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 |
18:52.17 | jonno11 | robmal: however, having it as 1 didn't work either |
18:52.34 | robmal | Ok, is the phone on a public ip? |
18:52.39 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
18:52.46 | jonno11 | robmal: I'll change it back to 1 anyway. No, it's behind a Router/NAT |
18:53.16 | robmal | So add nat=yes in extensions.conf |
18:53.30 | [TK]D-Fender | robmal: fail.... |
18:53.34 | robmal | What? |
18:53.50 | jonno11 | It's already set as yes. You guys were saying how it wasn't asterisk config... haha |
18:54.02 | [TK]D-Fender | Stop pouring windshield washer fluid in the gas tank... |
18:54.24 | robmal | jonno11: Not me, wait, the supermario of asterisk is here. |
18:54.30 | robmal | [TK]D-Fender: So, any clues? |
18:54.47 | [TK]D-Fender | nope, I've already been looking at this |
18:55.03 | jonno11 | [TK]D-Fender was really helpful earlier, but still no joy :-/ |
18:55.36 | robmal | Ok, so, go with format file system, it never hurt anyone. |
18:56.01 | jonno11 | robmal: It was fresh out of the box, though |
18:56.15 | jonno11 | (plus it takes ages) |
18:56.28 | robmal | About 5 minutes ;-) |
18:56.44 | robmal | Ok, let's try another way. |
18:56.50 | robmal | In asterisk, core set verbose 10 |
18:57.12 | robmal | Restart the phone, the log should explain what unauthorized means. |
18:57.48 | jonno11 | robmal: is that 5 mins including the 7 min reboot time? |
18:58.46 | robmal | I was also very annoyed by that in Polycoms. Then, one day, just like today, someone showed me how to provision them. |
18:59.04 | robmal | The smoke breaks got shorter :-( |
18:59.40 | carrar | Try this: |
18:59.40 | carrar | Number of Line Keys: 2 |
18:59.40 | carrar | Calls Per Line: 10 |
18:59.41 | carrar | Register: 1 |
19:01.38 | carrar | ALso show me the other web tabs |
19:03.38 | jonno11 | robmal: is it faster then? |
19:03.41 | jonno11 | when provisioned? |
19:03.50 | jonno11 | via ftp I mean |
19:04.24 | robmal | It's like switching from a horse carriage to a bullet train :-) |
19:04.49 | robmal | Also, upgrade to firmware 4.0.8, your phone will love you. |
19:05.27 | jonno11 | robmal: 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) |
19:06.41 | robmal | Well, you could run it once, but that misses the point. |
19:09.32 | Micc_ | 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. |
19:09.38 | Micc_ | could this cause one way audio? |
19:09.46 | jonno11 | robmal: for a single IP phone it seems a bit overkill |
19:10.02 | robmal | Yes. |
19:10.27 | jonno11 | robmal: so any other ideas? |
19:10.34 | Micc_ | 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. |
19:11.39 | robmal | Did setting verbose to 10 show anything interesting? |
19:11.47 | jonno11 | robmal: https://gist.github.com/jonlambert/1f12eb97ac2e12c4801a |
19:15.25 | jonno11 | robmal: anything else interesting? |
19:16.47 | robmal | In SIP set your asterisk as outbound proxy. |
19:18.53 | jonno11 | robmal: ok - have done. UDP Only yeah? |
19:18.58 | robmal | Yes. |
19:20.10 | jonno11 | robmal: now we wait... |
19:20.46 | robmal | It'll work. |
19:23.41 | jonno11 | robmal: I hope you're right |
19:26.21 | jonno11 | robmal: Not registered |
19:26.42 | robmal | Ok, what does asterisk log show? |
19:27.02 | robmal | Not the sip debug, disable that, just show me asterisk -r log with verbose 10 |
19:27.16 | robmal | It must be a password mismatch. |
19:27.37 | robmal | Or codecs. |
19:27.59 | jonno11 | robmal: asterisk shows nothing with sip debug disabled :/ |
19:28.07 | robmal | core set verbose 10 |
19:29.06 | jonno11 | yeah with verbose at 10 |
19:30.10 | jonno11 | robmal: it shows stuff when I register a softphone |
19:30.25 | jonno11 | robmal: but nothing for the 331 |
19:30.27 | robmal | Uhm, i just scrolled back up to your sip.conf |
19:30.39 | robmal | Why deny=all and permit=all ? |
19:30.47 | robmal | Drop the deny line. |
19:31.05 | jonno11 | I'm using FreePBBX |
19:31.08 | jonno11 | FreePBX |
19:31.27 | robmal | I've noticed, doesn't change a thing. Drop the deny. |
19:32.23 | jonno11 | robmal: 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 |
19:32.46 | jonno11 | robmal: allow: all, deny: none? |
19:33.11 | robmal | deny to 0.0.0.0/0.0.0.0, permit to your public ip addr / 255.255.255.255 |
19:34.02 | jonno11 | robmal: bear in mind the public ip isn't static |
19:34.43 | robmal | We'll handle that problem later. |
19:36.12 | jonno11 | robmal: like so? https://www.dropbox.com/s/esmnzaodz8100vp/Screenshot%202015-05-28%2020.36.05.png?dl=0 |
19:36.35 | jonno11 | OH |
19:36.36 | jonno11 | lol |
19:36.41 | robmal | No, in permit. |
19:36.44 | jonno11 | just saw I've been looking at the wrong bit |
19:37.03 | robmal | allow should be ulaw,alaw or something. |
19:37.22 | jonno11 | robmal: I'm going to leave it blank |
19:37.37 | jonno11 | robmal: https://www.dropbox.com/s/kt3hrdd8kiypx93/Screenshot%202015-05-28%2020.37.33.png?dl=0 |
19:37.50 | jonno11 | ^ like so? |
19:38.09 | robmal | Yep. |
19:42.55 | jonno11 | robmal: (there's a bug with FreePBX, takes 3-4 mins to save an extension) |
19:43.52 | robmal | Great software ;-) Luckily i'm making my own gui. |
19:51.27 | jonno11 | robmal: SIP/2.0 401 Unauthorized still |
19:51.42 | robmal | Asterisk log still not showing anything? |
19:53.12 | jonno11 | robmal: nope |
19:54.22 | robmal | Meh. |
19:54.35 | robmal | It doesn't work this way. Something has to appear. |
19:54.36 | *** join/#asterisk Leseratte (~florian@2001:470:70bc::1) |
19:55.21 | jonno11 | robmal: I know, that's why it's so bloody confusing |
19:55.37 | jonno11 | I've even checked by registering with another softphone account |
19:55.42 | jonno11 | It shows up fine |
19:55.49 | jonno11 | robmal: could this affect anything? https://www.dropbox.com/s/y7ugi7x29k0436q/Screenshot%202015-05-28%2020.55.12.png?dl=0 |
19:56.35 | robmal | You can set your public ip there but it should figure it out itself. |
19:57.02 | robmal | I hate that old firmware, could you try to upgrade it to 4.x ? |
19:57.28 | jonno11 | robmal: yeah ok I'll give it a go |
19:57.51 | jonno11 | robmal: do you know of any good guides? |
19:58.53 | robmal | http://www.voipsupply.com/blog/voip-insider/how-to-upload-firmware-to-a-polycom-unit/ |
19:58.55 | robmal | This looks good. |
20:00.31 | jonno11 | robmal: ok cool. where can I grab the firmware, and would you mind double checking the version so I don't brick it... haha |
20:00.42 | jonno11 | Really really appreciate the help |
20:01.02 | robmal | 4.0.8 from polycom website |
20:01.36 | jonno11 | robmal: https://www.dropbox.com/s/ptvvx3t22zljwrb/Screenshot%202015-05-28%2021.01.24.png?dl=0 so many options.. |
20:02.48 | robmal | 4.0.8, the split one. |
20:04.54 | jonno11 | robmal: ok, where does the bootrom upgrader come into it? |
20:05.12 | jonno11 | robmal: https://www.dropbox.com/s/uwhuvv4u69skiy1/Screenshot%202015-05-28%2021.05.07.png?dl=0 |
20:06.53 | robmal | You only need the UC software in your provisioning folder for now, the bootrom should be embedded. |
20:08.06 | robmal | And since we're at it, open the file reg-basic or something like that. |
20:08.15 | robmal | You 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) |
20:21.12 | ghostmediapro | help with code maintinace and cleaning: please see http://pastebin.com/k478XBpx |
20:21.40 | jonno11 | robmal: this is so overly complex |
20:22.04 | jonno11 | robmal: so the ftp server's root should contain files? |
20:22.09 | robmal | Yes. |
20:22.23 | jonno11 | FTP server's root should ONLY contain files for upgrade? |
20:22.32 | robmal | ghostmediapro: somethingsomethingcid=yes in sip.conf |
20:22.41 | jonno11 | robmal: I can't do that with OS X |
20:23.39 | robmal | We're talking about tftp, thats another protocol. But it can contain other files, polycom just won't ask for them. |
20:24.11 | jonno11 | robmal: ok so if I run tftp |
20:24.25 | jonno11 | robmal: what user credentials do I need? |
20:24.39 | jonno11 | (sorry I know this isn't asterisk related) |
20:24.47 | robmal | t stands for trival, there are no credentials by default. |
20:24.59 | [TK]D-Fender | by ever |
20:25.19 | robmal | I wasn't sure ;-) |
20:25.23 | jonno11 | robmal: ok cool |
20:27.36 | Jymmm | Has anyone played with OBi202 by chance? |
20:27.58 | jonno11 | robmal: why does the phone ask for a username and password? |
20:28.28 | robmal | Don't panic, it sometimes does that. |
20:28.58 | robmal | If you fill username and password in reg-basic it wont. |
20:29.16 | jonno11 | robmal: I mean in config |
20:29.45 | jonno11 | ServerName config section |
20:29.45 | robmal | Right 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. |
20:30.52 | robmal | You should fill reg.1.address reg.1.auth.password and reg.1.auth.userId |
20:33.19 | robmal | In sip-basic set voIpProt.server.1.address and .port |
20:37.04 | jonno11 | robmal: ok I've performed the upgrade |
20:37.13 | jonno11 | robmal: web interface is now blank... |
20:37.22 | robmal | Yay! |
20:37.26 | robmal | Go firefox ;-) |
20:37.41 | jonno11 | https://www.dropbox.com/s/2lhkvbgkkxlq5by/Screenshot%202015-05-28%2021.37.30.png?dl=0 |
20:37.51 | robmal | I hate it, because i use chrome, but it doesn't seem to work with anything other than IE or firefox |
20:43.34 | jonno11 | robmal: ok this is soooooooooo much better |
20:43.39 | robmal | :-) |
20:43.43 | jonno11 | robmal: (still not working) but so much better |
20:43.43 | robmal | I know. |
20:43.58 | robmal | So, export the config and paste it somewhere. |
20:44.24 | robmal | I think it'll be some weird router settings, like sip alg. |
20:45.33 | jonno11 | https://gist.github.com/jonlambert/2f57ddccc52b06dfdeb0 |
20:45.41 | jonno11 | robmal: ^ :) |
20:46.05 | robmal | Much better. |
20:46.55 | robmal | There should be another one with voipprot |
20:47.20 | robmal | Nvm ;-) |
20:47.24 | jonno11 | robmal: ah really? |
20:47.43 | robmal | Sorry, skipped that part, i don't know why ;-) |
20:47.48 | robmal | So... |
20:48.03 | jonno11 | robmal: haha no worries |
20:48.19 | robmal | Ok, last chance. |
20:48.20 | jonno11 | robmal: I've just added the same server as an outbound proxy |
20:48.23 | jonno11 | btw |
20:48.33 | robmal | Firewall on the asterisk server has blocked udp. |
20:49.12 | jonno11 | robmal: that would stop any softphone connecting though |
20:49.30 | robmal | Maybe softphones have udp then tcp. |
20:49.34 | robmal | Guessing right now. |
20:49.39 | robmal | Must be some freepbx shit. |
20:49.52 | drab | I 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:" |
20:50.10 | drab | anything I can do about that? guess it's somethign in the default/asmple configs conflicting with the stuff I compiled in? |
20:50.11 | jonno11 | Yeah I've no idea, this is SO FRUSTRATING |
20:50.25 | robmal | Ok, one sec. |
20:54.28 | jonno11 | robmal: Registration failed User: 101, Error Code:480 Temporarily not available |
20:55.10 | jonno11 | found that in the phone's logs |
20:58.16 | robmal | Try 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) |
21:26.32 | drab | so 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 |
21:26.44 | drab | but these are all stock, how come they produce a flood of warnings? |
21:27.05 | drab | also I only have minimal config files in /etc/asterisk and I don't understand what's loading them |
21:27.22 | drab | I see no "media" directive of sort, still reading up on it tho |
21:39.42 | newtonr | drab, hmm |
21:40.41 | newtonr | drab, 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) |
21:44.20 | drab | newtonr: sure, sec, thank you. |
21:44.56 | drab | btw 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 |
21:44.59 | drab | pasting |
21:47.33 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
21:52.44 | drab | I'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 |
21:55.35 | drab | newtonr: http://pastebin.com/Y7rZ6EjZ |
21:55.41 | drab | those are the first instances in the log file |
21:56.53 | drab | grep " Line too long" asterisk-output.txt | wc -l -> 594 |
21:57.25 | drab | when 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) |
22:07.56 | drab | I 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 |
22:08.03 | drab | anyway, brb, need to help somebody |
22:13.42 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
22:48.45 | *** join/#asterisk chandoo (~chandoo@ool-4a59659f.dyn.optonline.net) |
22:49.13 | *** join/#asterisk chandoo_ (~chandoo@ool-4a59659f.dyn.optonline.net) |
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23:02.32 | *** join/#asterisk Penguin (~xwQ5kwYl6@20264.odci.gov.united-states.rltk.us) |
23:33.23 | *** part/#asterisk kharwell (kharwell@nat/digium/x-othwuelysapzznlv) |
23:36.51 | drab | I'm new and reading the docs/learning and wondering if there's any reason on Asterisk 13 to look into chan_sip |
23:37.10 | drab | it seems like pjsip is default now and the sip phones I have work with that |
23:37.21 | drab | so I'm not sure if I should do anything with chan_sip at all |
23:44.07 | [TK]D-Fender | drab, No such thing as "default" |
23:44.21 | carrar | my setup is the default |
23:44.27 | [TK]D-Fender | pjsip has some bonus', but is newer and a littel buggier. |
23:44.51 | [TK]D-Fender | I'm sure it will soon eclipse chan_sip, but until them, chan_sip actually does the job for 99% of cases |
23:45.17 | carrar | lines all his audiocards thats he turned into ATA's along the window |
23:45.33 | WIMPy | ist sure that it's a lot less than 99%. |
23:49.09 | *** join/#asterisk Penguin (~xwQ5kwYl6@20264.odci.gov.united-states.rltk.us) |
23:52.42 | *** join/#asterisk italorossi (~Adium@179.211.186.236) |
23:55.54 | *** join/#asterisk RobertLaptop (~rmiddle@74.112.203.154) |
23:59.36 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |