IRC log for #asterisk on 20130929

00:03.21ChannelZI had a computer with an AMD K7
00:03.44snadgeim building a gnu hurd box
00:03.48snadgemaybe i should install asterisk on it ;)
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00:19.33scoobysnackhurd huh?
00:19.36scoobysnackthat's cool
00:19.53scoobysnackMaybe they'll get that thing to work one of these days.
00:20.23scoobysnackBeen working on it for 20 years now.  It takes a lot of time to do it right I guess.
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00:29.05snadgeno
00:29.23snadgethere was an article on slashdot.. gnu's 30th birthday
00:29.40snadgeand hurd 0.5 and mach 1.4 were released to celebrate.. so i thought i'd have a play with debian gnu/hurd
00:29.56snadgei have apache2, mysql, php5 installed and working.. X is installed
00:30.12snadgei have a graphical web browser (dillo)
00:30.59snadgeepiphany and arora are installable and run.. but don't work
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00:48.45phixHappy bday gnu!
00:48.48phixhip hip!
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01:14.56radenMy 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
01:15.05radenwe changed from cisco to mikrotik router the other day
01:15.14radenweird part is i can call in and recieve calls from the box
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01:42.43scoobysnackI've noticed oddities like that too raden
01:42.57scoobysnackHow do you like Mikrotik so far?
01:45.18radenmikrotik rocks
01:45.19radenlove it
01:45.23raden26 routers on network
01:45.26radenall cisco gone
01:45.30radenthings run much nicer
01:45.37radenespecially like the graphing and GUI
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05:40.36radenUGH
05:41.02radenok Asterisk is now working fine after we discovered Vitelity had us reporivisioned on another soft switch
05:41.09radenI have a sligh annoyance
05:41.41radenany  phone registering to the asterisk box from the WAN and not the LAN   shows up in sip show peers as the router IP
05:41.51radenso instead of 66.xxx.xxx.xxx its 172.17.1.1
05:58.37radenanyone ?
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06:22.38ChannelZis your router playing with the traffic?
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09:22.49vlad_starkovQuestion: 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'
09:22.49vlad_starkovt see that my softphone makes new INVITE with authentication. Anyone familiar with this issue?
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12:38.15Alex25hi
12:38.19Alex25Where can I download DTMF tones in a natural format for Asterisk?
12:43.18Tim_Toadygoogle is your friend, also you can generate them using something like audacity
12:43.50Tim_Toady16 bit 1 ch 8kHz wav files will be fine for asterisk
12:44.54Alex25on Google i found only wav type which cannot be played on asterisk, or MP3
12:47.31Tim_Toadyconvert them with sox
12:47.36Tim_Toadyin the above format
12:49.58Alex25I tried sox but it didn't work
12:50.20Alex25the wav files I downloaded came in format: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
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12:51.12Alex25it seems asterisk cannot play pcm_mulaw wav files
12:51.51Alex25how to convert it to uncompressed format?
12:53.49Tim_Toadywith sox as we aleady said, also why you need these and dont use the senddtmf() app
12:55.47Alex25I need it as a part of a complex setup. I tried senddtmf but it didn't work
12:56.37Alex25yesterday I tried to convert these wav files using several sox examples I found on Google
12:56.44Alex25none worked
12:57.12Alex25I got an error: check_header_fmt: Not a wav file 7
12:57.26Alex25fn_wrapper: Unable to open format wav
12:59.44Alex25how to convert wav files from PCA mulaw to PCM uncompressed?
12:59.53Alex25PCA=PCM
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18:37.43cuscohi
18:38.39cusco[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
18:38.51cuscois there a way to have the offending ip printed too?
18:40.36ChannelZIt should be in the security log, if you're looking to filter with fail2ban and things
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18:40.59ChannelZotherwise no, not without hacking the source.
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18:52.46toddrossGood 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?
18:55.04ChannelZDo you get an error on the console when you make/try to send them calls?  Chances are you actually broke something else
18:55.21ChannelZa type-o or something else causing a parse error for instance, such that none of your peers are even loaded
18:55.48toddrossIt's possible I introduced a typo. Where can I view the console?
18:56.02ChannelZasterisk -rvvv
18:56.12toddrossBut the order of the variables themselves shouldn't be the source of the problem, right?
18:58.05toddross@ChannelZ: Here's the console output when I attempted to call my phone number: http://pastebin.com/2jZM8Zdp
18:58.41toddross(It went straight to voicemail, in case that wasn't clear in the console output.)
18:58.54ChannelZ[Sep 29 13:57:24] WARNING[31492]: chan_sip.c:5652 sip_call: No audio format found to offer. Cancelling call to 8442
18:59.27ChannelZI'm guessing in some fashion you disallowed all codecs but then didn't allow any.
18:59.40ChannelZor something
19:01.15toddrossHmm… So the only file I've edited is users.conf. Are there any settings in that file that would do such a thing?
19:01.40ChannelZDo you still have the old one?
19:01.58toddrossI have a rather old backup that I can restore if all else fails.
19:02.45ChannelZPastebin 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.
19:03.19ChannelZI don't use users.conf but unless it has 'defaults' that flow down from the general section like some other configs...
19:09.48toddross@ChannelZ: Here's the NEW one (the one I edited which may contain problems): http://pastebin.com/0G2rBgiZ
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19:11.19toddross@ChannelZ: And here's a backup from a year ago: http://pastebin.com/BAALgywp
19:12.14ChannelZMove your disallow=all lines before the allow=ulaw lines.  In this case I think order probably does matter.
19:12.52PenguinIt definitely does.
19:12.57ChannelZThink about those two as "commands" that manipulate a buffer
19:13.17ChannelZRight now you "allow = ulaw" adding ulaw to the codec list, but then wipe the whole thing out afterwards with "disallow = all"
19:13.36toddrossAhh! Okay. I'll try that now.
19:13.42PenguinNice illustration.
19:13.53toddrossWould it make more sense to move allow and disallow to [general] rather that adding them to each extension?
19:14.02ChannelZUnless you have different codecs for different users (I haven't scrolled through the whole thing) I'd put them in [general] once
19:14.10ChannelZright
19:14.13PenguinExtensions don't have codecs, phones do.
19:14.48PenguinI always define my codecs in both general AND per device.
19:14.58toddrossInteresting. Okay, I'll do that. ^^^
19:15.39PenguinThat way unauthenticated devices have one codec list, while authenticated devices have their own.
19:16.19PenguinSometimes both lists will be the same and you don't need to specify it per device.
19:18.02ChannelZand BTW you forgot to remove the secrets, so you may want to now change all of them.... (not fun)
19:18.25toddrosslol
19:18.30toddrossGood grief.
19:18.49toddrossWhat is that a "secret" for, anyway? I don't follow the documentation.
19:18.57ChannelZSomeone should make an #asterisk pastebin that will automatically greek out secrets in pasted configs.
19:19.22ChannelZThat's the password your devices use to authenticate.
19:19.25PenguinThat's a great idea.  You'd better get to work!
19:19.44toddrossIs it also the password entered into, say, the Polycom web-based UI?
19:20.07toddrossPolycom's "Authentication password" field under Lines?
19:20.16PenguinThe secret in asterisk is the password in the phone.
19:20.21ChannelZPenguin: perhaps I will
19:20.43ChannelZYes, both need to match.  Which is why I said "not fun" because it would mean updating all the phones as well.
19:20.59toddrossToday's my lucky day, then.
19:22.07PenguinYou should have an option for the paster to unhide secrets in the case of where seeing the passwords is necessary.
19:22.43toddross@ChannelZ @Penguin: The ordering of disallow/allow fixed it! Thanks so much. Now I'm going to re-passwordize all my phones. :D
19:23.35ChannelZI 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.
19:25.21PenguinI just use the password "swordfish" for everything.
19:25.48ChannelZPenguin: 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
19:26.56PenguinThat's a good idea.  That should be the default, then have a box that can be checked to keep passwords visible.
19:27.11ChannelZright
19:28.17PenguinWould the passwords turn to hidden right there on the screen before submitting the paste?
19:28.23ChannelZMaybe 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
19:28.40ChannelZYes, it could
19:29.42PenguinI'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.
19:31.06PenguinI 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.
19:31.23PenguinAnd auto-detect would be nice.
19:31.44Penguin(for pastebin.com)
19:33.59ChannelZWell maybe I'll prototype out one as an example and then say "here, now add this into pastebin.com!"
19:34.34ChannelZI'd been thinking about it on and off for quite some time
19:38.45PenguinI 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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20:14.37wacomcitohi all.
20:15.25wacomcitoi have a headache with my dongle ....
20:15.44wacomcitoit's all ok. i can receibe and make calls..
20:16.13wacomcitobut when i call the dongle reinnnits passed 60 seconds.. with a msg:
20:16.28wacomcitoERROR[3300]: chan_dongle.c:436 do_monitor_phone: [dongle0] timedout while waiting 'OK' in response to 'AT'
20:16.40wacomcitoin a middle of the call!
20:17.26wacomcitosomeone can help me¿
20:22.23wacomcitosomeone alive?
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21:38.57davlefouhi,
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23:18.37vlad_starkovQuestion: If I've changed asterisk.conf is it possible to reload its content in CLI?
23:30.19pabelangervlad_starkov, no
23:30.26vlad_starkovOK
23:30.29vlad_starkovThnx
23:31.53vlad_starkovQuestion: Anyone familiar with the issue when Bria registers successfully with TLS and fails returning "503 Service unavailable" error on outbound call?
23:41.55vlad_starkovQuestion: 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"?
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