| 00:03.21 | *** join/#asterisk karelk (~karel@31.10.145.202) |
| 00:23.04 | *** join/#asterisk karelk (~karel@31.10.145.202) |
| 00:35.55 | [TK]D-Fender | wonders where the last hour went |
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| 01:16.49 | laptopdude90 | lvlinux, you on right now? |
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| 01:26.10 | lvlinux | laptopdude90: semi |
| 01:26.29 | laptopdude90 | Okay, so I'm thinking about upgrading my WiFi network this black friday |
| 01:26.38 | laptopdude90 | I've commonly heard Uqiquiti is the go-to people |
| 01:27.03 | laptopdude90 | I live in a pretty huge house btw |
| 01:27.16 | lvlinux | Yes ubiquity or Cisco |
| 01:27.20 | laptopdude90 | http://www.ncix.com/detail/ubiquiti-unifi-ap-802-11n-300mbps-c2-96004.htm |
| 01:27.26 | laptopdude90 | This is what I was looking at |
| 01:27.34 | [TK]D-Fender | Mikrotik <3 |
| 01:27.50 | *** join/#asterisk karelk (~karel@31.10.145.202) |
| 01:28.29 | lvlinux | Not familiar w mikrotik. Why u like them? |
| 01:29.10 | laptopdude90 | Oh, should I keep the default UniFi AP firmware or install openwrt |
| 01:29.21 | laptopdude90 | Also, what makes Ubiquiti so good? |
| 01:29.24 | [TK]D-Fender | They work well, are agressively priced and have a huge selection of products |
| 01:31.04 | lvlinux | Keep unifi |
| 01:31.17 | laptopdude90 | kk, why? |
| 01:31.25 | [TK]D-Fender | <PROTECTED> |
| 01:31.26 | [TK]D-Fender | SAD |
| 01:31.42 | laptopdude90 | lol I've got a 24dbi antenna nearby |
| 01:32.07 | [TK]D-Fender | Yes well taht 3pak doesn't say "external antenna input" from what I can tell |
| 01:32.19 | laptopdude90 | Yeah yeah I know :P |
| 01:32.25 | [TK]D-Fender | No kinda kill like over-kill... |
| 01:32.31 | laptopdude90 | Lol can you imagine a massive 24 dbi antenna hanging from the ceiling |
| 01:32.45 | [TK]D-Fender | Which is why I like Mikrotik. They have LOTS of overkill products. |
| 01:32.47 | lvlinux | I just installed one |
| 01:33.01 | laptopdude90 | How much can I expect to spend for this upgrade? |
| 01:33.12 | *** join/#asterisk karelk (~karel@31.10.145.202) |
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| 01:34.40 | laptopdude90 | Can you link me some good mikrotik gear? |
| 01:34.46 | lvlinux | If u have big hs u probly want 2 ap |
| 01:35.04 | laptopdude90 | Okay, maybe I can buy the 3 pack and split it w/ my dad |
| 01:35.40 | [TK]D-Fender | 1000mw + 15db Ant centrally mounted. If that doesn't cover a whole home (or neighbourhood) ...... |
| 01:35.41 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 01:36.06 | laptopdude90 | 15dbi antenna doesn't really do up or down tho |
| 01:36.19 | lvlinux | That's the prob u don't want cover ur whole neighborhood |
| 01:36.28 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 01:36.30 | laptopdude90 | My house is surrounded by trees |
| 01:36.33 | [TK]D-Fender | That's just a bonus :) |
| 01:36.37 | *** join/#asterisk DanQuinney (sid18169@gateway/web/irccloud.com/x-lcwvnmqxmhyvtqjv) |
| 01:36.41 | laptopdude90 | Even my 24dbi antenna can barely penetrate it |
| 01:36.58 | *** join/#asterisk GameGamer43 (sid5533@gateway/web/irccloud.com/x-ietvtrkavuuowixv) |
| 01:37.25 | laptopdude90 | anyway |
| 01:37.32 | lvlinux | I never used ubiquity gear :-D |
| 01:37.33 | laptopdude90 | What mikrotik do you use |
| 01:37.39 | laptopdude90 | then what do you use |
| 01:37.45 | [TK]D-Fender | Current RB2011 series |
| 01:37.47 | *** join/#asterisk Fanch (~fanch@reverse1.qth.fr) |
| 01:37.49 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 01:37.49 | *** mode/#asterisk [+o file] by ChanServ |
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| 01:38.07 | laptopdude90 | ew |
| 01:38.09 | [TK]D-Fender | And am looking at replacing my office AP's with some from MT |
| 01:38.09 | lvlinux | And 1000mw isn't legal |
| 01:38.11 | laptopdude90 | ugly af |
| 01:38.15 | laptopdude90 | It is in Canada |
| 01:38.20 | laptopdude90 | w/ a 6dbi antenna |
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| 01:38.36 | laptopdude90 | for every +3dbi antenna -1 db TX powe |
| 01:38.41 | [TK]D-Fender | What I'm using isn't JUST an AP |
| 01:38.47 | [TK]D-Fender | so UAF doesn't really apply |
| 01:38.51 | laptopdude90 | yeah |
| 01:38.56 | laptopdude90 | But I want it to be just an AP |
| 01:38.56 | [TK]D-Fender | They have other products |
| 01:39.00 | laptopdude90 | so I can put it wherever |
| 01:39.03 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 01:39.04 | [TK]D-Fender | Depends what you want and how you want to use it |
| 01:39.09 | laptopdude90 | their wifi'only products aren't that good |
| 01:39.15 | laptopdude90 | I want to mount it on my ceiling |
| 01:39.22 | *** join/#asterisk cbdev (~fnord@hieristdas.internetzuen.de) |
| 01:39.25 | laptopdude90 | which means it has to look good or my dad won't let me |
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| 01:41.56 | [TK]D-Fender | That's one thing Ubiquiti has.... |
| 01:42.02 | [TK]D-Fender | fire-alarm style models |
| 01:42.09 | laptopdude90 | yeah |
| 01:42.41 | [TK]D-Fender | I'm shopping for 10GB equipment though whichis one thing MT is getting better at. |
| 01:42.52 | [TK]D-Fender | Unfortunately only SFP+ so far.... |
| 01:42.53 | laptopdude90 | <PROTECTED> |
| 01:43.01 | lvlinux | Sweet |
| 01:43.09 | *** join/#asterisk ryang (sid10904@gateway/web/irccloud.com/x-fjgcjdfdvzardfpm) |
| 01:43.11 | laptopdude90 | what does aggerate speed mean |
| 01:43.32 | [TK]D-Fender | The same those 2 words mean from the dictionary ..... together |
| 01:43.42 | lvlinux | Yup |
| 01:43.57 | [TK]D-Fender | http://routerboard.com/RBcAP2n |
| 01:44.05 | [TK]D-Fender | My bad, they DO have one fire-alarm style AP |
| 01:44.22 | laptopdude90 | Okay, so 300mbps aggregate means 2 clients 150mbps each for example |
| 01:44.33 | laptopdude90 | >2dbi antenna |
| 01:44.35 | laptopdude90 | lol |
| 01:45.08 | laptopdude90 | how on earth do I run ethernet through the ceiling? |
| 01:45.16 | [TK]D-Fender | 2db yeah.. a little sadder than ubnt's |
| 01:45.35 | laptopdude90 | Well the lower the dbi, the more omnidirectional it is |
| 01:45.45 | lvlinux | Depends on ur ceiling! |
| 01:45.51 | laptopdude90 | truw |
| 01:48.59 | *** join/#asterisk karelk (~karel@31.10.145.202) |
| 01:50.06 | lvlinux | I just setup a church office system where the office is two houses from the church where the phone line comes in. So put up a ubiquiti power beam 18db in basement and 13db nanobeam on church and beam Internet and VoIP between. |
| 01:53.37 | lvlinux | They don't know they have VoIP, they just know they got dial tone lol. |
| 01:56.00 | laptopdude90 | So is ubiquiti good? |
| 01:56.09 | laptopdude90 | Do they have like 10 packs? |
| 02:00.30 | [TK]D-Fender | Ubiquiti is just fine. They have a rep |
| 02:00.38 | laptopdude90 | kek.exe |
| 02:00.52 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-ackudxvzxisffcvl) |
| 02:00.55 | [TK]D-Fender | check the specs for the product you're specifically looking at, check any reviews, etc |
| 02:01.55 | laptopdude90 | Uqibuiti does paid hotspots? |
| 02:02.01 | laptopdude90 | I totally need this |
| 02:03.44 | laptopdude90 | So do the APs connect together |
| 02:03.49 | laptopdude90 | Or do they connect to a central server |
| 02:11.05 | laptopdude90 | Ewww |
| 02:11.08 | laptopdude90 | It's only 10/100 |
| 02:11.12 | laptopdude90 | that's not gonna cut it |
| 02:16.49 | *** join/#asterisk livtyler (~livtyler@unaffiliated/livtyler) |
| 02:17.17 | livtyler | how do I include TTL value in Page application? |
| 02:22.22 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 02:23.08 | lvlinux | thinks it's funny that consumer people put such stock into labeled speeds... |
| 02:23.24 | laptopdude90 | Well my internet connection is 150/50 |
| 02:23.29 | laptopdude90 | So 10/100 isn't fast enough |
| 02:23.41 | laptopdude90 | Not to mention |
| 02:23.46 | laptopdude90 | What's the point of a 300mbps router |
| 02:23.56 | laptopdude90 | if it only has a 100mbps connection to the other stuff |
| 02:28.34 | livtyler | laptopdude90: bw growth in time? maybe this year it could have 100 but next year customer can have 200 or 300 |
| 02:29.01 | laptopdude90 | How exactly do I upgrade a 100mbps port |
| 02:29.37 | livtyler | laptopdude90: what environment? MPLS? VDSL? |
| 02:29.48 | laptopdude90 | dude |
| 02:29.51 | laptopdude90 | It's a 100mbps port |
| 02:29.57 | laptopdude90 | Protocol doesn't matter |
| 02:30.33 | livtyler | what side? provider? |
| 02:31.27 | laptopdude90 | No |
| 02:31.28 | laptopdude90 | The port |
| 02:31.30 | laptopdude90 | On the router |
| 02:31.32 | laptopdude90 | Is 100mbps |
| 02:31.41 | laptopdude90 | But |
| 02:31.43 | laptopdude90 | The same router |
| 02:31.44 | laptopdude90 | has |
| 02:31.44 | laptopdude90 | a |
| 02:31.45 | laptopdude90 | 300mbps |
| 02:31.51 | laptopdude90 | wifi chip |
| 02:35.35 | livtyler | wifi chip? you mean MU-MIMO backplane? |
| 02:35.47 | laptopdude90 | k you know how routers have these thing |
| 02:36.01 | laptopdude90 | that use electromagnetic radiation to broadcast packets |
| 02:36.04 | laptopdude90 | THAT TING |
| 02:37.05 | livtyler | broadcast? mmmm, sorry I work in networking electronics design, not sure what you're talking about |
| 02:37.28 | laptopdude90 | dude you're bad at trolling |
| 02:38.07 | livtyler | dude you don't know how networking works |
| 02:38.50 | livtyler | wifi chip lol |
| 02:39.07 | laptopdude90 | that's what's inside the router... |
| 02:39.16 | laptopdude90 | have you even opened a router up before? |
| 02:40.19 | livtyler | lol |
| 02:40.49 | laptopdude90 | no? |
| 02:40.52 | laptopdude90 | son, how old are you? |
| 02:41.02 | livtyler | ok, you're right, wifi chip, I won't argue with you |
| 02:41.12 | laptopdude90 | thank you. |
| 02:41.14 | livtyler | btw, I'm a girl |
| 02:42.51 | laptopdude90 | im a grill btw |
| 02:43.06 | livtyler | no, you are a laptopdude lol |
| 02:43.19 | livtyler | come on |
| 02:43.54 | livtyler | you made my day grandpa "wifi chip" XD |
| 02:44.10 | laptopdude90 | what do you call it |
| 02:44.20 | laptopdude90 | http://cdn.slashgear.com/wp-content/uploads/2010/01/wi2wi_wifi_meh_chip.jpg |
| 02:44.31 | laptopdude90 | actually tho are you 12 |
| 03:07.01 | *** join/#asterisk krishna1 (~krishna1@cpe-76-95-249-156.socal.res.rr.com) |
| 03:07.06 | krishna1 | Hi |
| 03:24.49 | igcewieling1 | Note that in past years the RFC Editor has sometimes published serious documents with April 1 dates. Readers who cannot distinguish satire by reading the text may have a future in marketing." |
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| 04:01.14 | *** join/#asterisk ruben23 (~OpenDIAL@203.111.224.69) |
| 04:02.33 | ruben23 | hi guys any link on how to setup fail2ban with asterisk 11, it seems it has new rules which called security..any guide please.. |
| 04:03.41 | *** join/#asterisk krishna1 (~krishna1@cpe-76-95-249-156.socal.res.rr.com) |
| 04:09.25 | ruben23 | guys anyone tried fail2ban with asterisk 11 |
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| 04:34.58 | [TK]D-Fender | Lots of people |
| 04:35.01 | [TK]D-Fender | Lots of guides |
| 04:35.15 | [TK]D-Fender | Takes about ... a few SECONDS to find them |
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| 05:17.32 | Schmee | Hi all. I'm trying to set up a confbridge instance in asterisk 11, but I've run into a small snag. The person running the conference want to dial out to get people to join, which is fine on it's own, but occasionally, she gets a recipient whose phone goes straight to voicemail. Is there any way to kick off the last joined member, or would I be better off changing the dialplan to filter those calls first? |
| 05:22.18 | [TK]D-Fender | There is ... and "better off"... probably |
| 05:27.22 | Schmee | If there's a better way, I'd like to learn it. There isn't a lot of examples kicking around |
| 05:28.44 | [TK]D-Fender | prompt them for confirmation before dumping them in confbridge |
| 05:29.00 | [TK]D-Fender | Or us AMD() |
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| 05:30.50 | Schmee | AMD() sounds like a possible solution. At least it's another avenue I can check. Thanks for the pointer [TK]D-Fender |
| 05:31.02 | [TK]D-Fender | you're welcome |
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| 06:25.26 | lvlinux | livtyler: what exactly do you mean by Paging TTL? |
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| 09:03.47 | hetii | Hi |
| 09:04.15 | hetii | <PROTECTED> |
| 09:04.31 | hetii | If I want to build my configuration from scrath ? |
| 09:05.57 | wdoekes | A: yes, you'll need at least a few: modules.conf and asterisk.conf. but you may strip them down as far as you can |
| 09:06.29 | hetii | ok thx for that info |
| 09:07.09 | wdoekes | for extra clean config, set autoload=no in modules.conf |
| 09:07.32 | wdoekes | you'll need to figure out which modules you'll need though, which can be quite a few |
| 09:21.24 | *** join/#asterisk whizzi (~whizzi@185.21.52.255) |
| 09:22.22 | whizzi | good day all. Quick question, what can cause Asterisk 11.17 to lose itâs Registration with a trunk while itâs peer is still online? |
| 09:22.39 | whizzi | and 'sip reload' doesnât send out a new REGISTER to the trunk |
| 09:22.54 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 09:23.19 | whizzi | the only way to get the trunks to register again, would be to actually stop / start asterisk |
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| 09:34.16 | hetii | wdoekes: propably I will just use SIPs trunk and endpoints so should not be much |
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| 09:47.12 | *** join/#asterisk bounceman (~bounceman@185.32.9.250) |
| 09:47.55 | bounceman | Running Asterisk as a Media gateway, is the number of concurrent calls and the transcoding all hardware dependent or is there a point were asterisk just starts to crumble? |
| 09:48.07 | bounceman | Meaning can we just increase and increase the hardware of a server and expect Asterisk to be doing good? |
| 09:54.24 | *** join/#asterisk sekil (~sekil@109-92-28-226.static.isp.telekom.rs) |
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| 10:03.06 | wdoekes | hetii: haha, you'd be surprised |
| 10:04.16 | wdoekes | hetii: res_rtp_asterisk, res_timing_timerfd, codec_alaw, format_sln. and then you'd need a channel driver (chan_sip? chan_pjsip?) which may have lots of dependencies |
| 10:06.40 | hetii | hmm odd |
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| 11:13.06 | *** join/#asterisk naamane (51c0bc91@gateway/web/freenode/ip.81.192.188.145) |
| 11:13.21 | naamane | hello everyone |
| 11:15.28 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 11:15.33 | afournier | hello |
| 11:16.11 | afournier | when I set a custom device_state, and i read it one priority under, he shows the old value, after hungup it's set for real, is it normal ? |
| 11:16.13 | naamane | hello afournier |
| 11:16.25 | afournier | s/under/later/ |
| 11:16.46 | afournier | thank you infobot :) |
| 11:16.46 | infobot | de rien, afournier |
| 11:16.49 | afournier | haha |
| 11:17.29 | naamane | can anyone help me here? :) |
| 11:18.26 | afournier | i though Set(DEVICE_STATE(Custom:X)=Y) was synchronous :( |
| 11:34.09 | afournier | nanoha-sama: what's your problem ? |
| 11:34.20 | afournier | naamane left :( |
| 11:43.44 | *** join/#asterisk stevenm (~stevenm@195.62.218.30) |
| 11:44.05 | stevenm | Lo, if astdb is missing - but I don't care what contents it had - how can I get asterisk to make another new one? |
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| 12:04.38 | *** join/#asterisk aberrios (~antonio@185.99.164.76) |
| 12:05.32 | aberrios | anyone sending asterisk logs straight to a remote syslog server? |
| 12:06.03 | aberrios | trying to find syntax documentation,, so far only found details of sending to local syslog process which then forwards on |
| 12:07.16 | Chainsaw | aberrios: That's how I do it. Syslog is better at pumping a lot of logs to specific places than Asterisk is. |
| 12:09.16 | aberrios | problem I see is if I'm running asterisk in docker, I *should* run one process per container, in order to get this going in docker I'd have to use supervisord to run both processes |
| 12:09.58 | aberrios | on our vanilla VMs that is how I do it atm, Asterisk > rsyslog > graylog |
| 12:10.19 | aberrios | asterisk(local) > rsyslog(local) > graylog(remote) |
| 12:11.02 | *** join/#asterisk Fanch (~fanch@reverse1.qth.fr) |
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| 12:17.00 | Chainsaw | aberrios: Yes, rsyslog or syslog-ng. Either would fit the bill. |
| 12:18.10 | aberrios | Chainsaw: yeah but I'd rather not have to run rsyslog or any other process on the same container just for asterisk to send its logs to so it can forward them on. I was wondering if logger.conf could be set up just to pipe logs to a UDP port |
| 12:19.37 | Chainsaw | aberrios: It's not a core function for Asterisk, so I wouldn't trust it to be implemented particularly robustly. At least rsyslog or syslog-ng would buffer if your graylog hickups. |
| 12:19.41 | stefan27 | When dialing a sip friend, i can use SIPFROMDOMAIN=A and CALLERID(num)=B to make asterisk write <A@B:20060> in the INVITE's from header but I cannot manipulate the port 20060? It's always set to whatever sip.conf has as bindport? |
| 12:19.51 | stefan27 | B@A actually |
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| 12:20.56 | aberrios | Chainsaw: I also wondered who else is using asterisk within docker and how they're dealing with logs.. I know lmadsen was playing with it. I guess if you can get asterisk to send straight to a remote logging machine I'd have to setup supervisord in the container to run asterisk+rsyslog |
| 12:22.08 | aberrios | can=can't |
| 12:27.53 | aberrios | ah looks like I can get docker to use a specific logging driver, asterisk can still send to a syslog facility locally and then docker deals with it |
| 12:27.57 | aberrios | yays |
| 12:28.15 | Chainsaw | aberrios: Excellent. That's equally good and still cleaner for you :) |
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| 13:22.08 | wonderworld | hi, my asterisk doesn't write logs. i found out that logger.conf was missing in /etc/asterisk. i moved the dist version of the file to /etc/asterisk and restarted asterisk. still no logs. is there anything else i need to do? thanks a lot. |
| 13:23.44 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:23.45 | wonderworld | sorry, actually it writes logs (messages, queue_log) but no CDRs |
| 13:24.14 | whizzi | check out cdr.conf ? |
| 13:24.32 | whizzi | or 'cdr show status' in Asterisk |
| 13:26.09 | wonderworld | ohh thanks, thought cdr's were configured in logger.conf as well |
| 13:26.12 | wonderworld | thats it |
| 13:26.36 | whizzi | :) |
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| 13:58.41 | stevenm | hey we've got a polycom phone that pulls its config over https from a third party provisioning server - but the phone can be obtained from anywhere in the world, the 3rd party just needs to be pre-warned what the MAC is |
| 13:59.03 | stevenm | presumably - am i right in thinking this? - i could simulate these requests from my desktop and get the details in plain text form? |
| 13:59.32 | *** join/#asterisk bluemerlin (~bluemerli@office-nat.westpoint.gradwell.net) |
| 13:59.36 | stevenm | like, use a polycom-looking ssl cert, contact their server using the right polycom user-agent - supply the MAC...and bang?! |
| 14:01.34 | Gugge | im pretty sure polycom phones support encrypted configs that only the phone can decrypt |
| 14:01.49 | Gugge | If the provisioning server uses that, you cant just get the plain text config |
| 14:02.34 | stevenm | Gugge, but i'm getting the phones from anywhere I like (any supplier) and putting them on their provisioning system... (which works) - then surely that key for decoding them - is the same for any polycom? a manufacturers thing. |
| 14:02.51 | Gugge | it could be a unique key pr phone |
| 14:03.17 | stevenm | but I don't need to supply that key (if one exists) to the 3rd party to get the phone accepted by their provisioning system |
| 14:03.20 | stevenm | only need to supply the mac |
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| 14:04.01 | Gugge | it _could_ be a unique private/public key pr phone, known by polycom ... providing the public key to the encryption software when given the mac? :) |
| 14:04.21 | stevenm | ooh like a salt |
| 14:04.28 | Gugge | i dont know if it works that way, im just guessing :) |
| 14:04.31 | stevenm | yeah |
| 14:05.04 | stevenm | irritates me this 3rd party hosted pbx only lets certain makes/models of phones connect (they check user agent) |
| 14:05.10 | Gugge | but ive seen provisioning servers giving plain text configs before :) |
| 14:05.21 | stevenm | so i've stuck asterisk in the middle to change the user agent... just need to double check my connection details |
| 14:05.23 | Gugge | you can fake the user-agent :P |
| 14:05.32 | Gugge | :) |
| 14:05.49 | stevenm | i wish more phones would let you fake it on the phone itself though |
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| 14:15.49 | whizzi | stevenm: my company builds provisioning for phones |
| 14:16.12 | stevenm | we used to for snom and aastra - never done polycom really before |
| 14:16.20 | whizzi | polycoms just receive a plain text xml-ish data |
| 14:16.24 | stevenm | snom and aastra are unfortunately makes/models banned from this silly hosted pbx |
| 14:16.51 | [TK]D-Fender | I have BUTTERFLIES : https://xkcd.com/378/ |
| 14:17.50 | whizzi | stevenm: Aastraâs (or Mitel as they are called these days) are quite nice for provisioning. You can let them do everything as longs as you know the correct SIP NOTIFY |
| 14:18.22 | whizzi | Panasonics are weird, everything is closed by default and they have actually plain text config files |
| 14:21.02 | whizzi | stevenm: Basically, most phones just read a HTTP-source where itâs information is stored |
| 14:21.11 | stevenm | yeah I get that |
| 14:21.27 | whizzi | Mitels used to have some sort of decoding, but since a few firmwares ago they stopped doing that |
| 14:21.59 | whizzi | for polycom, we sent this header: Content-type: application/octect-stream |
| 14:22.05 | whizzi | and then XML-ish data |
| 14:22.30 | whizzi | forget about the -ish, itâs XML |
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| 15:12.26 | prelude2004c | hey everyone good day.. anyone know why while i am usin speex.. i make a call and i hear them but a few seconds later the system crashes asterisk but no logs at all to tell me why |
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| 15:16.47 | [TK]D-Fender | prelude2004c: Nope. You'd need an actual dump for anyone to have a clue |
| 15:17.02 | [TK]D-Fender | No dump = no evidence |
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| 15:34.08 | prelude2004c | hey anyone here use speex code before ? |
| 15:39.11 | [TK]D-Fender | At least get specific about what versions you are using, etc... |
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| 15:45.31 | Ickmund | Realtime on 13.6 not working for me. On 1st register 401 goes out, but doesn't seem like the response is registered. Any pointers? cli: http://pastebin.com/qD6ZzvV1, ngrep: http://pastebin.com/fnJRjzbA, config: http://pastebin.com/WDa9qpyG |
| 15:47.18 | *** join/#asterisk drab (~administr@23-24-198-190-static.hfc.comcastbusiness.net) |
| 15:48.50 | Ickmund | Any attempts at registering after that first one times out is met with dead silence until * is restarted |
| 15:48.50 | drab | hi, I'm trying to play a message "invalid number etc" when people try to dial an extension that doesn't exist. I tried to use i in the context, but it doesnt work. google suggests that only works when Background() has been used and people seem to be creating a [bogus] context with a catchall match to include at the end of the main context. See for example http://www.planetwayne.com/forums/viewtopic.php?t=218 |
| 15:49.04 | drab | is that a good way to go about it or is there a better way to handle this? |
| 15:49.56 | WIMPy | Well, yes. chan_sip will reject calls if there's no match, so it will never go to the i extension. |
| 15:49.59 | [TK]D-Fender | SIP calls never touch "i" |
| 15:50.09 | [TK]D-Fender | You'll need a catch-all at the end of your matching process for this |
| 15:51.01 | [TK]D-Fender | SIP gives a 404 otherwise |
| 15:52.20 | drab | ok, thank you. are the any side effects gotchas when implementing such a catch-all context |
| 15:52.24 | drab | ? |
| 15:53.02 | drab | like that guy in that thread seems to say you have to define a h extension or stuff will be called twice |
| 15:53.10 | drab | not quite understanding, but wondering if there's more of that |
| 15:54.53 | WIMPy | That depends on the pattern you use for your catch-all. If you just use _. it will match the special extensions as well which is not a good thing. |
| 15:55.48 | [TK]D-Fender | Should take a few patterns to cover this |
| 15:56.10 | WIMPy | The other side effect is in situation where you dial interactively invalid numbers won't detected any more and you have to wait for a timeout to get to your catch-all. |
| 15:56.16 | [TK]D-Fender | unless you have matches for all the special ones |
| 15:58.50 | drab | ok, thanks, that gives me something to work with. Will work on a better pattern (was indeed using _. from the example thread) |
| 15:59.08 | [TK]D-Fender | I'd just stick to the standard patterns and do it in 3. |
| 16:03.19 | drab | I lost you there. what are the standard patterns and 3 as in priority 3? of what?. thanks for bearing with me |
| 16:05.14 | [TK]D-Fender | 3 patterns |
| 16:05.25 | drab | oh, I see |
| 16:05.32 | [TK]D-Fender | to cover everything that is a dialed "thing" from a numal phone |
| 16:05.44 | [TK]D-Fender | normal* |
| 16:16.55 | drab | just to be sure, are the includes => statements checked in series or loaded all at once and then the resulting flat context is matched from the most specific to the least ? |
| 16:17.40 | drab | I understand that teh current context is checked before any include, but not sure what then happens if there's more than one include and the extension is not found in the original context |
| 16:18.04 | [TK]D-Fender | They are search in the order you included them |
| 16:18.09 | WIMPy | Includes ar hecked in the order listed and only if there was no match in the context itself. |
| 16:18.26 | drab | thanks for confirming that |
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| 16:19.55 | [TK]D-Fender | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics-SECT-3.html#asterisk-DP-Basics-SECT-3.8 |
| 16:28.56 | afournier | When I set a custom device state "Set(DEVICE_STATE(Custom:X)=BUSY)", and read it one priority later "NoOp(${DEVICE_STATE(Custom:X)})", asterisk shows the old value of the device state, unless Wait(0.1) is executed before NoOp(), is it normal ? |
| 16:41.46 | prelude2004c | hey guys.. very very sorry about the delay. basically right now i am running the latest 13.6.1 version with the latest speex codec version 1.2rc1 .. basically i am trying to lower bw as much as possible but the asterisk show translations show me that i can only get 23kbs from ulaw to speex.. i thought speex can go as low as 2kbs ? how do i get there ? |
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| 16:45.33 | igcewieling1 | prelude2004c: there is a fixed overhead per packet regardless of what you are using. IIRC, the overhead is around 15kbps |
| 16:46.17 | igcewieling1 | as you can see it is *impossible* to have an RTP stream at 2kbps. |
| 16:47.01 | WIMPy | Well, you can increase packet size. But that also adds delay. |
| 16:47.12 | prelude2004c | ic..... what is the lowest usage we can have at reasonable voip quality ? doesn't have to be great.. just don't want it to be cutting out |
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| 16:49.38 | WIMPy | Well, as igcewieling1 already said, at some point the payload will only be a fratcion of the overhead, so the gain in bandwith will be much smaller than the decrease in quality. |
| 16:50.04 | WIMPy | That's where IAX comes handy if you have multiple calls between two peers. |
| 16:50.12 | [TK]D-Fender | UDP/RTP overhead = 20kbps @ 20ms |
| 16:55.15 | igcewieling1 | IAX trunking uses more bandwidth until you have more than one call active. |
| 16:56.55 | WIMPy | Does it? Can't be much. |
| 16:58.30 | [TK]D-Fender | Cry for the bytes |
| 16:59.40 | WIMPy | I never noticed, but it certainly scales a lot better. |
| 17:19.34 | igcewieling1 | you still have the overhead of trunking when there is only one call. |
| 17:20.20 | WIMPy | But you don't have RTP. So the result for one call should be pretty similar. |
| 17:21.44 | igcewieling1 | *shrug* I'll never use IAX again so it doens't matter much to me. |
| 17:28.07 | Ickmund | Is there a way with pjsip to not challenge registers? Like with a proxy/registrar in front, or is this kind of setup not even needed nowadays? |
| 17:35.11 | WIMPy | It does not look like a single call on an IAX trunk would take more bw than a RTP stream. |
| 17:37.20 | Ickmund | Jordan has a slide from last years Kamailio World that seems to suggest to noload res_pjsip_authenticator_digest and _registrar, but then I get a 501 Not Implemented |
| 17:39.05 | [TK]D-Fender | WIMPy: I think the comment was that with trunk mode enabled that might add a few BYTES to the header vs not using trunk mode so that might make a single stream meaninglessly bigger than not using trunk mode |
| 17:39.25 | WIMPy | That's true AFAIK. |
| 17:39.55 | WIMPy | But I don't see any disadvantage over RTP. |
| 17:40.14 | prelude2004c | hey . is g729 8Kbits + overhead of 15kbs ? |
| 17:40.36 | igcewieling1 | [TK]D-Fender: exactly. for normal people it doesn't matter, but might matter to those crazy people who think they need to save every byte of bandwidth possible |
| 17:40.40 | WIMPy | Can be. |
| 17:41.00 | WIMPy | There are many versions og G.729 and the packet size can vary as well. |
| 17:41.20 | igcewieling1 | prelude2004c: does it say g729 uses 8Kbps? |
| 17:41.30 | prelude2004c | but can't be 8kbs only right ? maybe i am going about this the wrong way.. i am trying to do this over 3g.. best quality seeing as jitter and package loss are a problem |
| 17:41.36 | prelude2004c | also low priority |
| 17:41.49 | igcewieling1 | prelude2004c: where are you connecting to? an ITSP? |
| 17:42.02 | prelude2004c | some 3g tower for my cellular company |
| 17:42.13 | prelude2004c | using ulaw gives me constant cut outs and stuff |
| 17:42.16 | igcewieling1 | prelude2004c: when I ran SIP on 3G I used my purchased G729 license with little issue. |
| 17:42.22 | WIMPy | >>... at some point the payload will only be a fratcion of the overhead, so the gain in bandwith will be much smaller than the decrease in quality. |
| 17:42.33 | prelude2004c | but can't speex do the smae thing as g729 ? |
| 17:42.43 | WIMPy | sure |
| 17:42.46 | igcewieling1 | prelude2004c: does the other side of the call support speex? |
| 17:43.04 | prelude2004c | um.. i have ot connected to a server and then the server is ulaw to the PSTN lines |
| 17:43.10 | prelude2004c | but yes the other users will also be speex |
| 17:43.15 | igcewieling1 | prelude2004c: what server? |
| 17:43.16 | prelude2004c | if sip > sip |
| 17:43.21 | prelude2004c | asterisk |
| 17:44.43 | igcewieling1 | prelude2004c: wouldn't it be easier to get it working with the gsm codec first? Then figure out why a part of Asterisk which relys on 3rd party external libraries? |
| 17:44.53 | igcewieling1 | ..doesn't work. |
| 17:46.14 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-yifgdwclrtxbyyfd) |
| 17:50.07 | prelude2004c | gsm ? never used that codec before |
| 17:50.19 | prelude2004c | is gsm better than speex ? |
| 17:51.43 | WIMPy | What kind of "better"? |
| 17:52.16 | WIMPy | But the Asterisk supported GSM CODEC is not that great. |
| 17:52.58 | lvlinux | meh GSM is at the bottom of my codec list. |
| 17:52.59 | igcewieling1 | prelude2004c: GSM is EASIER. Once you get it working, then you should concentrate on making it BETTER |
| 17:54.00 | prelude2004c | i have speex working |
| 17:54.05 | prelude2004c | using up 23kbs average |
| 17:54.23 | igcewieling1 | prelude2004c: sounds like a speex config issue. |
| 17:54.30 | prelude2004c | which you guys say is the overhead that makes it get there.. i thought it was from 2kbs - 44kbs |
| 17:55.04 | WIMPy | 23kbps speex or 23kbps RTP? |
| 17:55.09 | igcewieling1 | what problem remains? |
| 17:55.30 | prelude2004c | rtp i guess.. data.. whatever .. 23kbs of total data |
| 17:55.39 | igcewieling1 | prelude2004c: BTW, IAX trunking only works between Asterisk boxes. |
| 17:55.56 | prelude2004c | i am using SIP |
| 17:56.12 | WIMPy | AFAIK Freeswitch and Yate also do IAX, don't they? |
| 17:56.26 | igcewieling1 | WIMPy: They don't count. 8-) |
| 17:57.05 | igcewieling1 | Though I was thinking of hardware (phones, ATAs, etc) and softphones |
| 17:57.12 | prelude2004c | here is another off topic.. can a user be set to use the reinvite to a specific media server ? |
| 17:57.27 | igcewieling1 | prelude2004c: is your SIP device behind NAT? |
| 17:57.40 | WIMPy | Yes, that's a pretty non-existing area. |
| 17:57.47 | prelude2004c | like geo location or something.. where i can update so when a call is made it can be made to the latest media server near them but the control for auth can be done in a remote location |
| 17:57.48 | igcewieling1 | nevermind. You cannot do re-invites when there is nat involved. |
| 17:58.35 | igcewieling1 | prelude2004c: more and more it sounds like you need Kamailio, not Asterisk. |
| 17:58.47 | prelude2004c | no ? you eman if i have my home router it can't reinvite to talk to the user ? |
| 17:58.51 | igcewieling1 | Asterisk is not a SIP Proxy |
| 17:59.10 | igcewieling1 | prelude2004c: no. I mean if there is NAT involved then you can't re-invite |
| 18:02.32 | prelude2004c | well if reinvite fails wont it just go without back to the source server |
| 18:02.40 | prelude2004c | never heard of kamailio |
| 18:02.59 | [TK]D-Fender | no, if it fails you've get no audio and it will simply fail |
| 18:03.03 | igcewieling1 | prelude2004c: usualy the call simply drops |
| 18:03.09 | *** join/#asterisk stinger (~stinger@unaffiliated/stinger) |
| 18:03.44 | WIMPy | Failure protection only exists in H.323 an IAX. |
| 18:03.46 | prelude2004c | ya makes sense... so the idea is to never use reinvite right ? |
| 18:03.46 | igcewieling1 | prelude2004c: kamailio is a SIP proxy. It allows you to do lower level stuff in SIP. Stuff Asterisk will never be able to do since it isn't a SIP proxy. |
| 18:03.58 | prelude2004c | so using iax is a good idea / |
| 18:04.00 | prelude2004c | ? |
| 18:04.13 | igcewieling1 | prelude2004c: Do both ends support IAX? |
| 18:04.17 | prelude2004c | yup |
| 18:04.20 | WIMPy | I think anything is a better idea than SIP. |
| 18:04.33 | prelude2004c | so speex over IAX possible or some low usage codec |
| 18:04.36 | prelude2004c | same thing right ? |
| 18:05.01 | [TK]D-Fender | IAX offers nothing for this |
| 18:05.04 | igcewieling1 | prelude2004c: you'll find virtually no assistance in working with IAX. So few people use it the chances on one being on the channel when you are..... |
| 18:05.10 | prelude2004c | but really the only thing that makes a lot of sense for me is .. if i move to iax . can i set media servers for each customer differnet based on location ? |
| 18:05.17 | [TK]D-Fender | NO |
| 18:05.20 | igcewieling1 | prelude2004c: no.m |
| 18:05.30 | prelude2004c | shoot... |
| 18:05.43 | [TK]D-Fender | You have failed to understand the basics of what IAX2 is |
| 18:05.50 | [TK]D-Fender | ONE port. |
| 18:05.55 | [TK]D-Fender | Signalling AND media |
| 18:06.06 | [TK]D-Fender | There is no RTP |
| 18:06.17 | [TK]D-Fender | IAx is not a routed protocol. |
| 18:06.35 | prelude2004c | so in your guys expert openion what should i do.. forget codecs for a sec.... i have a asterisk auth server in point A ... and now i deploy media servers into B and C . customer is roaming close to B or C .. and calls someone else also close to B or C .. they are talking to eeachother.. id on't want them both routing all the way back to A as it may be 200ms apart |
| 18:06.35 | igcewieling1 | IAX2 was an attempt to overcome the massive shortcomings of SIP and RTP. The problem is nobody outside of the Asterisk community cared. SIP won the VoIP protocol wars. |
| 18:07.10 | prelude2004c | yup single port i get that |
| 18:07.25 | igcewieling1 | prelude2004c: once you say "direct to a specific media server" then it becomes impossible with Asterik. |
| 18:08.11 | WIMPy | Why don't you make them talk to B or C then? Ir did I miss something? |
| 18:08.18 | WIMPy | Or |
| 18:08.46 | igcewieling1 | WIMPy: you are missing he is a Gentoo user (or at least appears to be) 8-| |
| 18:09.10 | WIMPy | What's wrong with that? |
| 18:09.23 | WIMPy | gets a feeling igcewieling1 doesn't like good things. |
| 18:09.39 | igcewieling1 | Lets redesign our network into something horribly complex so we get 0.1% latency or bandwidth savings. sounds like Gentoo to me. |
| 18:09.57 | prelude2004c | lol |
| 18:10.01 | igcewieling1 | WIMPy: I like stable working things which don't need babysitting. |
| 18:10.02 | prelude2004c | ubuntu here |
| 18:10.02 | [TK]D-Fender | http://fun.irq.dk/funroll-loops.org/ |
| 18:10.20 | WIMPy | Tell that too the likes of Google etc :-) |
| 18:10.55 | igcewieling1 | WIMPy: google throws money at the problem. |
| 18:10.58 | WIMPy | How does SIp fit in to stable? |
| 18:12.16 | pjensen00 | Man, I wish I had enough money to "throw at a problem" until it was fixed. Maybe I should become Google. That'd make my project easier. |
| 18:13.30 | WIMPy | Might be a lot cheaper to get a better internet connection. |
| 18:13.50 | pjensen00 | .... mmm.... k. |
| 18:14.01 | igcewieling1 | indeed. There is throwing money at the problem in an intellegent way. |
| 18:14.21 | pjensen00 | Like, turning dollar bills into paper airplanes so the money throwing goes further |
| 18:14.25 | WIMPy | Seriousely. Trying to use VOIP without a good and stable internet connection is not a clever idea. |
| 18:14.36 | pjensen00 | clever or good? |
| 18:14.58 | WIMPy | neither |
| 18:15.03 | lvlinux | not smart |
| 18:15.08 | pjensen00 | XOR |
| 18:15.41 | pjensen00 | I just want google fiber to make it to North Dakota. I'll be a skeleton before that happens. :( |
| 18:15.47 | igcewieling1 | pjensen00: I spent $10 on a g729 codec and never had trouble with audio issue son 3G again. I could have spent days getting it to workout a license, but I'd rather when $10 and avoid all that extra work. |
| 18:16.42 | pjensen00 | Ha yes. That's 10$ well spent |
| 18:20.07 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 18:31.36 | cmendes0101 | Currently I have a call going on with Dial. I also have Chanspy going on that channel and playing an audio file but this doesn't get picked up my mixmonitor that is on the original call. I was playing around with Bridge right now but that hangs up the main call when the bridge channel hangs up. Any ideas for alternatives? |
| 18:35.01 | pjensen00 | I'm assuming you're playing the audio message with 'barge' enabled? |
| 18:35.02 | pjensen00 | https://wiki.asterisk.org/wiki/display/AST/Application_ChanSpy |
| 18:35.55 | *** join/#asterisk TazzNZ (~TazzNZ@mail.insync.za.net) |
| 18:36.15 | cmendes0101 | Yah barge is being passed in |
| 18:38.25 | *** join/#asterisk eofster (~eofster@ip5b4063a6.dynamic.kabel-deutschland.de) |
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| 19:35.33 | pjensen00 | Hrm that's weird. I've not had that side effect in my experience |
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| 19:44.41 | igcewieling1 | Amazing how fast people who have been unreachable via phone, voicemail, fax, e-mail, etc respond when you shut off their phones. |
| 19:45.05 | pjensen00 | Agreed! *adjusts mustache* |
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| 21:19.00 | *** join/#asterisk hetii (~na@31-178-122-20.dynamic.chello.pl) |
| 21:19.04 | hetii | Hi |
| 21:19.52 | hetii | I have such simple configuration : http://pastebin.ca/3257185 and wonder why when I try call to 2001 or 1001 nothing happen |
| 21:20.14 | hetii | I mean my default extension is not executed |
| 21:20.53 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 21:22.32 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 21:22.41 | [TK]D-Fender | <--- Reliably Transmitting (no NAT) to 192.168.0.14:5060 ---> |
| 21:22.43 | [TK]D-Fender | SIP/2.0 403 Forbidden |
| 21:22.54 | [TK]D-Fender | Very clear auth failure |
| 21:23.24 | hetii | hmm |
| 21:23.24 | hetii | odd |
| 21:24.07 | hetii | <PROTECTED> |
| 21:24.07 | hetii | Name/username Host Dyn Forcerport Comedia ACL Port Status Description |
| 21:24.07 | hetii | sorry |
| 21:24.16 | hetii | The result is: 2000/2000 192.168.0.14 D Auto (No) No 5060 Unmonitored |
| 21:24.29 | hetii | AND also: Saved useragent "qutecom/rev-g-trunk" for peer 2001 |
| 21:24.34 | hetii | so looks like its registered |
| 21:24.46 | [TK]D-Fender | hcall is not authorized |
| 21:25.50 | hetii | don`t get it? Clien can be registered but not authorized to make a call ? |
| 21:27.01 | hetii | *client |
| 21:27.19 | [TK]D-Fender | maybe you got the auth right for the first part and not the second |
| 21:27.34 | [TK]D-Fender | then again you have lots of other stuff missing |
| 21:27.40 | [TK]D-Fender | no codecs defined, etc |
| 21:27.52 | [TK]D-Fender | Or the context |
| 21:28.09 | [TK]D-Fender | even touching one named [default] is a big mistake |
| 21:28.46 | WIMPy | wouldn't be surprised if it rejected calls for peers without context. |
| 21:29.16 | robmal | wouldn't be surprised if touching [default] was illegal in some countries |
| 21:30.04 | hetii | illegal ?:) |
| 21:30.07 | [TK]D-Fender | heads home... |
| 21:31.03 | robmal | hetii: There are some context which shouldn't be available. |
| 21:31.19 | robmal | Even with type=friend |
| 21:32.10 | hetii | by from law perspective or just by misconfiguration ?:) |
| 21:32.51 | robmal | Depends. |
| 21:33.15 | robmal | Both meet at some point. |
| 21:33.20 | robmal | Or extension. |
| 21:34.47 | hetii | ok, this instance is just for my internal testing stuff so don`t need to worry about legal aspects |
| 21:35.10 | robmal | That's what they all say. |
| 21:35.12 | robmal | ~book |
| 21:35.12 | 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 |
| 21:35.16 | robmal | Read some. |
| 21:35.30 | WIMPy | Like the nuclear boy scout? |
| 21:35.45 | hetii | but even when set context and codec still don`t see that any extension is triggered |
| 21:37.11 | robmal | Spend a moment, copy and paste the most complicated sip peer config you can find in the book, adopt it to your peers, try again. |
| 21:38.06 | *** join/#asterisk d00gster_ (~d00gster@unaffiliated/d00gster) |
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| 22:55.49 | *** join/#asterisk mcjoppy (~mcjoppy@li227-61.members.linode.com) |
| 22:55.53 | mcjoppy | Hello |
| 22:55.59 | *** join/#asterisk JonathanS (~JonathanD@freenode/staff/jonathand) |
| 22:56.14 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 22:56.19 | mcjoppy | Is there a nice config option to ensure IAX trunks reconnect after network issues? |
| 22:56.29 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
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| 23:07.38 | *** join/#asterisk Champi (Champi@damn.e-leet.be) |
| 23:08.36 | mcjoppy | Or, does SIP handle network issues better than IAX ? |
| 23:10.49 | *** join/#asterisk alex9999 (~alex999@host55-115-dynamic.5-87-r.retail.telecomitalia.it) |
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| 23:37.44 | *** part/#asterisk CyberPony (~CyberPony@74.254.113.163) |
| 23:51.38 | *** join/#asterisk laptopdude90 (laptopdude@HLFXNS016CW-142134091178.dhcp-dynamic.FibreOp.ns.bellaliant.net) |
| 23:51.51 | laptopdude90 | good news everyone |
| 23:52.25 | *** join/#asterisk aness (~aness@2a02:fe0:c310:e0:48:3667:31aa:2717) |
| 23:55.22 | laptopdude90 | I have been approved to cut holes in the wall to run ethernet :D |
| 23:57.25 | MaliutaLap | one does not "cut" holes for cable - one smashes them with a sledge hammer |
| 23:57.34 | MaliutaLap | far more satisfying |
| 23:57.52 | laptopdude90 | So how do I deal with the wood in the way? |
| 23:58.16 | laptopdude90 | http://www.julianhancock.com/bs2l.jpg |
| 23:58.32 | MaliutaLap | bigger sledge hammer, and fire ;) |
| 23:58.41 | laptopdude90 | >wooden house |
| 23:58.42 | laptopdude90 | >fire |
| 23:58.44 | laptopdude90 | seems legit |
| 23:58.57 | MaliutaLap | that picture has no walls |
| 23:59.14 | paraxor | MaliutaLap: good call! |
| 23:59.16 | paraxor | :) |
| 23:59.29 | laptopdude90 | But the studs |
| 23:59.42 | MaliutaLap | I'm not in that picture either ;) |
| 23:59.42 | laptopdude90 | I'm running a cable from the 3rd floor to the 2nd |
