default] which logs recvip and hangs up. A fail2ban process then checks logs and bans ips that fail too much. This has been good enough for a long time, but it doesn't protect against someone spamming invites and temporarily eating up all rtp-ports. What are best practices with concerns like this |
| 10:41.49 | WIMPy | Why do you accept the calls? |
| 10:42.40 | WIMPy | If you immediately reject the clls, there shouldn't be any RTP going on. |
| 10:43.28 | wdoekes | WIMPy: IIRC, ports are allocated really early, so it doesn't have to reject the call later on if it turns out that there is no RTP port free |
| 10:43.37 | WIMPy | Hmm. Coming to think of it: Do you have inband progress enabled? |
| 10:44.07 | wdoekes | (I could recall wrong, but I remember some discussion about this earlier) |
| 10:44.49 | WIMPy | Hmm. Are you saying RTP ports get allocated before any SDP? |
| 10:45.20 | stefan27 | yes |
| 10:45.40 | WIMPy | I'm doing the same thing as stefan27 (just without fail2ban) and making sure there's no audio seemed to have fixed the issue for me. |
| 10:46.25 | stefan27 | call is never answered, [default] only does NoOp, System, and Hangup |
| 10:46.34 | stefan27 | Hmm let me check inband progress |
| 10:51.39 | *** join/#asterisk Draecos (~Draecos@1.127.49.11) |
| 10:53.11 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 10:54.48 | stefan27 | WIMPy, what's the name of the option? Is it ;progressinband=no ? That option seems commented out in sip.conf [general] so it should be the default value whatever that is, how can i query it from cli? |
| 10:56.01 | WIMPy | 'sip show settings' |
| 10:56.30 | WIMPy | And you praobably want it to be set to 'never'. |
| 10:56.52 | WIMPy | There's another related option... |
| 10:57.04 | stefan27 | Progress inband: No |
| 10:57.18 | stefan27 | Never is different from No? |
| 10:57.25 | WIMPy | 'prematuremedia'. |
| 10:57.45 | stefan27 | No premature media: Yes |
| 10:58.09 | stefan27 | So I guess that means I dont have premature media nor progress inband, yet rtp ports are allocated |
| 10:58.16 | WIMPy | Yuk. A negative option. |
| 10:58.16 | stefan27 | That does not happen for you? |
| 10:58.52 | WIMPy | Well, I haven't tried on purpose, but I didn't have any issues any more. |
| 10:59.06 | WIMPy | But I have 'progressinband=never'. |
| 10:59.47 | WIMPy | And also 'prematuremedia=no' but So far I haven't fond a combination that works in all cases. |
| 11:00.02 | *** join/#asterisk iamb (~iamb@unaffiliated/iamb) |
| 11:00.30 | WIMPy | I probably should try different settings for different peers. But that's another topic. |
| 11:02.45 | WIMPy | Does it send any messages with SDP for you? |
| 11:06.28 | stefan27 | no, just trying into declined |
| 11:11.02 | stefan27 | even with prematuremedia=no and progressinband=never in sip.conf there's rtp ports before any sdp |
| 11:12.02 | WIMPy | I guess I have just been lucky then. |
| 11:12.43 | stefan27 | Hmm this issue I must dig deeper in, thanks for help and let me know if you think of a new explanation :) |
| 11:14.52 | WIMPy | Maybe I need to invite the bad guys to find out what's really happening. |
| 11:21.49 | *** join/#asterisk doome_ (~doome@82.150.48.146) |
| 11:30.24 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 11:40.31 | *** join/#asterisk keithf (~keithf@ool-2f151dc2.static.optonline.net) |
| 11:40.43 | *** join/#asterisk coinbird (~coinbird@ip-178-203-251-11.hsi10.unitymediagroup.de) |
| 11:41.47 | bounceman | WIMPy: you know if there is a way to tell asterisk to wait 2 seconds before dialing the .call? |
| 11:42.14 | bounceman | I guess Wait(2) in the dialplan will not stop it from sending the INVITE |
| 11:48.25 | WIMPy | Yes |
| 12:05.39 | *** join/#asterisk italorossi (~Adium@177.65.203.21) |
| 12:25.15 | *** join/#asterisk AviiNL (~AviiNL@185.21.52.255) |
| 12:31.26 | *** join/#asterisk Draecos (~Draecos@203-121-194-218.e-wire.net.au) |
| 12:35.53 | *** join/#asterisk jasonwert (~quassel@71.89.137.28) |
| 12:36.17 | Gugge | bounceman: set it to dial some LOCAL/extension, and the use Wait(2) in there, before you use Dial() |
| 12:42.20 | *** join/#asterisk saxa (~sasa@2-234-206-126.ip224.fastwebnet.it) |
| 12:42.34 | saxa | hi ppl, can anybody help me out with this ? |
| 12:42.40 | saxa | http://pastebin.com/mKGXUiGn |
| 12:43.06 | saxa | I had to buy a new motherboard as the old one went away |
| 12:43.31 | saxa | now i start getting those troubles with seems irq and kernel |
| 12:43.40 | saxa | system is same as was before. |
| 12:47.37 | *** join/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
| 13:01.29 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:02.24 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 13:06.55 | *** join/#asterisk mub (~jub@65-123-23-2.dia.static.qwest.net) |
| 13:07.57 | *** join/#asterisk doome_ (~doome@82.150.48.146) |
| 13:09.15 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 13:10.57 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 13:31.03 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 13:32.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:51.01 | *** join/#asterisk Garibaldo (~smuxi@187.113.98.25) |
| 13:51.25 | *** join/#asterisk Lope (~Lope@192.228.145.10) |
| 13:52.35 | Lope | if I specify dtmfmode=rfc2833 inside [general] in sip.conf. does that mean that all my other sip sections will use rfc2833 also, or do I need to specify it in every section explicitly? |
| 13:53.04 | [TK]D-Fender | It will apply to any that you don't set explicitly |
| 13:53.26 | [TK]D-Fender | but I prefer to actually do the job explicitly every time so I'm never dealing with assumptions |
| 13:54.20 | bounceman | Can anyone see why I am getting this error? http://pastebin.com/JV7T7Wma |
| 13:55.10 | Lope | okay. Last time I worked with this I was having an issue with a call transfer (the context immediately Dial's my SIP phone in extensions.conf when it gets an incoming call) Should RFC2833 work in that case? |
| 13:57.21 | Lope | so should it work that my phone that gets dialled can send DTMF back to the caller with RFC2833? |
| 13:58.24 | [TK]D-Fender | it applies to the leg it applies to |
| 13:58.46 | [TK]D-Fender | that is what it will use for ONLY that hop. Each can be differnt and it needs to match what the other side is expecting |
| 13:58.54 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 13:59.06 | [TK]D-Fender | We also don't know what your actual issue was with "call transfer" |
| 13:59.06 | Lope | what is a leg? the connection from my asterisk to my softphone? |
| 13:59.17 | [TK]D-Fender | What phone were you having this with exactly? |
| 13:59.28 | Lope | SFLphone (linux softphone)\ |
| 13:59.36 | [TK]D-Fender | Person1 -> * -> Person2 |
| 13:59.38 | bounceman | ah nvm, db typo |
| 13:59.39 | [TK]D-Fender | That is 2 channels |
| 13:59.43 | [TK]D-Fender | each can be different |
| 13:59.49 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 14:00.00 | l0st-soul | hi there, i'm wondering if anyone else uses AEL for the dialplan, or if I am alone? |
| 14:00.03 | Lope | I see, checking my sip.conf to make sure |
| 14:00.09 | l0st-soul | I really like AEL but I feel not many ppl are using it |
| 14:00.40 | [TK]D-Fender | Correct |
| 14:01.08 | [TK]D-Fender | AEL = cheap parser that can only do less than you can do in standard notation since that is what it gets parsed back to anyway |
| 14:01.50 | [TK]D-Fender | So it is possible to introduce bugs that don't need to exist, and makes debugging actualy flow harder with the fluff that it adds and the terms it screws up and gets backwards |
| 14:01.57 | Lope | TKD: so when I have a call transfer and my softphone presses a DTMF tone, will asterisk pass that to the calling party using whatever DTMF tech is selected for the caller? Can I see the DTMF tones inside the asterisk -r terminal? |
| 14:02.48 | [TK]D-Fender | with core debug you can see what * acks as DTMF |
| 14:02.51 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 14:03.26 | [TK]D-Fender | it you tell it an OOB option it won't go looking at the audio. Now if the audio pases through and the other side DOPES decode that... then it will do what it does |
| 14:04.37 | *** join/#asterisk d00gster (~d00gster@unaffiliated/d00gster) |
| 14:05.44 | Lope | TKD: awesome, it worked! |
| 14:06.02 | Lope | thank you :) |
| 14:06.08 | l0st-soul | [TK]D-Fender: ok thanks for the info. It's true it's just translating into regular dialplan, but being a php user i find the if/elseif/else, case, etc, much easier to understand than jumps in dialplan. Makes it easier to insert or move dialplan parts around |
| 14:06.29 | l0st-soul | regular dialplan feels like basic, while AEL feels like PHP (kinda) |
| 14:06.55 | Lope | however no DTMF digits appeared inthe asterisk console. I had both debug and verbose on 10. |
| 14:07.17 | l0st-soul | but yeah, the clutter it translates to might not be nice |
| 14:08.30 | [TK]D-Fender | l0st-soul: Indeed, but I'd rather have control and be able to debug cleanly that see it one way and have to hope I can reverse-engineer it for the other |
| 14:08.47 | [TK]D-Fender | "both debug"? |
| 14:08.47 | l0st-soul | fair enough |
| 14:08.48 | [TK]D-Fender | as in? |
| 14:09.08 | Lope | core set verbose 10 |
| 14:09.11 | Lope | core set debug 10 |
| 14:10.32 | [TK]D-Fender | core debug should show you DTTM start/end notifications |
| 14:10.50 | [TK]D-Fender | Uless it is just dumb audio passing through and you're luck that the other end processes both |
| 14:12.16 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:15.08 | Lope | TKD: I don't know. I don't see any strings containing DTTM in the console at the time I used DTMF (that was received on the other side) |
| 14:19.42 | *** join/#asterisk eschmidbauer__ (~eschmidba@unaffiliated/eschmidbauer) |
| 14:21.07 | *** join/#asterisk TandyUK2 (~admin@2a02:13a0:a006:1:0:dead:beef:cafe) |
| 14:23.14 | *** join/#asterisk ketas- (~ketas@229-211-191-90.dyn.estpak.ee) |
| 14:24.14 | Lope | night night |
| 14:28.26 | *** join/#asterisk Svedrin (svedrin@elwing.funzt-halt.net) |
| 14:29.11 | *** join/#asterisk stephan48 (stephan@opennic/stephan) |
| 14:43.12 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 14:45.20 | *** join/#asterisk K0HAX (~K0HAX@shellhost.home.englehorn.com) |
| 14:49.34 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 14:57.32 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-xlklufsmwvzcamaj) |
| 14:59.39 | *** join/#asterisk zway (474c9f97@gateway/web/freenode/ip.71.76.159.151) |
| 15:07.09 | *** join/#asterisk ruied (~ruied@125.44.62.94.rev.vodafone.pt) |
| 15:19.37 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 15:21.41 | *** join/#asterisk jaytagdamian (~jaytagdam@office.jaytag.co.uk) |
| 15:26.45 | *** join/#asterisk kharwell (kharwell@nat/digium/x-gsgkgehcgwbizzws) |
| 15:32.02 | *** join/#asterisk Garibaldo (~smuxi@187.114.210.30) |
| 15:52.38 | *** join/#asterisk jwpierce3 (~jwpierce3@50-82-153-66.client.mchsi.com) |
| 15:55.25 | *** join/#asterisk acidfoo_ (~acidfoo@69-165-149-25.dsl.teksavvy.com) |
| 16:11.08 | *** join/#asterisk Garibaldo (~smuxi@187.114.210.30) |
| 16:13.54 | *** join/#asterisk captain118 (~Adium@247.sub-70-193-96.myvzw.com) |
| 16:14.52 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:14.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:17.57 | *** join/#asterisk hfb (~hfb@96.247.52.240) |
| 16:27.16 | *** join/#asterisk AviiNL (~AviiNL@185.21.52.255) |
| 16:27.29 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 17:07.11 | *** join/#asterisk jab416171 (~jab416171@c-76-27-96-12.hsd1.ut.comcast.net) |
| 17:10.48 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 17:12.42 | *** join/#asterisk karelk (~karel@31.10.159.144) |
| 17:19.32 | *** join/#asterisk Kevin` (~kevin@99-96-57-112.lightspeed.milwwi.sbcglobal.net) |
| 17:20.02 | Kevin` | is there a way to either prevent asterisk from continuously keeping the sound card open or allow it to work with pulseaudio? |
| 17:20.27 | [TK]D-Fender | Disable the modules that use it |
| 17:20.37 | [TK]D-Fender | chan_alsa, etc |
| 17:20.47 | Kevin` | I want it to send sound output when a call requires it |
| 17:21.17 | [TK]D-Fender | like? |
| 17:21.51 | Kevin` | i'm sending audio through a radio connected to an audio output. the driver for audio is alsa |
| 17:22.15 | Kevin` | if I have asterisk use chan_alsa, it monopolizes the sound card and nothing else can play sound |
| 17:22.34 | Kevin` | if enable pulseaudio, sound is choppy for some reason |
| 17:22.40 | Kevin` | like it's using the wrong sample rate |
| 17:24.06 | Kevin` | similar thing happens if I use hw:0,0 or similar specific alsa devices instead of default with pulse disabled |
| 17:24.11 | Kevin` | (wrong sample rate) |
| 17:24.43 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 17:26.51 | *** join/#asterisk RadicalDev (~jfindley@2601:2c3:8002:ec00:b947:f87b:267e:5a02) |
| 17:28.49 | [TK]D-Fender | Tried OSS? |
| 17:29.36 | Kevin` | I enabled oss->alsa emulation in the kernel last time, I guess I could try it. that's kind of just working around asterisk bugs if it works though.. and how do I tell it which alsa device to use? |
| 17:30.14 | [TK]D-Fender | not sure |
| 17:32.32 | Kevin` | using oss emulation, asterisk works but again hangs the sound card, since pulse can't intercept it |
| 17:39.54 | *** join/#asterisk coinbird (~coinbird@ip-178-203-251-11.hsi10.unitymediagroup.de) |
| 17:48.40 | *** join/#asterisk Khronos (~Khronos@72-238-20-238.res.bhn.net) |
| 17:48.44 | Khronos | Hi guys. |
| 17:49.07 | Khronos | Anyone know of solutions I can use for voip call qulity monotring on aws ec2? |
| 17:49.11 | Khronos | quality |
| 18:00.08 | *** join/#asterisk jasonwert (~quassel@75-134-81-98.static.aldl.mi.charter.com) |
| 18:15.56 | Kevin` | how can I set the sample rate on chan_alsa? it's forcing transcoding to 8khz |
| 18:16.46 | WIMPy | AFAIK it doesn't support anything else. |
| 18:22.25 | *** join/#asterisk resist0r (~resist0r@unaffiliated/mrresist0r) |
| 18:36.23 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 18:56.06 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 19:02.18 | *** part/#asterisk Khronos (~Khronos@72-238-20-238.res.bhn.net) |
| 19:10.09 | *** join/#asterisk captain118 (~Adium@247.sub-70-193-96.myvzw.com) |
| 19:10.21 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 19:11.22 | *** join/#asterisk grim_ (~bikerGrim@rrcs-98-102-12-82.central.biz.rr.com) |
| 19:11.42 | grim_ | hello all.... got a weird problem here that I'm having a hard time wrapping my brain around. |
| 19:11.55 | grim_ | I have a digium D40 and an yealink T42G |
| 19:12.30 | grim_ | Both factory reset, then configured with the bear minimum, to register with asterisk. |
| 19:12.54 | grim_ | both configured with same username and password. |
| 19:13.32 | grim_ | when I connect the digium phone everything works fine. calling other phones at the office here, calling other phones connected to a cisco call manager system at our other office, calling out to my cell phone. |
| 19:14.35 | grim_ | I disconnect the digium, connect the yealink, I can call other phones connected here, all good. I can call my cell phone, all good. However, when I call to the remote office on the cisco call manager, I have one way audio. far end can here me, I can't here them. |
| 19:14.50 | grim_ | I've checked all nat settings on the yealink. all settings disabled. |
| 19:15.00 | grim_ | direct media on asterisk set to no. |
| 19:15.11 | [TK]D-Fender | Show us actual calls |
| 19:15.40 | grim_ | local office to remote office is metro ethernet. all routed, no nat. |
| 19:15.41 | [TK]D-Fender | we should SEE what it is doing and not assuming the settings you think are in effect actually are |
| 19:16.20 | grim_ | what do you mean by see the actual calls. there's a lot of various information that can be obtained. are you looking for a packet dump on the calls? |
| 19:16.35 | [TK]D-Fender | The full SIP comms of the calls |
| 19:16.42 | [TK]D-Fender | proof about what's offered to where |
| 19:16.43 | grim_ | ah ok. brb |
| 19:18.36 | *** join/#asterisk Yast (~lluisyast@229.Red-217-126-124.staticIP.rima-tde.net) |
| 19:36.27 | grim_ | ok, pulled the sip debug from both phones. what I'm noticing is that when the yealink places the call to the system my sip uri is: sip:1268@10.31.55.10, however when the call is forwarded to the cisco call manager the uri changes to: sip:asterisk@10.31.55.10. This doesn't happen with the digium phone places the call. |
| 19:38.16 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 19:38.27 | Kevin` | what am I doing wrong to cause this? it has to be something stupid |
| 19:38.31 | Kevin` | username mismatch, have <central>, digest has <s> |
| 19:39.29 | grim_ | yealink -> http://pastebin.com/DJXc93Sg |
| 19:39.40 | grim_ | digium -> http://pastebin.com/DJXc93Sg |
| 19:44.16 | [TK]D-Fender | ... |
| 19:46.11 | [TK]D-Fender | waits for the actual 2nd link |
| 19:49.30 | grim_ | sorry.... yealink -> http://pastebin.com/G69815QE |
| 19:50.24 | grim_ | call to cisco line 501 |
| 19:52.24 | Kevin` | is there an example config for an sip trunk between two asterisk systems? |
| 19:55.17 | [TK]D-Fender | Kevin`: Tons. Basically it's the same as for any other device or service |
| 19:55.19 | [TK]D-Fender | SIP is SIP is SIP |
| 19:55.35 | [TK]D-Fender | You are eitgher authing by username ... or IP, with PW. Or not. |
| 19:56.39 | [TK]D-Fender | grim_: I'm not yet seeing the orrigin of that "asterisk" in the from: |
| 19:57.02 | [TK]D-Fender | grim_: However you are in a now old version. We recommend up upgrade immediately. |
| 19:57.18 | [TK]D-Fender | grim_: And prove this is not a but that's already been fixed |
| 19:58.01 | Kevin` | [TK]D-Fender: I must be an idiot then, because i've done this before and i'm lookin at the voip-info docs |
| 19:58.08 | Kevin` | and yet it's failing to register correctly |
| 19:58.23 | grim_ | <PROTECTED> |
| 19:59.48 | Kevin` | http://pastebin.com/24nCa0N7 |
| 20:01.24 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 20:02.24 | grim_ | Kevin`: easiest, cleanest, and simplest way to setup a trunk betweek 2 boxes.... http://pastebin.com/C3K9AsZi |
| 20:02.51 | Kevin` | needs auth |
| 20:09.14 | *** join/#asterisk acidfoo_ (~acidfoo@69-165-149-25.dsl.teksavvy.com) |
| 20:09.33 | grim_ | this should work. for auth. http://pastebin.com/hteeAZaq |
| 20:11.53 | grim_ | scratch that Kevin` use this one instead.... that way it will allow you to set the context for calls from server 1 to server 2 http://pastebin.com/Ls7cZtL3 |
| 20:16.12 | RadicalDev | how do I get asterisk to continue in the dialplan when a DIAL(DAHDI/x/y) command is cancelled by the caller? |
| 20:16.45 | RadicalDev | I've specified g as a parameter, which works if the caller hangs up after the call is bridged, but if no one answers and the call is canceled, execution stops there |
| 20:16.58 | *** join/#asterisk xnaron (~xnaron@S0106b4750e5de3b2.ed.shawcable.net) |
| 20:17.24 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 20:18.07 | rmudgett | RadicalDev: The call has ended. Look at the h exten and hangup-handlers. |
| 20:19.19 | *** join/#asterisk w9sh (~sph@c-73-54-187-250.hsd1.ga.comcast.net) |
| 20:21.59 | RadicalDev | rmudgett: Will do, thanks. |
| 20:23.27 | RadicalDev | rmudgett: Perfect =) |
| 20:24.00 | Kevin` | grim_: chasing other problems right now, will test soon |
| 20:24.32 | Kevin` | grim_: there's no users specified, does that just make it ignore usernames? |
| 20:25.48 | grim_ | the username is server2 |
| 20:26.00 | grim_ | so you could make that whatever you like. |
| 20:33.05 | [TK]D-Fender | Kevin`: Your issue is that when you pass that call over your FROM is saying a different name than the peer normally uses |
| 20:33.08 | [TK]D-Fender | thus the mismatch |
| 20:33.17 | [TK]D-Fender | CID is effectively screwing this up. |
| 20:33.52 | [TK]D-Fender | set "fromuser=THEUSERNAME", "sendrpid=yes", "trustrpid=yes" on both sides |
| 20:34.05 | [TK]D-Fender | And that should fix the auth (for that reason) |
| 20:37.04 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 20:46.32 | *** join/#asterisk bl4ckb1rd (~z_freenod@209.141.50.124) |
| 20:49.14 | Kevin` | [TK]D-Fender: didn't make any difference |
| 20:49.18 | [TK]D-Fender | show us |
| 20:49.23 | [TK]D-Fender | Don't just say "didn't work" |
| 20:49.25 | [TK]D-Fender | ~pb |
| 20:49.26 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:49.27 | [TK]D-Fender | ^^^ |
| 20:49.39 | [TK]D-Fender | configs & the actual output. |
| 20:49.47 | Kevin` | it's pretty much the same, but i'll pastebin it all |
| 20:55.07 | *** join/#asterisk gusto (~gusto@77.47.123.111.dynamic.cablesurf.de) |
| 20:56.03 | Kevin` | [TK]D-Fender: http://pastebin.com/KKaxuvvf |
| 20:57.22 | [TK]D-Fender | your names are wrong there |
| 20:57.29 | [TK]D-Fender | You shouldn't have a separate username & fromuser |
| 20:57.36 | [TK]D-Fender | SAME on both |
| 20:57.50 | [TK]D-Fender | Also in modern versions "username" needs to be changed to "defaultuser" |
| 20:57.53 | Kevin` | neither of them are "s", but i'll try that |
| 20:57.59 | [TK]D-Fender | And you shoul dbe using "type=peer", not "friend" |
| 20:58.17 | [TK]D-Fender | register => csc1:3wejkcas@server <--- THIS is what sets the "s" |
| 20:58.23 | [TK]D-Fender | because you are registering. |
| 20:58.30 | [TK]D-Fender | it sets the expected contact |
| 20:58.46 | [TK]D-Fender | so fix all the usernames to match on both sides with both fields each |
| 20:59.12 | Kevin` | I don't see an "s" in that line |
| 20:59.25 | Kevin` | I need to explicitly set fromuser and defaultuser to the same thing? |
| 21:00.32 | [TK]D-Fender | yes |
| 21:01.21 | [TK]D-Fender | you get "s" because you didn't specify a contact at the end of that. |
| 21:01.36 | [TK]D-Fender | But this isn't important if you get the rest right |
| 21:02.43 | gusto | WIMPy: hey |
| 21:03.17 | Kevin` | I now have "csc1" specified 4 times on the client and 3 times on the server, and it's still using "s" |
| 21:03.28 | Kevin` | also it feels very wrong to have to change the same thing 4 times |
| 21:03.47 | [TK]D-Fender | they sides have to MATCH |
| 21:04.16 | Kevin` | they have different ips and one registers with the other, they can't match |
| 21:04.24 | [TK]D-Fender | THE USERNAMEANS have to |
| 21:04.41 | Kevin` | right, I have the username specified 4 times on the client and 3 times on the server. all the SAME name |
| 21:04.49 | [TK]D-Fender | I say "I'm John", then the other side better say "that guy better be John" |
| 21:04.57 | [TK]D-Fender | client? |
| 21:05.00 | [TK]D-Fender | What do you meaen client? |
| 21:05.09 | Kevin` | I mean the one registering |
| 21:05.17 | [TK]D-Fender | 2 systems. |
| 21:05.24 | [TK]D-Fender | That's all we're talking about ehre |
| 21:05.28 | [TK]D-Fender | Show the new configs |
| 21:05.35 | [TK]D-Fender | and the net call attempt |
| 21:05.55 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 21:06.23 | [TK]D-Fender | new* |
| 21:08.05 | Kevin` | http://pastebin.com/G01yJaeg |
| 21:08.27 | [TK]D-Fender | type=friend |
| 21:08.41 | [TK]D-Fender | these were supposed to be changed |
| 21:09.15 | Kevin` | changed now, same result |
| 21:09.56 | [TK]D-Fender | Chow the call |
| 21:10.00 | [TK]D-Fender | Show |
| 21:10.05 | [TK]D-Fender | sip set debug on |
| 21:10.06 | [TK]D-Fender | ^^^^ |
| 21:15.19 | Kevin` | [TK]D-Fender: http://kwzs.be/~remote/asteriskfoo.txt |
| 21:17.44 | [TK]D-Fender | defaultip=10.0.1.78 <- if you're dynamic then you should just remove this |
| 21:17.50 | [TK]D-Fender | Also, are they indeed dynamic? |
| 21:18.07 | Kevin` | short term yes, long term no |
| 21:18.16 | [TK]D-Fender | remove that for starters. |
| 21:18.30 | Kevin` | focus on the problem preventing it from working |
| 21:18.45 | [TK]D-Fender | fix the register to add "/csc1" at the end |
| 21:18.54 | [TK]D-Fender | do that for both |
| 21:19.31 | [TK]D-Fender | And on each add the following: |
| 21:19.38 | [TK]D-Fender | insecure=port,invite |
| 21:19.42 | [TK]D-Fender | directmedia=no |
| 21:19.46 | Kevin` | why insecure invite? |
| 21:19.54 | [TK]D-Fender | fix all those , retest, then pastebin all if it fails |
| 21:20.04 | [TK]D-Fender | Because once they have registered the IP should be trusted. |
| 21:20.19 | [TK]D-Fender | registering alone should get the IP in there |
| 21:20.52 | Kevin` | I don't know yet if that's a guarantee |
| 21:21.14 | [TK]D-Fender | it's fine |
| 21:21.28 | [TK]D-Fender | doesn't change being literally a moving target or not |
| 21:25.17 | Kevin` | the register is being transmitted "correctly" now that I have the usernam specified a fifth time at the end of the register line, but now it's not being added to the sip registry list, I susepct because of "peer" |
| 21:25.53 | [TK]D-Fender | no |
| 21:26.15 | [TK]D-Fender | register => csc1:3wejkcas@csc1 |
| 21:26.26 | [TK]D-Fender | that is no using the HOSTT after the @ |
| 21:26.34 | [TK]D-Fender | you don't specify a "section name" |
| 21:26.40 | Kevin` | I have register => csc1:3wejkcas@csc1/csc1 now |
| 21:26.54 | [TK]D-Fender | register => csc1:3wejkcas@HOSTNAME |
| 21:26.55 | [TK]D-Fender | ^ |
| 21:26.59 | [TK]D-Fender | not "csc1" |
| 21:27.02 | [TK]D-Fender | that is not the host |
| 21:27.12 | [TK]D-Fender | ngtest2.x.com |
| 21:27.14 | [TK]D-Fender | ^ |
| 21:27.15 | Kevin` | the register IS being sent correctly |
| 21:27.16 | [TK]D-Fender | is |
| 21:27.27 | Kevin` | but i'll duplicate the hostname a few times too if it makes you happy |
| 21:27.36 | [TK]D-Fender | this isn't "a few times" |
| 21:27.39 | [TK]D-Fender | I jsut told you to fix the register |
| 21:27.47 | [TK]D-Fender | ONCE |
| 21:28.17 | Kevin` | no change in behavior |
| 21:29.29 | [TK]D-Fender | Okk, You nI haven't seen the fixed versions for several revisions now |
| 21:32.12 | Kevin` | [TK]D-Fender: http://pastebin.com/c8cmvyDN |
| 21:32.42 | [TK]D-Fender | [16:19][TK]D-FenderAnd on each add the following: |
| 21:32.43 | [TK]D-Fender | [16:19][TK]D-Fenderinsecure=port,invite |
| 21:32.45 | [TK]D-Fender | [16:19][TK]D-Fenderdirectmedia=no |
| 21:32.46 | [TK]D-Fender | I told you to add those |
| 21:32.48 | [TK]D-Fender | And you didn't |
| 21:32.53 | [TK]D-Fender | type=friend |
| 21:32.57 | [TK]D-Fender | Told youi to change that as well |
| 21:32.59 | [TK]D-Fender | And you haven't |
| 21:33.28 | [TK]D-Fender | defaultip=10.0.1.78 <-= STILL THERE |
| 21:33.38 | [TK]D-Fender | I do not see you following anything based on this |
| 21:33.49 | [TK]D-Fender | And I'm out of time here. |
| 21:33.51 | [TK]D-Fender | BBL |
| 21:33.57 | Kevin` | I changed friend, and changed it back |
| 21:34.08 | Kevin` | but I don't think you are helping with the relevant problem anyway, so that's fine |
| 21:34.14 | [TK]D-Fender | You need all the pieces in place |
| 21:34.21 | [TK]D-Fender | You can't just change ONE thing then change it back |
| 21:35.25 | Kevin` | [csc1] |
| 21:35.25 | Kevin` | type=peer |
| 21:35.25 | Kevin` | host=dynamic |
| 21:35.25 | Kevin` | defaultuser=csc1 |
| 21:35.25 | Kevin` | fromuser=csc1 |
| 21:35.27 | Kevin` | secret=3wejkcas |
| 21:35.30 | Kevin` | sendrpid=yes |
| 21:35.30 | [TK]D-Fender | PASTBEING |
| 21:35.32 | [TK]D-Fender | aefskdl:Jaskdfjhlijskladjhkdas |
| 21:35.32 | Kevin` | trustrpid=yes |
| 21:35.35 | Kevin` | no change |
| 21:36.08 | [TK]D-Fender | [16:32][TK]D-Fender[16:19][TK]D-Fenderinsecure=port,invite |
| 21:36.09 | [TK]D-Fender | [16:32][TK]D-Fender[16:19][TK]D-Fenderdirectmedia=no |
| 21:36.19 | Kevin` | what does directmedia do? |
| 21:36.40 | Kevin` | I suspect insecure=invite would "fix" this without setting the user ANYWHERE, but that's really not what I want to solve here |
| 21:36.48 | grim_ | [TK]D-Fender> my ast is 13.1 cert 2. that's old? |
| 21:37.00 | [TK]D-Fender | griYes, look at the topic |
| 21:37.29 | [TK]D-Fender | time is up here |
| 21:42.15 | Kevin` | disabling security does indeed "solve" the authentication problem. helpful. |
| 21:55.39 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 22:10.41 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:15.51 | RadicalDev | Can anyone speculate why with dahdi v 2.4.0 and wanpipe 7.0.15 on one box, why unplugging a crossover cable from an A102 installed on the box which is connected to a digium card in another server with a digium PRI card, everything is fine, but reversing that operation causes the first box to hard freeze with no kernel panics or anything log-wise to indicate what happened? |
| 22:16.47 | RadicalDev | * and asterisk 11.21.0 |
| 22:17.19 | RadicalDev | the freeze takes place regardless of whether or not asterisk is running, however. I'm guessing wanpipe may be exploding. |
| 22:21.35 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 22:23.24 | klow | Hello. Using pjsip, I am trying to specify my TLS cipher so that it will use one supported by sngrep , so I can debug sip messages. Sngrep author says TLS_RSA_WITH_AES_128_CBC_SHA and TLS_RSA_WITH_AES_256_CBC_SHA are the supported ciphers, but these strings don't work in the cipher= field . I've also tried AES-256 and aes-256-cbc as cipher strings, no luck. Any ideas ? Thanks |
| 22:24.56 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 22:30.31 | *** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy) |
| 22:33.24 | *** join/#asterisk Draecos (~Draecos@203-121-194-218.e-wire.net.au) |
| 22:36.14 | *** join/#asterisk troyt (~troyt@c-67-161-210-245.hsd1.ut.comcast.net) |
| 22:40.41 | *** join/#asterisk AviiNL (~AviiNL@h178213.upc-h.chello.nl) |
| 22:41.51 | *** join/#asterisk petris (~petris@2604:180:0:ad7::73db) |
| 22:50.50 | *** join/#asterisk Etep (~Etep@S0106b4750e5de38a.cg.shawcable.net) |
| 22:51.39 | *** join/#asterisk captain118 (~Adium@50-81-28-72.client.mchsi.com) |
| 23:00.39 | Etep | Wondering if someone can help here. I'm trying to get ARI going. I installed AsteriskNow. I've switched it to asterisk 13. I've gone through the tutorial at https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI but when I call wscat -c "ws://localhost:8088/ari/events?api_key=asterisk:asterisk&app=hello-world" I get a 404 error. |
| 23:01.15 | Etep | webserver is running on 8088 but it's as if /ari or /ari/events or the hello-world app doesn't exist? Welcome any suggestions. |
| 23:09.01 | DivideBy0 | Etep: you'll get help in here, but #asterisk-ari is dedicated to that |
| 23:09.38 | Etep | Ah ok. Thanks Divide. I'll try there as well. much appreciated. |
| 23:09.50 | DivideBy0 | Etep: you can set "core debug 10" and you'll the actual url it tries to use - and what matches |
| 23:10.11 | Etep | Ah ok. I'll give that a shot. |
| 23:10.22 | DivideBy0 | it'll say "trying /ari" or something |
| 23:10.29 | DivideBy0 | and then "matched /ari" if it did |
| 23:11.59 | DivideBy0 | Etep - also, beginner's tips from an astricon presentation at http://www.markingles.com/ingles_astricon15.pdf |
| 23:12.47 | Etep | ah perfect. thank you! I'll have a read. |
| 23:13.29 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1:c15f:f908:c5d7:8e8a) |
| 23:14.08 | DivideBy0 | try to get swagger working as soon as you can too, it's very helpful. http://ari.asterisk.org you'll have to set allowed_origins or something similar in a config. swagger is really neat |
| 23:14.09 | Etep | Ok best I can see as the problem is "Requested URI [ari/events] has no handler" |
| 23:14.25 | DivideBy0 | makes me wonder if ari is enabled? |
| 23:14.33 | Etep | Ah ok, I read about swagger. I'll get that going. |
| 23:14.44 | DivideBy0 | did you setup a user/pass in http.conf and ari.conf |
| 23:15.17 | DivideBy0 | ari.conf specifically, also, restart asterisk to see if it rereads your changes |
| 23:16.36 | Etep | you know what.. ari_general_additional.conf has the seetings but they were set to no. Looks like when I switched to asterisk 13 it reset my configs. (I figured out that it was running 11 originally so figured that's why it wasn't working originally) doh. so stupid. Ok I'll recheck the configs. |
| 23:17.06 | Etep | to restart asterisk just simply use /etc/init.d/asterisk restart? |
| 23:17.24 | DivideBy0 | <PROTECTED> |
| 23:17.54 | DivideBy0 | sure. I'm usually in the asterisk console and do "core restart now" |
| 23:18.33 | DivideBy0 | freepbx will overwrite, them so you should probably set everything in their web interface |
| 23:19.02 | DivideBy0 | under settings->asterisk rest interface users |
| 23:20.17 | Etep | Got it working! :) |
| 23:20.19 | DivideBy0 | and enable the rest inteface under settings->advanced, "asterisk rest interface" |
| 23:20.21 | DivideBy0 | ok cool |
| 23:20.53 | Etep | Ah good advice. I put my user in the ari_additional_custom.conf for now. But ya I should do through the web int. |
| 23:21.10 | DivideBy0 | in ari_general_custom.conf add allowed_origins=http://ari.asterisk.org |
| 23:21.15 | DivideBy0 | that'll let you use swagger |
| 23:23.49 | *** join/#asterisk mahables (46b574d5@gateway/web/cgi-irc/kiwiirc.com/ip.70.181.116.213) |
| 23:25.00 | mahables | Anyone familiar with the status of RTCP stats in Asterisk 11? It appears to be incomplete. |
| 23:25.23 | Etep | Perfect. hank you very much DivideBy0! |
| 23:25.36 | Etep | er.. thank you even. :) |
| 23:26.12 | DivideBy0 | my pleasure. have a nice night |
| 23:29.07 | *** join/#asterisk Draecos (~Draecos@1.127.49.45) |
| 23:29.51 | *** join/#asterisk Garibaldo (~smuxi@177.159.215.91) |
| 23:32.05 | *** join/#asterisk coinbird (~coinbird@ip-178-203-251-11.hsi10.unitymediagroup.de) |
| 23:35.42 | *** join/#asterisk Micc_ (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 23:42.01 | *** part/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 23:51.40 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |