00:03.21 | ChannelZ | I had a computer with an AMD K7 |
00:03.44 | snadge | im building a gnu hurd box |
00:03.48 | snadge | maybe i should install asterisk on it ;) |
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00:19.33 | scoobysnack | hurd huh? |
00:19.36 | scoobysnack | that's cool |
00:19.53 | scoobysnack | Maybe they'll get that thing to work one of these days. |
00:20.23 | scoobysnack | Been working on it for 20 years now. It takes a lot of time to do it right I guess. |
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00:29.05 | snadge | no |
00:29.23 | snadge | there was an article on slashdot.. gnu's 30th birthday |
00:29.40 | snadge | and hurd 0.5 and mach 1.4 were released to celebrate.. so i thought i'd have a play with debian gnu/hurd |
00:29.56 | snadge | i have apache2, mysql, php5 installed and working.. X is installed |
00:30.12 | snadge | i have a graphical web browser (dillo) |
00:30.59 | snadge | epiphany and arora are installable and run.. but don't work |
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00:48.45 | phix | Happy bday gnu! |
00:48.48 | phix | hip hip! |
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01:14.56 | raden | My asterisk box shows in show sip peers that my phone is registereing from 172.17.1.1 not 66.x.x.x.x like it should |
01:15.05 | raden | we changed from cisco to mikrotik router the other day |
01:15.14 | raden | weird part is i can call in and recieve calls from the box |
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01:42.43 | scoobysnack | I've noticed oddities like that too raden |
01:42.57 | scoobysnack | How do you like Mikrotik so far? |
01:45.18 | raden | mikrotik rocks |
01:45.19 | raden | love it |
01:45.23 | raden | 26 routers on network |
01:45.26 | raden | all cisco gone |
01:45.30 | raden | things run much nicer |
01:45.37 | raden | especially like the graphing and GUI |
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05:40.36 | raden | UGH |
05:41.02 | raden | ok Asterisk is now working fine after we discovered Vitelity had us reporivisioned on another soft switch |
05:41.09 | raden | I have a sligh annoyance |
05:41.41 | raden | any phone registering to the asterisk box from the WAN and not the LAN shows up in sip show peers as the router IP |
05:41.51 | raden | so instead of 66.xxx.xxx.xxx its 172.17.1.1 |
05:58.37 | raden | anyone ? |
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06:22.38 | ChannelZ | is your router playing with the traffic? |
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09:22.49 | vlad_starkov | Question: I have an issue with outbound calls from SIP softphone connected to Asterisk using TLS. Softhone registers normally. But when I make outbound call it just shows "ERROR" (I'm using GroundWire SIP softphone). In Asterisk CLI I've enabled sip debug for my softphone peer, and I see that whne I make outbound call, Asterisk returns "SIP/2.0 401 Unauthorized" and "WWW-Authenticate: Digest algorithm=MD5,...". And that's it. I don' |
09:22.49 | vlad_starkov | t see that my softphone makes new INVITE with authentication. Anyone familiar with this issue? |
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12:38.15 | Alex25 | hi |
12:38.19 | Alex25 | Where can I download DTMF tones in a natural format for Asterisk? |
12:43.18 | Tim_Toady | google is your friend, also you can generate them using something like audacity |
12:43.50 | Tim_Toady | 16 bit 1 ch 8kHz wav files will be fine for asterisk |
12:44.54 | Alex25 | on Google i found only wav type which cannot be played on asterisk, or MP3 |
12:47.31 | Tim_Toady | convert them with sox |
12:47.36 | Tim_Toady | in the above format |
12:49.58 | Alex25 | I tried sox but it didn't work |
12:50.20 | Alex25 | the wav files I downloaded came in format: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s |
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12:51.12 | Alex25 | it seems asterisk cannot play pcm_mulaw wav files |
12:51.51 | Alex25 | how to convert it to uncompressed format? |
12:53.49 | Tim_Toady | with sox as we aleady said, also why you need these and dont use the senddtmf() app |
12:55.47 | Alex25 | I need it as a part of a complex setup. I tried senddtmf but it didn't work |
12:56.37 | Alex25 | yesterday I tried to convert these wav files using several sox examples I found on Google |
12:56.44 | Alex25 | none worked |
12:57.12 | Alex25 | I got an error: check_header_fmt: Not a wav file 7 |
12:57.26 | Alex25 | fn_wrapper: Unable to open format wav |
12:59.44 | Alex25 | how to convert wav files from PCA mulaw to PCM uncompressed? |
12:59.53 | Alex25 | PCA=PCM |
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18:37.43 | cusco | hi |
18:38.39 | cusco | [2013-09-29 19:21:44] NOTICE[20666][C-000001c9]: chan_sip.c:25081 handle_request_invite: Sending fake auth rejection for device 5000<sip:5000@91.121.25.175>;tag=51b795e2 |
18:38.51 | cusco | is there a way to have the offending ip printed too? |
18:40.36 | ChannelZ | It should be in the security log, if you're looking to filter with fail2ban and things |
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18:40.59 | ChannelZ | otherwise no, not without hacking the source. |
18:51.26 | *** join/#asterisk toddross (toddross@rrcs-24-173-111-218.sw.biz.rr.com) |
18:52.46 | toddross | Good afternoon. I have a quick question: I edited my users.conf file and alphabetized the variables in each [1234] extension heading. Now the handsets won't ring or make calls, but I can still receive voicemail. Is the order of variables in users.conf important? |
18:55.04 | ChannelZ | Do you get an error on the console when you make/try to send them calls? Chances are you actually broke something else |
18:55.21 | ChannelZ | a type-o or something else causing a parse error for instance, such that none of your peers are even loaded |
18:55.48 | toddross | It's possible I introduced a typo. Where can I view the console? |
18:56.02 | ChannelZ | asterisk -rvvv |
18:56.12 | toddross | But the order of the variables themselves shouldn't be the source of the problem, right? |
18:58.05 | toddross | @ChannelZ: Here's the console output when I attempted to call my phone number: http://pastebin.com/2jZM8Zdp |
18:58.41 | toddross | (It went straight to voicemail, in case that wasn't clear in the console output.) |
18:58.54 | ChannelZ | [Sep 29 13:57:24] WARNING[31492]: chan_sip.c:5652 sip_call: No audio format found to offer. Cancelling call to 8442 |
18:59.27 | ChannelZ | I'm guessing in some fashion you disallowed all codecs but then didn't allow any. |
18:59.40 | ChannelZ | or something |
19:01.15 | toddross | Hmm… So the only file I've edited is users.conf. Are there any settings in that file that would do such a thing? |
19:01.40 | ChannelZ | Do you still have the old one? |
19:01.58 | toddross | I have a rather old backup that I can restore if all else fails. |
19:02.45 | ChannelZ | Pastebin both, let's see what you did (remove any peer secrets.) In general no the order of things within an entry shouldn't matter much. |
19:03.19 | ChannelZ | I don't use users.conf but unless it has 'defaults' that flow down from the general section like some other configs... |
19:09.48 | toddross | @ChannelZ: Here's the NEW one (the one I edited which may contain problems): http://pastebin.com/0G2rBgiZ |
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19:11.19 | toddross | @ChannelZ: And here's a backup from a year ago: http://pastebin.com/BAALgywp |
19:12.14 | ChannelZ | Move your disallow=all lines before the allow=ulaw lines. In this case I think order probably does matter. |
19:12.52 | Penguin | It definitely does. |
19:12.57 | ChannelZ | Think about those two as "commands" that manipulate a buffer |
19:13.17 | ChannelZ | Right now you "allow = ulaw" adding ulaw to the codec list, but then wipe the whole thing out afterwards with "disallow = all" |
19:13.36 | toddross | Ahh! Okay. I'll try that now. |
19:13.42 | Penguin | Nice illustration. |
19:13.53 | toddross | Would it make more sense to move allow and disallow to [general] rather that adding them to each extension? |
19:14.02 | ChannelZ | Unless you have different codecs for different users (I haven't scrolled through the whole thing) I'd put them in [general] once |
19:14.10 | ChannelZ | right |
19:14.13 | Penguin | Extensions don't have codecs, phones do. |
19:14.48 | Penguin | I always define my codecs in both general AND per device. |
19:14.58 | toddross | Interesting. Okay, I'll do that. ^^^ |
19:15.39 | Penguin | That way unauthenticated devices have one codec list, while authenticated devices have their own. |
19:16.19 | Penguin | Sometimes both lists will be the same and you don't need to specify it per device. |
19:18.02 | ChannelZ | and BTW you forgot to remove the secrets, so you may want to now change all of them.... (not fun) |
19:18.25 | toddross | lol |
19:18.30 | toddross | Good grief. |
19:18.49 | toddross | What is that a "secret" for, anyway? I don't follow the documentation. |
19:18.57 | ChannelZ | Someone should make an #asterisk pastebin that will automatically greek out secrets in pasted configs. |
19:19.22 | ChannelZ | That's the password your devices use to authenticate. |
19:19.25 | Penguin | That's a great idea. You'd better get to work! |
19:19.44 | toddross | Is it also the password entered into, say, the Polycom web-based UI? |
19:20.07 | toddross | Polycom's "Authentication password" field under Lines? |
19:20.16 | Penguin | The secret in asterisk is the password in the phone. |
19:20.21 | ChannelZ | Penguin: perhaps I will |
19:20.43 | ChannelZ | Yes, both need to match. Which is why I said "not fun" because it would mean updating all the phones as well. |
19:20.59 | toddross | Today's my lucky day, then. |
19:22.07 | Penguin | You should have an option for the paster to unhide secrets in the case of where seeing the passwords is necessary. |
19:22.43 | toddross | @ChannelZ @Penguin: The ordering of disallow/allow fixed it! Thanks so much. Now I'm going to re-passwordize all my phones. :D |
19:23.35 | ChannelZ | I doubt anyone here is going to try to exploit you but pastebin is a cesspool so now "a list" of peers and passwords are out there for someone to pick up. I might be a one in a billion chance they find your particular system, but the worst security policy is thinking you don't need one. |
19:25.21 | Penguin | I just use the password "swordfish" for everything. |
19:25.48 | ChannelZ | Penguin: I was thinking it would just be a local javascript modifier so that if enabled it would replace all the secrets before it even got posted to the server, so you wouldn't have people accusing you of secretly storing their stuff prior to substitution |
19:26.56 | Penguin | That's a good idea. That should be the default, then have a box that can be checked to keep passwords visible. |
19:27.11 | ChannelZ | right |
19:28.17 | Penguin | Would the passwords turn to hidden right there on the screen before submitting the paste? |
19:28.23 | ChannelZ | Maybe I'll just email the pastebin.com guy and see if he's interested in adding a new feature, rather than reinventing the wheel. Not that it's hard to do, but might as well use all the other functionality already existing |
19:28.40 | ChannelZ | Yes, it could |
19:29.42 | Penguin | I'm visualizing something like Windows wireless manager's password box, where it has a tick box that you can toggle to hide and unhide the password. |
19:31.06 | Penguin | I guess if using the paste type of asterisk, lines with ;secret= or secret= should be changed after the equal sign to the end of the line. |
19:31.23 | Penguin | And auto-detect would be nice. |
19:31.44 | Penguin | (for pastebin.com) |
19:33.59 | ChannelZ | Well maybe I'll prototype out one as an example and then say "here, now add this into pastebin.com!" |
19:34.34 | ChannelZ | I'd been thinking about it on and off for quite some time |
19:38.45 | Penguin | I think it's a good idea. So many people paste their passwords and don't make the paste unlisted and don't set an expiration time. |
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20:14.37 | wacomcito | hi all. |
20:15.25 | wacomcito | i have a headache with my dongle .... |
20:15.44 | wacomcito | it's all ok. i can receibe and make calls.. |
20:16.13 | wacomcito | but when i call the dongle reinnnits passed 60 seconds.. with a msg: |
20:16.28 | wacomcito | ERROR[3300]: chan_dongle.c:436 do_monitor_phone: [dongle0] timedout while waiting 'OK' in response to 'AT' |
20:16.40 | wacomcito | in a middle of the call! |
20:17.26 | wacomcito | someone can help me¿ |
20:22.23 | wacomcito | someone alive? |
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21:38.57 | davlefou | hi, |
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22:13.46 | *** join/#asterisk kresp0 (~kresp0@1.Red-83-50-237.dynamicIP.rima-tde.net) |
22:17.10 | *** join/#asterisk serafie (~erin@24.96.64.240) |
22:25.57 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
22:32.43 | *** join/#asterisk jpoz (~jpoz@c-24-21-130-6.hsd1.or.comcast.net) |
22:42.27 | *** join/#asterisk wacomcito (~ircap@11.Red-83-33-81.dynamicIP.rima-tde.net) |
22:44.42 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
23:00.42 | *** join/#asterisk serafie (~erin@24.96.64.240) |
23:18.37 | vlad_starkov | Question: If I've changed asterisk.conf is it possible to reload its content in CLI? |
23:30.19 | pabelanger | vlad_starkov, no |
23:30.26 | vlad_starkov | OK |
23:30.29 | vlad_starkov | Thnx |
23:31.53 | vlad_starkov | Question: Anyone familiar with the issue when Bria registers successfully with TLS and fails returning "503 Service unavailable" error on outbound call? |
23:41.55 | vlad_starkov | Question: What is the standard SIP response that Asterisk should send to remote peer on initial INVITE from this peer: 401 "Unauthorized" OR 407 "Proxy Authentication Required"? |
23:57.56 | *** join/#asterisk jpoz (~jpoz@c-24-21-130-6.hsd1.or.comcast.net) |