| 00:01.31 | *** join/#asterisk Minotaur01 (~minotaur0@S01069094e4d4df29.hm.shawcable.net) |
| 00:11.05 | *** join/#asterisk petris (~petris@209.141.38.122) |
| 00:12.46 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 00:13.56 | *** join/#asterisk aruntomar (~Thunderbi@49.248.159.101) |
| 00:16.13 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 00:31.58 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 00:39.09 | hjf | Got Siren14 offer at 24000 bps, but only 48000 bps supported |
| 00:39.17 | hjf | i'm using CSipSimple |
| 00:39.27 | hjf | does it not support 48kbps? |
| 00:41.16 | hjf | or, how can i configure asterisk to allow for 24/32k? |
| 01:02.25 | ChannelZ | I don't know if you can control that in codecs.conf or not |
| 01:03.32 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 01:07.04 | j4jackj | We have a bofh called dfj in the network, look out |
| 01:08.37 | ChannelZ | huh? |
| 01:09.46 | j4jackj | pm him |
| 01:09.52 | j4jackj | he has me in the bozo bin |
| 01:12.11 | ChannelZ | pass |
| 01:15.25 | j4jackj | pass? |
| 01:20.59 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 01:24.26 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:29.02 | *** join/#asterisk miroesq (~quassel@196.219.92.125) |
| 01:29.25 | miroesq | Has anyone successfuly setup a trunk for DIDWW? |
| 02:19.58 | hjf | hm |
| 02:20.02 | hjf | so it's a codec thing |
| 02:20.38 | hjf | i don't understand though, i have directrtpsetup and canreinvite=yes |
| 02:20.45 | hjf | and my phones are all in the same LAN |
| 02:21.04 | hjf | and tcpdump shows no UDP data, just some SIP every now and then during a call |
| 02:21.14 | hjf | why can't the phones talk whatever codec they want |
| 02:21.15 | hjf | ? |
| 02:21.22 | Penguin | If your codec bit rate isn't supported, you're going to have problems. Choose a codec that works. |
| 02:21.52 | hjf | yeah that's weird, csipsimple has g7221 at 8 16 32 |
| 02:22.02 | hjf | but that doesn't seem standard |
| 02:22.12 | hjf | i wanted HD calls :( |
| 02:22.22 | hjf | g.711 works but it's.. meh |
| 02:22.23 | Penguin | Asterisk supports g722. |
| 02:22.40 | hjf | at 48k |
| 02:22.42 | Penguin | And so does csipsimple. It is what I use. |
| 02:23.46 | Penguin | I'm using g722 16 kHz. |
| 02:25.22 | hjf | oh |
| 02:25.37 | hjf | i just realize there's a g722 not g7221 |
| 02:25.42 | hjf | lol |
| 02:25.50 | hjf | also, i get no sound with g726 |
| 02:25.56 | j4jackj | heh |
| 02:26.16 | Penguin | I thought when you said g7221, that was a typo. |
| 02:27.42 | hjf | i suppose it's g.722.1 |
| 02:27.57 | hjf | also, is alaw supposed to sound pretty much like ulaw? |
| 02:28.17 | Penguin | They're basically the same. |
| 02:28.21 | hjf | yeah |
| 02:28.25 | Penguin | Both g711. |
| 02:28.34 | hjf | but for some reason i heard my brother a little more "robotic" when he set up the call |
| 02:28.41 | hjf | and noticed it was ulaw |
| 02:28.48 | hjf | then i called him and it sounded more normal |
| 02:28.50 | hjf | and it was alaw |
| 02:28.54 | hjf | maybe i'm just crazy |
| 02:29.06 | Penguin | That tells me that you have misconfigured a bunch of stuff. |
| 02:29.16 | Penguin | You're clearly allowing too many codecs. |
| 02:29.33 | hjf | yes, i disallwed all now, and only allowed 722, 711 and gsm |
| 02:31.25 | j4jackj | Also, I need to know whether dialup over VoIP is possible, let alone plausible. |
| 02:31.55 | hjf | no |
| 02:32.05 | hjf | my credit card terminal connects at 1200bps |
| 02:32.09 | hjf | and it doesnt work |
| 02:32.13 | hjf | :P |
| 02:32.34 | hjf | no idea what codec it uses though |
| 02:32.56 | hjf | cause i never bothered to check. it's one of those linksys thingies preconfigured by the provider |
| 02:39.56 | *** join/#asterisk dms (~dms@2601:0:5480:b4:7c1e:4792:7177:8d9f) |
| 02:40.38 | hjf | for indications.conf, i only have to set up the default country? |
| 02:40.45 | hjf | or do i have to set it somewhere else? |
| 02:41.06 | ChannelZ | Assuming your bandwidth is stable, it should work with ulaw. Faxing is possible over ulaw |
| 02:41.36 | Penguin | What country are you in? |
| 02:41.42 | ChannelZ | indications you set the country, presumably one that matches the definitions below |
| 02:41.59 | hjf | Argentina |
| 02:42.01 | ChannelZ | else it's of no use |
| 02:42.11 | hjf | i found a template and copied it to a section called [ar] i added |
| 02:42.14 | hjf | and then i set default=ar |
| 02:42.36 | Penguin | That should do it, as long as the values you set are all good values. |
| 02:42.55 | hjf | how do i apply the changes? |
| 02:43.00 | hjf | on the asterisk cli |
| 02:43.15 | Penguin | I think you have to restart asterisk for that one. |
| 02:43.30 | *** part/#asterisk ticklemeozmo (ticklemeoz@2600:3c03::f03c:91ff:feae:f7b1) |
| 02:45.46 | hjf | ring = 425/1000,0/4500 |
| 02:46.09 | hjf | still sounds like burrrrr when it rings |
| 02:46.55 | ChannelZ | When what rings |
| 02:47.07 | hjf | when it's calling |
| 02:47.17 | hjf | the ringback tone |
| 02:47.23 | ChannelZ | "it" what |
| 02:47.30 | ChannelZ | Keep in mind that most devices generate their own progress tones |
| 02:49.19 | ChannelZ | If you Answer the channel and then call something else, asterisk will generate inband progress.. but prior to Answer()ing, it's usually the local device. |
| 02:50.04 | ChannelZ | wow. I just realized * keeps all the indication sets loaded regardless of selection. |
| 02:53.07 | j4jackj | <PROTECTED> |
| 02:54.11 | hjf | ChannelZ: i see |
| 03:04.49 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 03:09.12 | *** join/#asterisk dimitry7 (~antonello@187.174.147.162) |
| 03:09.36 | dimitry7 | hi guys |
| 03:09.55 | *** join/#asterisk aruntomar (~Thunderbi@49.248.155.197) |
| 03:10.33 | dimitry7 | have you ever found this error: v |
| 03:10.34 | dimitry7 | <PROTECTED> |
| 03:10.46 | dimitry7 | In a call from a cell phone to my network |
| 03:16.45 | *** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
| 03:16.47 | *** part/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
| 03:19.58 | ChannelZ | never. What technology |
| 03:22.32 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 03:33.44 | dimitry7 | ChannelZ, really? .. :-S, What would you do in my case? |
| 03:34.03 | j4jackj | dimitry7: what technology is in use? |
| 03:34.56 | dimitry7 | j4jackj, asterisk 1.8, dahdi 2.7, OpenR2 and Wanpipe 7.0.5 |
| 03:35.52 | dimitry7 | j4jackj, open R2 1.3.3 |
| 03:36.00 | j4jackj | dimitry7: what protocol between cell and *? |
| 03:37.18 | dimitry7 | j4jackj, it is an external call, cell has it's normal coverage network, could be GSM maybe |
| 03:37.57 | j4jackj | dimitry7: between the PSTN and asterisk? |
| 03:38.06 | dimitry7 | j4jackj, yes, exactly |
| 03:38.17 | dimitry7 | j4jackj, would you like to see any configuration that I have? |
| 03:39.04 | j4jackj | sip.conf, dahdi.conf(?) or celliax.conf, and extensions.conf. |
| 03:39.36 | dimitry7 | j4jackj, perfect, hold on a sec plase |
| 03:39.39 | dimitry7 | * please |
| 03:40.20 | j4jackj | can you email me the files (anonymised of course) to jjovereats-other@yahoo.co.uk ? |
| 03:40.48 | Penguin | What? That's a first. |
| 03:40.50 | Penguin | ~pb |
| 03:40.51 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 03:41.12 | j4jackj | I can't use pastebins, I'm at the wirtual console and my mouse is FUBAR. |
| 03:41.21 | dimitry7 | j4jackj, of course |
| 03:41.30 | *** join/#asterisk pcAngel (yoink@96.49.186.163) |
| 03:41.31 | Penguin | /msg |
| 03:42.01 | j4jackj | long config files belong in inboxes, not in PMboxes. |
| 03:42.01 | dimitry7 | j4jackj, hold on please |
| 03:42.42 | pcAngel | Does anybody know the scope of the AST-2013-005 and AST-2013-004 vulnerabilities/bugs patched in Asterisk 11.5.1? I have servers on 11.5, 11.4 11.2 and 10.5 and am curious how to know if I need to update all of them |
| 03:45.17 | j4jackj | assume all vulns are 'update now if you have even the slightest inklng that you're afflicted' |
| 03:46.13 | pcAngel | that's actually a terrible policy |
| 03:46.23 | j4jackj | not really |
| 03:46.24 | pcAngel | I hope noone listens to you saying that |
| 03:46.38 | j4jackj | its a better safe than sorry policy |
| 03:46.45 | pcAngel | If someone is on an LTS release, like 1.8, and it isn't affected, you'd suggest they upgrade to 11.5.1? |
| 03:47.00 | j4jackj | if it doesn't affect you, of course don't upgrade. |
| 03:47.09 | pcAngel | Which is why I asked |
| 03:47.12 | pcAngel | my servers are all off |
| 03:48.07 | j4jackj | heh |
| 03:48.40 | dimitry7 | j4jackj, ready! |
| 03:48.54 | pcAngel | you're a bit ignorant j4jackj. I really hope nobody listens to you, since you're asking really stupid questions like how to reboot a linux server...hah. |
| 03:49.01 | pcAngel | Go suck a dick |
| 03:49.09 | j4jackj | I KNOW HOW TO REBOOT LINUX. |
| 03:49.11 | pcAngel | don't offer incorrect advice |
| 03:49.20 | j4jackj | pcAngel: the dick is yours to suck |
| 03:49.54 | pcAngel | <j4jackj> How Must I now reboot the computer? |
| 03:49.55 | pcAngel | <j4jackj> should I lspci and see what happens? |
| 03:50.06 | j4jackj | I was confused. |
| 03:50.19 | j4jackj | pcAngel: Yaou're riling me up, you're ignored. Bye. |
| 03:50.20 | pcAngel | you can't be a noob one time I'm here and then an idiot pretending to know anything, the next time |
| 03:50.24 | pcAngel | cheers. |
| 03:50.29 | pcAngel | slaps j4jackj around a bit with a large trout |
| 03:50.53 | dimitry7 | j4jackj, i've sent you my config dude |
| 03:51.03 | pcAngel | Don't trust him dimitry, seriously |
| 03:51.16 | dimitry7 | pcAngel, could you help me? |
| 03:51.21 | pcAngel | What's the problem? |
| 03:51.32 | dimitry7 | pcAngel, let me explain please |
| 03:52.08 | dimitry7 | pcAngel, Chan 4 - MF back cycle timed out! .... I get that error when sb calls from PSTN (cellular mobile) |
| 03:53.13 | pcAngel | I've never used Dahdi |
| 03:53.16 | dimitry7 | i am on asterisk -rvvvv |
| 03:54.50 | j4jackj | I'm not fun to talk to... |
| 03:55.09 | wm_domino | that user has been banned from 1/2 spotchat and a good portion of freenode, i really really hope you aren't allowing him to give any info in here :P |
| 03:56.01 | wm_domino | i'm waiting for the troll to leave so i can ask some newbie questions :)) |
| 03:56.40 | dimitry7 | wm_domino, lol ... -S |
| 03:56.57 | j4jackj | Tis all greek to me |
| 03:57.05 | j4jackj | wm_domino: what are they? |
| 03:57.34 | pcAngel | am I the troll or j4jackj? |
| 03:58.25 | wm_domino | well, i was lead to believe asterisk would be better for me, but it's well over my head .. my main trouble is getting the actual number cheat & i'm only looking to hook up a small home to analog on the cheap |
| 03:58.30 | wm_domino | not you pcAngel |
| 03:58.44 | wm_domino | cheap* not cheat |
| 03:59.17 | pcAngel | Which country? |
| 03:59.43 | wm_domino | so i suppose my real question is how difficult is it to bypass the hardware for magicejack or a way to plug in directly to google voice -- usa - midwest |
| 04:00.25 | *** join/#asterisk aruntomar (~Thunderbi@49.248.155.214) |
| 04:00.44 | j4jackj | you need a real ATA |
| 04:00.48 | wm_domino | google voice without google on my own server via asterisk would be perfect |
| 04:00.54 | pcAngel | Back when I only needed one or two, there was a company I used specifically because you could purchase just one DID, and it was flat rate (5/month or 10/month) for 2 channels.. I don't know what the rules in this channel are though, it'd probably appear as an advertisement |
| 04:01.05 | j4jackj | sorry guy! but you need a real ATA, a magicjack won't do. |
| 04:01.52 | dimitry7 | let's talk ablout asterisk |
| 04:01.55 | wm_domino | fyi, there is zero way i'm taking anything j4jackj says in here as i've seen what he tried to pull off in linuxmint and a few other places |
| 04:02.09 | pcAngel | you could forward one in to asterisk through SIP and then bridge it to gtalk |
| 04:02.16 | j4jackj | wm_domino: you're quite trolllike. I am only a troll when others troll me. |
| 04:02.40 | j4jackj | you're clearly pulling c**p on me. |
| 04:03.44 | j4jackj | wm_domino: stop with the poopulling and maybe I'll help. |
| 04:03.53 | wm_domino | perfect setup for me would be 5 inbound free googlevoice #'s, set up to 5 outbound analog in the house with being able to do all those sweet things asterisk can do, voicemail, caller id, can even forware to cell and spoof the id to the voip line? |
| 04:04.01 | j4jackj | wm_domino: but otherwise, give up trying to get help from anyone. |
| 04:04.17 | j4jackj | wm_domino: that's an interesting nut. |
| 04:05.07 | j4jackj | I think you need a real, non-free VoIP Service provider. It will get you further with better calls. |
| 04:05.17 | wm_domino | so i should look into acquiring did numbers? |
| 04:05.45 | j4jackj | wm_domino: maybe you should. but right now, if your Asterisk supports it, try GV. |
| 04:05.51 | wm_domino | pstn, i read that, hold on a sec |
| 04:06.41 | pcAngel | Depending on your use I recommend just setting up a pay as you go sip trunk & DIDs for whatever your purpose is - if you're looking for desk phones or phones on your counter, lots of companies, including digium, have base models suitable for homes |
| 04:06.45 | wm_domino | yes, pstn is what i need/want i believe, just voice & fax really, but being able to do vid would be sweet |
| 04:06.50 | pcAngel | I have an snom 870 on my desk and love it |
| 04:08.02 | wm_domino | i don't think i could go for real digium phones, mostly because i wouldn't really use what they are for and $$ |
| 04:08.21 | wm_domino | my roomates can't operate the current phone nor a vcr LOL |
| 04:09.34 | wm_domino | the pci card looked great, perfect for me, just have to figure out how to get magic jack/gv or "sip trunk & did's" ? into the card |
| 04:10.01 | j4jackj | SIP trunks don't need PCI cards. |
| 04:10.50 | pcAngel | a SIP trunk and a DID will just get asterisk to communicate with the outbound world |
| 04:11.15 | pcAngel | From there the deskphone example was just because I imagine you don't want to be calling from a computer with a headset |
| 04:11.19 | pcAngel | although I could be wrong |
| 04:11.40 | pcAngel | Personally I use a regular SIP based phone, as well as a sip client on my cell phone, and one on each of my computers |
| 04:11.58 | pcAngel | phones range from 50 to ... any price |
| 04:12.41 | pcAngel | depending on the features. I don't know the price Digium charges for them. It's definitely not necessary to use asterisk & to be able to use asterisk to call out to the rest of the world |
| 04:13.36 | pcAngel | x-lite is a soft phone (one that runs on a computer) that was turned in to freeware, I use their higher end soft phone on my computers, Bria |
| 04:14.42 | wm_domino | hmm |
| 04:15.07 | j4jackj | SIP trunks only need PCI if a PCI ADSL modem is your internet conection |
| 04:16.18 | wm_domino | ok, well i'll be back with more questions i suppose if you all don't mind, but i'll be waiting for the berk to go away tbh - thanks much |
| 04:16.49 | j4jackj | i'm not a c**t because you just called me one, and I',m not, so get your fat ass out of here |
| 04:16.53 | j4jackj | and don't come back |
| 04:21.44 | *** join/#asterisk Kalidarn (~Kalidarn@unaffiliated/kalidarn) |
| 04:23.04 | j4jackj | wm_domino: I can see right through the rhyming slang |
| 04:28.27 | ChannelZ | scratches his head |
| 04:28.52 | Kalidarn | does anyone know if BETTER_BACKTRACES is possible on FreeBSD? |
| 04:34.51 | *** join/#asterisk aruntomar (~Thunderbi@49.248.156.36) |
| 04:40.24 | ChannelZ | no idear |
| 04:44.56 | dimitry7 | how can I see how many trunks does exist here in this office? |
| 04:45.02 | dimitry7 | I mean telephony trunks |
| 04:50.04 | *** part/#asterisk plut0 (~cory@pool-71-244-121-220.albyny.fios.verizon.net) |
| 04:54.24 | dimitry7 | unable to create channel of type unicall |
| 04:54.42 | dimitry7 | plut0 es bien plut0 |
| 04:54.42 | Penguin | Sounds like you don't have a unicall device available. |
| 05:09.20 | Kalidarn | does anyone know if BETTER_BACKTRACES is possible on FreeBSD, in menuselect I see: XXX BETTER_BACKTRACES |
| 05:09.31 | Kalidarn | despite having binutils |
| 05:11.06 | Kalidarn | also devel/libbfd conflicts with devel/binutils |
| 05:11.25 | Kalidarn | it won't let me have libbfd if I already have binutils installed |
| 05:21.00 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:24.38 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 05:25.33 | Penguin | And you need both? |
| 05:26.29 | Kalidarn | i think libbfd is inside binutils |
| 05:26.43 | Kalidarn | either way i've tried having either installed and BETTER_BACKTRACES comes up with XXX |
| 05:26.58 | Kalidarn | nobody seems to know *sigh* guess it's bad backtraces for me then |
| 05:27.23 | Kalidarn | also does anyone know if asterisk actually compiles under clang? |
| 05:27.44 | Kalidarn | I had a look at https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=10649742 doesn't seem to indicate |
| 05:27.59 | Kalidarn | but i'm pretty sure FreeBSD 10 is going to be using clang as the default ports compiler |
| 05:31.28 | *** join/#asterisk Akuma (~Akuma@modemcable122.80-21-96.mc.videotron.ca) |
| 05:33.10 | Penguin | You could probably ask mjordan about the BETTER_BACKTRACES thing, but you will probably have to wait until USA regular business hours. |
| 05:33.32 | dimitry7 | Penguin, I had to comment out the unicall from extensions |
| 05:33.37 | dimitry7 | that's the solutions |
| 05:33.41 | dimitry7 | solution* |
| 05:33.53 | Penguin | I don't even have any idea what you're talking about. |
| 05:34.13 | Penguin | s/you're talking about/that means/ |
| 05:40.42 | dimitry7 | Penguin, I am talking what I asked you about! |
| 05:41.01 | dimitry7 | Penguin, if you don't have any idea reread the conversation or rephrase your sentence. |
| 05:41.20 | Penguin | I understand the words that you said, but I have no idea what it means when you arrange them in that order. |
| 05:41.20 | dimitry7 | tu turututu tu tu |
| 05:41.59 | Penguin | I understand what commenting out something is. I do not understand the "unicall from extensions" part. It does not make sense to me. |
| 05:42.26 | Penguin | "from extensions" says to me that you mean from the extensions.conf file. |
| 05:42.45 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 05:43.23 | Penguin | But the unicall thing... in extensions.conf... does not make sense. |
| 05:54.37 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.70) |
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| 06:34.26 | ChannelZ | Crystal ball malfunction |
| 06:44.53 | j4jackj | I had a crystal ball malfunction too trying to read his conffiles |
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| 10:20.34 | smirker | with standard voicemail config, can anyone point me to where mailbox passwords are stored once a user changes it from the voicemail app? |
| 10:26.20 | smirker | nevermind, i guessed it as /var/spool/asterisk/ ;) |
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| 10:31.39 | Kalidarn | hmm, if i did happen to build asterisk with clang would that be unsupported? |
| 10:36.52 | *** join/#asterisk salz212 (~chatzilla@li345-39.members.linode.com) |
| 10:37.17 | salz212 | hi. guys I having this asterisk: error while loading shared libraries: libopenh323.so.1.18.0: cannot open shared object file: No such file or directory. Need to get it resolved asap, it happend on a working Asterisk instance. |
| 10:37.51 | salz212 | I have unloaded h232 but its still not letting Asterisk to Start or accessed |
| 10:42.40 | salz212 | any clue? |
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| 11:39.34 | dongola7 | smirker: They are written back into the voicemail.conf file. |
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| 12:54.40 | _omer | hello, |
| 12:55.21 | _omer | Speech recognition (English) is working fine in my system. I need to know is there a way to recognize Hebrew Speech ? |
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| 13:26.05 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 13:27.38 | _omer | Hi Fender and tzafrir ... I have a question :) |
| 13:27.40 | _omer | Speech recognition (English) is working fine in my system. I need to know is there a way to recognize Hebrew Speech ? |
| 13:28.10 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 13:29.44 | tzafrir | _omer, not any system that I'm aware of |
| 13:33.09 | _omer | I could not find any system either so I thought to share experts knowledge .... |
| 13:33.22 | _omer | thanks. Your answer really helps :) |
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| 14:56.24 | shailender123 | I have just installed AsteriskNow 3 on my Vmware Workstation, i am facing 2 major issues: a)GUI is not coming b)after installation unable to see any asterisk commands, no sip command,no telnet and unable to see dhadi drivers loaded using ldmod |
| 14:56.24 | Nugget | telnet is eeeeeeevil! |
| 14:57.37 | shailender123 | even there is no http/apache service on which i can browse, so please help me in identifying the issue and progress in usage of asterisk system |
| 15:00.08 | *** join/#asterisk Faithful (~Faithful@123-2-11-105.static.dsl.dodo.com.au) |
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| 15:01.39 | WIMPy | If you installed it with freepbx ask in #freepbx if you installed with asterisk-gui, ask in #asterisk-gui. Iy you scrap them and want to edit config files, ask here. |
| 15:07.44 | *** join/#asterisk vlad_starkov (~vlad_star@nat.canmos.ru) |
| 15:07.50 | shailender123 | i am using asteriknNow , and major concern is why the required driver/tools are not in installed CentOs, do i need to install everything from scratch using yum, as per AsteriskNow doc,it is ready to use product |
| 15:09.29 | WIMPy | That's what I thought. |
| 15:18.08 | hjf | how can i debug what's going on with calls? i'm getting several strange situations: |
| 15:18.15 | hjf | both phones register fine |
| 15:18.22 | hjf | i call phone A to phone B |
| 15:18.30 | hjf | and both are unregistered |
| 15:18.43 | hjf | i try again and it works |
| 15:18.59 | hjf | when i hang up, both phones get unreg and register again |
| 15:19.27 | hjf | i noticed with sip set debug on in the command line, that when the call fails there's a 401 Unauthorized error |
| 15:21.42 | shailender123 | anyother point over my issue? |
| 15:22.21 | WIMPy | shailender123: Any errors while installing? |
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| 15:23.01 | *** part/#asterisk Defraz (~Defraz@209.141.122.71) |
| 15:23.28 | WIMPy | hjf: There's always a 401 while authorizing. |
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| 15:27.28 | hjf | WIMPy: now i see that, when i call phone A to B, I pick up on B, B goes back to csipsimple's home screen and it's not registered, and it registers again. meanwhile, phone A is still "ringing" (ringback) |
| 15:29.14 | shailender123 | as it is no major error, autoinstaller is installing and choosing all options and gives msg for missing gcc,make,module-dev after somehow it get installed but when i boots it not goes in GUI mode but runlevel is still 5, after checking basic uitlitis i found no http/asteris/dhadi/telnet etc |
| 15:30.01 | coreyf1513 | shailender123: how did you install the system? |
| 15:31.24 | coreyf1513 | shailender123: sorry I meant to ask what ISO did you use? |
| 15:31.37 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 15:31.56 | shailender123 | i downloaded the image from Asterix site(AsteriskNOW-3.0.0-x86_64-DVD.iso) |
| 15:32.43 | WIMPy | DVD? |
| 15:33.11 | shailender123 | i am using vmware workstation and manully attached this DVD iso to CDROM and booting from it |
| 15:34.17 | coreyf1513 | weird that it's labeled DVD, it's only 475mb |
| 15:34.38 | WIMPy | Interesting. |
| 15:35.30 | shailender123 | looks like the image available on asterisk site is not OK or not customized, it looks to me pure CentOS with removed tools :-) |
| 15:36.11 | shailender123 | even there is no sign that this image is built by digium |
| 15:36.49 | coreyf1513 | shailender123: on the first screen i'm assuming you hit option 1 (install asterisk 11 and freepbx)? |
| 15:37.17 | shailender123 | it is not asking for anything,directly it starts installation |
| 15:39.01 | shailender123 | if required i can show you installation on join.me |
| 15:40.04 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 15:40.10 | coreyf1513 | shailender123: you want help from #freepbx or #asterisk-gui depending which interface you want.. they would probably be able to help you install those |
| 15:41.09 | shailender123 | i will go there if required, but here the question is of base installation that i am not getting as it suppose to come in basic installation |
| 15:41.38 | WIMPy | Bad situation. |
| 15:41.55 | WIMPy | But I have no idea where you could get support with the installation itself. |
| 15:42.29 | WIMPy | FreePBX do teir own distro as well. You could try that. |
| 15:42.45 | WIMPy | OTOH, are you sure you want to use a GUI? |
| 15:42.53 | shailender123 | is freebox part of asterisk if not then i want to stick with asterisk only |
| 15:43.26 | shailender123 | yes GUI is required for me as i want to test softphone within that same vm image |
| 15:43.52 | hjf | I'm also getting this error (tcpdump): |
| 15:43.53 | hjf | 12:39:54.066556 IP 10.42.42.35 > 10.42.42.100: ICMP 10.42.42.35 udp port 14337 unreachable, length 36 |
| 15:44.05 | hjf | but there is no firewall |
| 15:44.27 | hjf | the asterisk process should bind to as many ports as it wants |
| 15:44.29 | WIMPy | No, Asterisk does not come with a GUI. They are all 3rd party stuff. |
| 15:44.31 | shailender123 | my requirement is to setup a new call center and as i comes from dev-ops profile so want to use asterisk only as i can do modification as per my requirement |
| 15:45.00 | WIMPy | What does using softphones have to do with using a configuration GUI? |
| 15:45.43 | WIMPy | Well, then you should install your favourite Linux distro and then instll Asterisk on that. |
| 15:45.47 | shailender123 | i will test my basic setup as i am new to asterisk , i want all required tools in same image to avoid complexity during initial setup |
| 15:46.17 | coreyf1513 | shailender123: you realize asterisk GUI's are web interfaces? |
| 15:46.21 | WIMPy | Asterisk is a very complex thing. No way around that. |
| 15:46.47 | shailender123 | agree with you, but the question is why the AsteriskNow not working,this problem reporting will help other users those who want to startup with |
| 15:47.16 | shailender123 | as i said there is no http/apache so even can 't use web browser to do testing |
| 15:47.29 | WIMPy | I guess you'd have to contact Digiums support about that. |
| 15:48.07 | shailender123 | other wise i need to strugle with config/make/mak install :-) |
| 15:48.36 | WIMPy | Which is the best way to do it. |
| 15:48.49 | shailender123 | i thought they are providing AsteriskNow , i thought to use it to minimize installation/setup time |
| 15:48.53 | WIMPy | But there are packages for the highstreet distros. |
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| 15:50.15 | shailender123 | is there any specifi emails ids where i can report the issue, as i am going to use asterisk heavily for my use in future so i require much more support from them |
| 15:50.50 | shailender123 | is there any other in this forum did asteriskNow installation? |
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| 16:10.47 | hjf | hmmm |
| 16:10.53 | hjf | so i saw this pattern: |
| 16:11.19 | hjf | when i call from phone A to B, when B picks up, it's instatly unregistered, while A keeps ringing |
| 16:11.22 | hjf | I hang up on A |
| 16:11.43 | hjf | then try to call back, from B to A. Every time it re-registers when i try to call |
| 16:11.58 | hjf | so i stop... and in the debug log i see several "really destroying session" |
| 16:12.08 | hjf | when the last really destroying appears, i can call from B to A |
| 16:16.34 | Kobaz | so i guess canreinvite/directmedia is patented |
| 16:16.36 | Kobaz | http://arstechnica.com/tech-policy/2013/08/report-after-patent-loss-apple-tweaks-facetime-and-logs-500000-complaints/ |
| 16:16.54 | Kobaz | what a crock |
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| 16:41.20 | [TK]D-Fender | [11:50]shailender123is there any specifi emails ids where i can report the issue, as i am going to use asterisk heavily for my use in future so i require much more support from them <- there is no issue |
| 16:41.41 | [TK]D-Fender | shailender123: AsteriskNOW is for installing on a server, not to act as a desktop OS. |
| 16:42.34 | [TK]D-Fender | shailender123: And there is no way they are going to add all of the extra's required to act as a desktop. So if that's what you want, you're just going to have build your own |
| 16:44.12 | shailender123 | technically there is no difference for installation of asteriskNow on a VMware or on bare metal server, it should get install |
| 16:45.26 | [TK]D-Fender | It will. And you'll get a CLI out of it just the same |
| 16:45.28 | Kobaz | that sounds like a big technical difference to me |
| 16:45.38 | Kobaz | one is real hardware and one is virtualized |
| 16:45.58 | tzafrir | if you want to use it as a desktop: grab the rest of the packages from centos |
| 16:46.11 | tzafrir | Won't be fun, but it should work |
| 16:46.30 | [TK]D-Fender | With the expected amount of work.. |
| 16:46.36 | [TK]D-Fender | to personalize X, add drivers, etc |
| 16:46.49 | Kobaz | or do what i do... run debian for your desktop and then virtualbox your development system |
| 16:47.01 | tzafrir | We have one such system (though we generally prefer Debian) |
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| 16:50.19 | shailender123 | still not agree over your point, i know the alternatives but as per current documents AsteriskNow can be installed over Vmware as vm |
| 16:50.51 | [TK]D-Fender | It should install fine. |
| 16:57.14 | Kobaz | yeap |
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| 17:01.59 | Kobaz | [TK]D-Fender: what do you think about the apple facetime stuff? |
| 17:17.15 | coppice | I think all serious criminals should face time |
| 17:20.01 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:20.56 | WIMPy | Like the great artitst that steal? |
| 17:24.42 | coppice | Many artists have criminal records. I believe Justin Beiber has produced several |
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| 18:07.34 | j4jackj | I wonder, if I compressed a G722 with <insert high compression ratio real time lossless encoder here>, and the filesize went below 50%, would I be able to send G722 over dialup in real time? |
| 18:12.34 | *** join/#asterisk _zoom_ (~zoom@197.252.1.168) |
| 18:12.36 | _zoom_ | hi, |
| 18:13.20 | j4jackj | what? |
| 18:13.24 | _zoom_ | do you think in those countries where they block voip, using only signaling encryption enough to bypass blocking |
| 18:13.34 | _zoom_ | ? |
| 18:15.19 | j4jackj | It's illegal |
| 18:15.49 | j4jackj | Also, ask your Sultan if he's willing to pass a law that will legally unblock VoIP |
| 18:16.20 | j4jackj | :D |
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| 19:37.12 | j4jackj | hi dash_ |
| 19:37.48 | dash_ | hi |
| 19:38.46 | hjf | when i call the mailbox it connects with g722 (its the first allowed codec in my default list. i have g722, alaw and gsm) |
| 19:38.54 | hjf | but my mailbox prompts are all gsm |
| 19:39.13 | hjf | can i force voicemail to connect as gsm and avoid transcoding? |
| 19:39.54 | j4jackj | not without making gslm your prefered |
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| 20:03.47 | ericmcray | hello guys. I have a old PC and I want to turn that into a PBX server with asterisk. I'm going to buy TDM400 for phone line. I'll just use ip phones and softphones (Win7). So, which modules i need for TDM400, FXS or FXO ? and can i use motherboard's ethernet for ip phones or need i buy a card for that too?? Thank! |
| 20:09.35 | hjf | ericmcray: FXO -> receives RING. FXS -> makes RING |
| 20:09.37 | ChannelZ | To connect to a phone line, you need an FXO card |
| 20:10.00 | ChannelZ | And whatever ethernet connection you have is fine |
| 20:10.45 | ericmcray | So i dont need FXS for ip phones and softphone right? |
| 20:10.53 | ChannelZ | nope |
| 20:10.56 | hjf | ericmcray: only if you want to connect analog phones |
| 20:11.04 | ChannelZ | FXS would be if you wanted to plug an analog phone in (or a FAX machine, etc.) |
| 20:12.18 | ericmcray | Okey. last question :) do you know any cheaper solution for single line? |
| 20:12.39 | ChannelZ | Linksys SPA3102 but your happiness may vary |
| 20:12.48 | ChannelZ | They have echo problems that are tough to get rid of |
| 20:13.12 | ChannelZ | (and Linksys is now Cisco, you'll find either) |
| 20:13.40 | hjf | i need a cheap fxo card that will work on freebsd |
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| 20:13.53 | hjf | i was surprised at just how expensive Sangoma cards are |
| 20:14.02 | *** part/#asterisk russum (~russum@ool-4574dc67.dyn.optonline.net) |
| 20:14.02 | hjf | $400 for a single-liner (here in argentina) |
| 20:14.36 | ericmcray | you are lucky to find a card |
| 20:15.15 | ericmcray | in turkey i couldnt find any card either 56k :D |
| 20:16.34 | ericmcray | Linksys SPA3120 is 75$ and TDM400 with 2 FXS and 2FXO is 75$ at ebay |
| 20:17.44 | ChannelZ | Well given ued at the same price. go for the TDM |
| 20:18.36 | ChannelZ | s/ued/used/ |
| 20:20.56 | ericmcray | Thanks guys! |
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| 20:26.13 | ChannelZ | Yup good luck |
| 20:27.47 | j4jackj | Does Asterisk support OPUS? |
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| 20:45.10 | *** join/#asterisk monsterco (40e7ac8e@gateway/web/freenode/ip.64.231.172.142) |
| 20:46.34 | monsterco | Hi everyone - we have never opened our SIP network to unknown IPs but now we have to open RTP ports as our provider uses third parties for media stream which means we have to allow them in for UDP ports with unknown IPs. This opens us to UDP flooding and who knows what other types of attacks.....so my question: What are some of the ways we can protect ourselves? We use pfSense as firewall |
| 20:47.34 | j4jackj | You can open the SIP network to your provider ONLY and tell them to proxy it |
| 20:50.06 | monsterco | j4jackj - So, our provider has given us their SIP proxy IP which we have on white-list but then they don't handle media stream so I think media comes through different IPs so we have to have UDP port 10000-12000 for example opened to all IPs |
| 20:50.11 | monsterco | am I missing something there? |
| 20:50.21 | j4jackj | Don't use that provider. |
| 20:50.38 | j4jackj | Sorry guy, but that provier is !@#$. |
| 20:51.49 | monsterco | ????? but a lot of providers don't handle media streams as they are connected with other partners |
| 20:52.18 | j4jackj | Open to their third parties or get ISDN |
| 20:53.50 | monsterco | I am sure the list of third parties is a dynamic one - so it's not easy to keep up either for them or for their clients |
| 20:54.32 | monsterco | ISDN is not an option - as for as I can see many providers work on same basis and they take precautions and have systems in place - that is what I need advice on |
| 20:55.10 | j4jackj | Then don't use that provider |
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| 20:59.24 | WIMPy | monsterco: Use the sip connection tracker. But what does it help you to filter? |
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| 21:03.35 | monsterco | WIMPy - thanks for the feed back - what is SIP tracker? |
| 21:04.00 | WIMPy | A firewall module. |
| 21:05.05 | monsterco | so it reads each SIP packet and adds the media stream IP to allowed list in iptables otherwise all UDP packets get dropped? |
| 21:05.16 | WIMPy | But no firewall will help agains you beeing flooded. |
| 21:05.24 | j4jackj | Yup |
| 21:05.28 | WIMPy | Exactely. |
| 21:05.59 | monsterco | I understand DDoS can still happen - but if I can make a system where I don't give the impression that we are running VoIP servers then attackers might just move on quickly |
| 21:06.29 | monsterco | so does SIP Tracker work quick enough? and does it work with fail2ban? |
| 21:06.43 | WIMPy | How would they know, just because you have UDP ports open? |
| 21:06.44 | j4jackj | we do need f2b...? |
| 21:07.00 | WIMPy | It's realtime. |
| 21:07.28 | WIMPy | And I don;t see what fail2ban can do for you. |
| 21:09.27 | monsterco | ok, so what happens with the first packet coming in? does that get responded to and then SIP Tracker kicks in or even the first packet can be dropped? |
| 21:10.16 | WIMPy | It only reads the SIP packets and adds the connections it finds to the list of related connections. |
| 21:11.00 | WIMPy | All just normal firewall shit. |
| 21:12.37 | monsterco | and then drops them from iptables after call is done? |
| 21:13.45 | WIMPy | I don't think they are actively dropped. |
| 21:14.33 | monsterco | thanks for the tip - are there any other precautions I have to take? wondering if anything should be done on pfSense end as well |
| 21:15.21 | WIMPy | Ask there. |
| 21:16.02 | WIMPy | hasn't touched anythign BSD for over a decade. |
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| 22:54.13 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 23:13.08 | *** join/#asterisk zerohalo (~zerohalo@c-174-62-142-83.hsd1.ma.comcast.net) |
| 23:37.57 | *** join/#asterisk Meaw (~salem@146.185.62.88) |
| 23:39.06 | *** part/#asterisk monsterco (40e7ac8e@gateway/web/freenode/ip.64.231.172.142) |
| 23:55.25 | *** join/#asterisk Draecos (~Draecos@124.148.202.165) |