IRC log for #asterisk on 20150528

18:52.04*** join/#asterisk infobot (ibot@rikers.org)
18:52.04*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
18:52.17jonno11robmal: however, having it as 1 didn't work either
18:52.34robmalOk, is the phone on a public ip?
18:52.39*** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl)
18:52.46jonno11robmal: I'll change it back to 1 anyway. No, it's behind a Router/NAT
18:53.16robmalSo add nat=yes in extensions.conf
18:53.30[TK]D-Fenderrobmal: fail....
18:53.34robmalWhat?
18:53.50jonno11It's already set as yes. You guys were saying how it wasn't asterisk config... haha
18:54.02[TK]D-FenderStop pouring windshield washer fluid in the gas tank...
18:54.24robmaljonno11: Not me, wait, the supermario of asterisk is here.
18:54.30robmal[TK]D-Fender: So, any clues?
18:54.47[TK]D-Fendernope, I've already been looking at this
18:55.03jonno11[TK]D-Fender was really helpful earlier, but still no joy :-/
18:55.36robmalOk, so, go with format file system, it never hurt anyone.
18:56.01jonno11robmal: It was fresh out of the box, though
18:56.15jonno11(plus it takes ages)
18:56.28robmalAbout 5 minutes ;-)
18:56.44robmalOk, let's try another way.
18:56.50robmalIn asterisk, core set verbose 10
18:57.12robmalRestart the phone, the log should explain what unauthorized means.
18:57.48jonno11robmal: is that 5 mins including the 7 min reboot time?
18:58.46robmalI was also very annoyed by that in Polycoms. Then, one day, just like today, someone showed me how to provision them.
18:59.04robmalThe smoke breaks got shorter :-(
18:59.40carrarTry this:
18:59.40carrarNumber of Line Keys: 2
18:59.40carrarCalls Per Line: 10
18:59.41carrarRegister: 1
19:01.38carrarALso show me the other web tabs
19:03.38jonno11robmal: is it faster then?
19:03.41jonno11when provisioned?
19:03.50jonno11via ftp I mean
19:04.24robmalIt's like switching from a horse carriage to a bullet train :-)
19:04.49robmalAlso, upgrade to firmware 4.0.8, your phone will love you.
19:05.27jonno11robmal: That's good to hear. Do I need a provisioning server setup all the time?
19:05.52*** join/#asterisk riess82 (~riessma@93-82-78-35.adsl.highway.telekom.at)
19:06.41robmalWell, you could run it once, but that misses the point.
19:09.32Micc_Only difference I can see with call that works and call that doesn't work is a=ptime:20 is in the one that works and only a=maxptime:20 is in the one that doesn't work.
19:09.38Micc_could this cause one way audio?
19:09.46jonno11robmal: for a single IP phone it seems a bit overkill
19:10.02robmalYes.
19:10.27jonno11robmal: so any other ideas?
19:10.34Micc_How do I get asterisk to send the ptime? These are both asterisk 13.1-cert2, but different sip.conf so here must be some diferences in the sip.conf.
19:11.39robmalDid setting verbose to 10 show anything interesting?
19:11.47jonno11robmal: https://gist.github.com/jonlambert/1f12eb97ac2e12c4801a
19:15.25jonno11robmal: anything else interesting?
19:16.47robmalIn SIP set your asterisk as outbound proxy.
19:18.53jonno11robmal: ok - have done. UDP Only yeah?
19:18.58robmalYes.
19:20.10jonno11robmal: now we wait...
19:20.46robmalIt'll work.
19:23.41jonno11robmal: I hope you're right
19:26.21jonno11robmal: Not registered
19:26.42robmalOk, what does asterisk log show?
19:27.02robmalNot the sip debug, disable that, just show me asterisk -r log with verbose 10
19:27.16robmalIt must be a password mismatch.
19:27.37robmalOr codecs.
19:27.59jonno11robmal: asterisk shows nothing with sip debug disabled :/
19:28.07robmalcore set verbose 10
19:29.06jonno11yeah with verbose at 10
19:30.10jonno11robmal: it shows stuff when I register a softphone
19:30.25jonno11robmal: but nothing for the 331
19:30.27robmalUhm, i just scrolled back up to your sip.conf
19:30.39robmalWhy deny=all and permit=all ?
19:30.47robmalDrop the deny line.
19:31.05jonno11I'm using FreePBBX
19:31.08jonno11FreePBX
19:31.27robmalI've noticed, doesn't change a thing. Drop the deny.
19:32.23jonno11robmal: yeah how do I do that with a gui? https://www.dropbox.com/s/g2zsn7jlz21za9z/Screenshot%202015-05-28%2020.31.58.png?dl=0
19:32.46jonno11robmal: allow: all, deny: none?
19:33.11robmaldeny to 0.0.0.0/0.0.0.0, permit to your public ip addr / 255.255.255.255
19:34.02jonno11robmal: bear in mind the public ip isn't static
19:34.43robmalWe'll handle that problem later.
19:36.12jonno11robmal: like so? https://www.dropbox.com/s/esmnzaodz8100vp/Screenshot%202015-05-28%2020.36.05.png?dl=0
19:36.35jonno11OH
19:36.36jonno11lol
19:36.41robmalNo, in permit.
19:36.44jonno11just saw I've been looking at the wrong bit
19:37.03robmalallow should be ulaw,alaw or something.
19:37.22jonno11robmal: I'm going to leave it blank
19:37.37jonno11robmal: https://www.dropbox.com/s/kt3hrdd8kiypx93/Screenshot%202015-05-28%2020.37.33.png?dl=0
19:37.50jonno11^ like so?
19:38.09robmalYep.
19:42.55jonno11robmal: (there's a bug with FreePBX, takes 3-4 mins to save an extension)
19:43.52robmalGreat software ;-) Luckily i'm making my own gui.
19:51.27jonno11robmal: SIP/2.0 401 Unauthorized still
19:51.42robmalAsterisk log still not showing anything?
19:53.12jonno11robmal: nope
19:54.22robmalMeh.
19:54.35robmalIt doesn't work this way. Something has to appear.
19:54.36*** join/#asterisk Leseratte (~florian@2001:470:70bc::1)
19:55.21jonno11robmal: I know, that's why it's so bloody confusing
19:55.37jonno11I've even checked by registering with another softphone account
19:55.42jonno11It shows up fine
19:55.49jonno11robmal: could this affect anything? https://www.dropbox.com/s/y7ugi7x29k0436q/Screenshot%202015-05-28%2020.55.12.png?dl=0
19:56.35robmalYou can set your public ip there but it should figure it out itself.
19:57.02robmalI hate that old firmware, could you try to upgrade it to 4.x ?
19:57.28jonno11robmal: yeah ok I'll give it a go
19:57.51jonno11robmal: do you know of any good guides?
19:58.53robmalhttp://www.voipsupply.com/blog/voip-insider/how-to-upload-firmware-to-a-polycom-unit/
19:58.55robmalThis looks good.
20:00.31jonno11robmal: ok cool. where can I grab the firmware, and would you mind double checking the version so I don't brick it... haha
20:00.42jonno11Really really appreciate the help
20:01.02robmal4.0.8 from polycom website
20:01.36jonno11robmal: https://www.dropbox.com/s/ptvvx3t22zljwrb/Screenshot%202015-05-28%2021.01.24.png?dl=0 so many options..
20:02.48robmal4.0.8, the split one.
20:04.54jonno11robmal: ok, where does the bootrom upgrader come into it?
20:05.12jonno11robmal: https://www.dropbox.com/s/uwhuvv4u69skiy1/Screenshot%202015-05-28%2021.05.07.png?dl=0
20:06.53robmalYou only need the UC software in your provisioning folder for now, the bootrom should be embedded.
20:08.06robmalAnd since we're at it, open the file reg-basic or something like that.
20:08.15robmalYou can enter you sip account details there.
20:13.34*** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it)
20:19.29*** join/#asterisk ghostmediapro (~Misc-Desk@unaffiliated/ghostmediapro)
20:21.12ghostmediaprohelp with code maintinace and cleaning: please see http://pastebin.com/k478XBpx
20:21.40jonno11robmal: this is so overly complex
20:22.04jonno11robmal: so the ftp server's root should contain files?
20:22.09robmalYes.
20:22.23jonno11FTP server's root should ONLY contain files for upgrade?
20:22.32robmalghostmediapro: somethingsomethingcid=yes in sip.conf
20:22.41jonno11robmal: I can't do that with OS X
20:23.39robmalWe're talking about tftp, thats another protocol. But it can contain other files, polycom just won't ask for them.
20:24.11jonno11robmal: ok so if I run tftp
20:24.25jonno11robmal: what user credentials do I need?
20:24.39jonno11(sorry I know this isn't asterisk related)
20:24.47robmalt stands for trival, there are no credentials by default.
20:24.59[TK]D-Fenderby ever
20:25.19robmalI wasn't sure ;-)
20:25.23jonno11robmal: ok cool
20:27.36JymmmHas anyone played with OBi202 by chance?
20:27.58jonno11robmal: why does the phone ask for a username and password?
20:28.28robmalDon't panic, it sometimes does that.
20:28.58robmalIf you fill username and password in reg-basic it wont.
20:29.16jonno11robmal: I mean in config
20:29.45jonno11ServerName config section
20:29.45robmalRight now it doesn't know where to connect, so if you entered some username and password it would ask the provisioning server for an user profile.
20:30.52robmalYou should fill reg.1.address reg.1.auth.password and reg.1.auth.userId
20:33.19robmalIn sip-basic set voIpProt.server.1.address and .port
20:37.04jonno11robmal: ok I've performed the upgrade
20:37.13jonno11robmal: web interface is now blank...
20:37.22robmalYay!
20:37.26robmalGo firefox ;-)
20:37.41jonno11https://www.dropbox.com/s/2lhkvbgkkxlq5by/Screenshot%202015-05-28%2021.37.30.png?dl=0
20:37.51robmalI hate it, because i use chrome, but it doesn't seem to work with anything other than IE or firefox
20:43.34jonno11robmal: ok this is soooooooooo much better
20:43.39robmal:-)
20:43.43jonno11robmal: (still not working) but so much better
20:43.43robmalI know.
20:43.58robmalSo, export the config and paste it somewhere.
20:44.24robmalI think it'll be some weird router settings, like sip alg.
20:45.33jonno11https://gist.github.com/jonlambert/2f57ddccc52b06dfdeb0
20:45.41jonno11robmal: ^ :)
20:46.05robmalMuch better.
20:46.55robmalThere should be another one with voipprot
20:47.20robmalNvm ;-)
20:47.24jonno11robmal: ah really?
20:47.43robmalSorry, skipped that part, i don't know why ;-)
20:47.48robmalSo...
20:48.03jonno11robmal: haha no worries
20:48.19robmalOk, last chance.
20:48.20jonno11robmal: I've just added the same server as an outbound proxy
20:48.23jonno11btw
20:48.33robmalFirewall on the asterisk server has blocked udp.
20:49.12jonno11robmal: that would stop any softphone connecting though
20:49.30robmalMaybe softphones have udp then tcp.
20:49.34robmalGuessing right now.
20:49.39robmalMust be some freepbx shit.
20:49.52drabI just finsihed compiling v13 and when I start asterisk the console/logs are filled with "WARNING[7487] media_index.c: Line too long, skipping. It begins with:"
20:50.10drabanything I can do about that? guess it's somethign in the default/asmple configs conflicting with the stuff I compiled in?
20:50.11jonno11Yeah I've no idea, this is SO FRUSTRATING
20:50.25robmalOk, one sec.
20:54.28jonno11robmal: Registration failed User: 101, Error Code:480 Temporarily not available
20:55.10jonno11found that in the phone's logs
20:58.16robmalTry using what i sent you in the msg.
21:17.51*** join/#asterisk cryptic (~cryptic@pool-98-113-133-214.nycmny.fios.verizon.net)
21:20.16*** join/#asterisk Dovid (~dovid@ool-4356e96f.dyn.optonline.net)
21:26.32drabso it seems those lines I'm being flooded by comes from these files: /var/lib/asterisk/sounds/en/core-sounds-en.txt:conf-adminmenu-162
21:26.44drabbut these are all stock, how come they produce a flood of warnings?
21:27.05drabalso I only have minimal config files in /etc/asterisk and I don't understand what's loading them
21:27.22drabI see no "media" directive of sort, still reading up on it tho
21:39.42newtonrdrab, hmm
21:40.41newtonrdrab, i've loaded up 13 several times recently and haven't seen that. Can you pastebin a debug log? https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
21:40.58*** join/#asterisk aandrew (foobar@gromit.mixdown.ca)
21:44.20drabnewtonr: sure, sec, thank you.
21:44.56drabbtw I was just looking at reducing things to the min, so right now I'm down to only 4 config files and the hello world still works
21:44.59drabpasting
21:47.33*** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire)
21:52.44drabI'm doing asterisk -rvvvv | tee asterisk-output.txt , hope that's ok, a bit more convenient and should be able to show the problem pretty clerly
21:55.35drabnewtonr: http://pastebin.com/Y7rZ6EjZ
21:55.41drabthose are the first instances in the log file
21:56.53drabgrep " Line too long" asterisk-output.txt | wc -l -> 594
21:57.25drabwhen I installed asterisk I selected all en audio files available, wasn't really sure which ones I needed/wanted and was kinda curious to try them out
22:06.46*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
22:07.22*** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de)
22:07.46*** join/#asterisk Chotaire (~chotaire@vegetarian.cannibal.club)
22:07.56drabI bet it's because I'm loading stuff like format_wav or something and those are pulling in those files, altho still don't understand why default files would be triggering a warning
22:08.03drabanyway, brb, need to help somebody
22:13.42*** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net)
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23:33.23*** part/#asterisk kharwell (kharwell@nat/digium/x-othwuelysapzznlv)
23:36.51drabI'm new and reading the docs/learning and wondering if there's any reason on Asterisk 13 to look into chan_sip
23:37.10drabit seems like pjsip is default now and the sip phones I have work with that
23:37.21drabso I'm not sure if I should do anything with chan_sip at all
23:44.07[TK]D-Fenderdrab, No such thing as "default"
23:44.21carrarmy setup is the default
23:44.27[TK]D-Fenderpjsip has some bonus', but is newer and a littel buggier.
23:44.51[TK]D-FenderI'm sure it will soon eclipse chan_sip, but until them, chan_sip actually does the job for 99% of cases
23:45.17carrarlines all his audiocards thats he turned into ATA's along the window
23:45.33WIMPyist sure that it's a lot less than 99%.
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