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| 05:38.08 | kmyst | hi, just getting into asterisk and i've just installed it and working my way through the hello world bit of the wiki, my softphone registers, i dial the extension and i can see the right output in the cli but i don't hear the hello-world.gsm file play; it is instaled i see that it's under /var/lib/asterisk/sounds/en so i'm kind of stuck...any suggestions? |
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| 06:12.25 | ChannelZ | Does the console say it can't find it or that it is playing it or..? |
| 06:12.30 | ChannelZ | (core set verbose 3) |
| 06:14.34 | *** join/#asterisk rexwin_ (~rexwin@223.30.53.254) |
| 06:14.44 | rexwin_ | I am getting file.c:666 ast_openstream_full: File hello-world does not exist in any format |
| 06:14.54 | kmyst | ChannelZ: it says it is playing it |
| 06:15.11 | rexwin_ | file.c:957 ast_streamfile: Unable to open hello-world (format 0x4 (ulaw)): No such file or directory |
| 06:15.11 | ChannelZ | how ironic |
| 06:15.43 | ChannelZ | rexwin_, it says what it means. Either the sounds are in the wrong place, or perhaps you have a permissions problem |
| 06:16.17 | ChannelZ | kmyst, what softphone? Is it on the same LAN as asterisk or is the asterisk box remote? |
| 06:16.48 | kmyst | ChannelZ: express talk softphone, same lan |
| 06:16.51 | ChannelZ | Make an extension that does Answer and then Echo, see if you can even hear yourself. Chances are you've got one-way audio |
| 06:17.37 | ChannelZ | although it's unusual for a simple LAN |
| 06:17.58 | rexwin_ | how to get those sounds? I am using centos 6 |
| 06:18.27 | kmyst | ChannelZ: i think i tried that and i got a pile of NOTICE:res_pjsip errors |
| 06:19.22 | rexwin_ | It is empty in /usr/share/asterisk/sounds and /var/lib/asterisk/ |
| 06:19.44 | ChannelZ | rexwin_, you installed from a package? |
| 06:19.59 | rexwin_ | yes, from epel repository I believe |
| 06:20.17 | ChannelZ | there's probably a separate "asterisk-sounds" one.. possibly "asterisk-sounds-en" or similar for language |
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| 06:20.46 | ChannelZ | else you can just download the sounds from digium, http://downloads.asterisk.org/pub/telephony/sounds/ |
| 06:20.47 | kmyst | i installed asterisk-now centos repo and installed asterisk-13 |
| 06:21.10 | ChannelZ | kmyst, would need to know what the errors actually are/say |
| 06:22.54 | kmyst | [Jul 4 01:22:27] NOTICE[2992]: res_hep.c:418 hep_queue_cb: Unable to send packet: Address Family mismatch between source/destination |
| 06:23.33 | kmyst | ChannelZ: i switched to zoiper and it now throws that up, express talk didn't show the notice |
| 06:23.39 | ChannelZ | that doesn't sound good |
| 06:23.54 | kmyst | no :/ |
| 06:24.06 | ChannelZ | I don't even know what the hell res_hep is |
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| 06:24.44 | kmyst | i haven't touched a config other than create extensions.conf and sip.conf like in the wiki so i've no idea if any of the umpteen configs in /etc/asterisk are the cause |
| 06:25.02 | ChannelZ | oh well that's part of the problem probably... |
| 06:25.37 | kmyst | just followed directions :) |
| 06:25.38 | ChannelZ | asterisk 13 has two SIP stacks, the original chan_sip and now pjsip. Sounds like you're following an old guide that uses chan_sip |
| 06:25.45 | ChannelZ | which is fine, but you need to totally turn off pjsip |
| 06:26.11 | kmyst | well i also tried deleting sip.conf and making a pjsip.conf |
| 06:26.13 | ChannelZ | make sure you have no pjsip.conf in /etc/asterisk first of all |
| 06:26.15 | kmyst | no joy there either |
| 06:26.25 | ChannelZ | well they are two completely different configs |
| 06:26.37 | ChannelZ | that are nothing alike |
| 06:26.54 | kmyst | yeah so the wiki mentions on the hello world page |
| 06:27.08 | kmyst | either way using one or the other but not both gives me no love |
| 06:28.30 | ChannelZ | well, do "core set verbose 3", and then if you're using chan_sip, do "sip set debug on" or if you're doing pjsip, do "pjsip set logger on" |
| 06:28.39 | ChannelZ | then make a test call, and pastebin the result |
| 06:32.02 | kmyst | ChannelZ: http://pastebin.com/wFmUHEkh |
| 06:32.09 | kmyst | ChannelZ: fwiw :) |
| 06:35.12 | ChannelZ | your softphone seems to be giving out its external IP and Asterisk thinks it's NAT. Your config is a little messed. |
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| 06:35.29 | ChannelZ | You also have a permissions problem (see line 469) |
| 06:35.55 | kmyst | ChannelZ: i noticed the perm bit just now |
| 06:36.30 | ChannelZ | So in your softphone, turn off any STUN/ICE/TURN support |
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| 06:37.08 | ChannelZ | as asterisk is trying to send RTP to 74.193.225.112 rather than the LAN IP 192.168.2.227 |
| 06:37.49 | ChannelZ | In sip.conf your NAT options are probably not quite right either but it's not necessarily fatal. |
| 06:38.31 | kmyst | ChannelZ: holy crap! |
| 06:38.37 | kmyst | ChannelZ: you rock :) |
| 06:38.50 | kmyst | stun whatever that was was checked in the settings |
| 06:38.59 | ChannelZ | You should set externaddr to your external IP, localnet to 192.168.0.0/16, and turn off nat= options for your softphone/whatever other devices are on the LAN |
| 06:39.19 | kmyst | righto |
| 06:40.51 | kmyst | lemme ask you this since this is my whole purpose for delving into asterisk: at my job we (for lengthy reasons) get batches of ~500 numbers every month, is there a way to feed these into asterisk and dial them to find out if they are working (should go to voicemail) or not working (disconnects) |
| 06:41.21 | kmyst | and get some kind of report? i'm sick and tired of manually checking :) |
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| 06:42.09 | ChannelZ | Probably |
| 06:44.12 | kmyst | i figured that it was possible but frankly whilst interested in asterisk it's setup being complex as it is has kept me from trying until i finally couldn't stand dialing numbers to check |
| 06:44.15 | ChannelZ | It depends on how you actually can determine an 'active' number from not. |
| 06:45.35 | kmyst | ChannelZ: well when all is right it goes straight to voicemail if the voip provider did their job, doesn't always ring though....if they botch it i either have a circuit busy signal or it just disconnects |
| 06:46.00 | ChannelZ | Are they telco disconnected numbers? Via SIP if you get a CONGESTION or something back it's simple. You can just make call files and let it dial out all day and then sift through the call logs. |
| 06:47.42 | kmyst | ChannelZ: don't get me to lying...all i know is i request X amount of numbers in specific area codes, they email me the list of numbers they assign me for the month and i manually check them since they tend to misconfigure things from time to time and using them in our application when they don't work causes me no end of extra work |
| 06:50.01 | ChannelZ | well the short answer is yes. The 'how exactly' is more nebulous |
| 06:50.22 | kmyst | so i got the bright idea of using asterisk to possibly autodial and log the results, check the report, if good use the new batch, if problems kick it back to voip provider but seeing as i know little of asterisk i'm out of my realm |
| 06:51.09 | kmyst | yeah i just need some kinda point in direction of how exactly and fumble my way thru |
| 06:51.59 | kmyst | if it wasn't nebulous it wouldn't be fun :) |
| 06:52.04 | ChannelZ | Do you just want it to autodial your softphone so you can hear what's going on, or you want this to make its own determination completely? |
| 06:53.26 | kmyst | ideally? feed it a list of numbers, auto dial, log results, parse log into something readable saying number X went to voicemail and number Y didn't do squat, etc. |
| 06:53.33 | kmyst | i don't gotta listen |
| 06:53.43 | kmyst | i just want to know the result |
| 06:54.24 | ChannelZ | well "number X went to voicemail" is the hard part, because that has no meaning to asterisk |
| 06:54.51 | kmyst | ok how about number X connected? |
| 06:54.55 | ChannelZ | There's an answering machine detection app but it's prone to false positives, a "this number has been disconnected" from a remote telco can trigger that |
| 06:55.11 | ChannelZ | But whether a call went through or was rejected by your provider is simple enough |
| 06:55.53 | kmyst | yeah no in my case it's either voicemail=working, or immediate disconnect=broken, rarely it just circuit busys which is also broken |
| 06:56.39 | ChannelZ | in any event, you could potentially do it with call files, little text files you write into asterisk's spool directory to make it fire off calls.. have it just do a Playback of 'hello-world' if you want.. and then look at the call logs afterwards to see the channel status. |
| 06:57.10 | ChannelZ | for 'immediate disconnect' it might be enough for you to see the call duration and make that determination. |
| 06:57.32 | kmyst | at this point that's kind of what i was thinking i might have to do |
| 06:58.13 | kmyst | yeah because if it does go to voicemail that's like say 15 seconds of blah blah blah beep |
| 06:58.40 | kmyst | anything that's like say 1 sec would denote bad number config |
| 06:58.43 | ChannelZ | Since you're running asterisk 13, there's ARI which is asterisk's REST interface that you could build an app out of |
| 06:58.51 | kmyst | or that's my 30,000 ft view of the problem |
| 06:59.23 | kmyst | lol programmer i am not, bash scripts and some C for sysadmin stuff sure :) |
| 07:00.09 | ChannelZ | well then look at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files |
| 07:00.37 | ChannelZ | it's the easiest way I can think of. Write a script to read the numbers from wherever they come from and generate callfiles from it |
| 07:02.26 | kmyst | hmm |
| 07:02.41 | ChannelZ | What I can't remember is I think asterisk tries to fire off any/all files if their dates are now/past. And for calling out SIP at some point your ITSP will either get pissed at you, or fail the calls.. |
| 07:03.33 | ChannelZ | so it makes sifting through the lots a bit more complicated. Although actually it writes status into the call files when it's finished with them, so you could actually parse those instead... hmmm |
| 07:03.35 | kmyst | so use iax instead of sip? |
| 07:03.57 | kmyst | dont want them getting pissed |
| 07:04.34 | ChannelZ | well IAX would do the same thing. Basically you'd like to keep only a few active channels at once, whatever their limit might be. |
| 07:04.47 | kmyst | ah i see what you mean |
| 07:05.00 | kmyst | firing off ~500 at once bad |
| 07:05.47 | ChannelZ | yeah. I mean you can modify the timestamps some seconds apart to schedule them in a manner |
| 07:06.04 | kmyst | yeah that sounds easyish |
| 07:08.50 | ChannelZ | so like I said originally, "probably" :) There's several ways to approach it, and you'll have some logic things to figure out based on your situation. |
| 07:09.31 | kmyst | thanks for the input, i'll take probably any day :) |
| 07:09.44 | kmyst | what's the worst that can happen? :) |
| 07:10.07 | kmyst | besides, always found asterisk from the little i've heard/read about it to be interesting |
| 07:10.23 | ChannelZ | it's fun to screw with to be sure |
| 07:11.01 | kmyst | prefer network administration and system administration but hey i could see digging this software |
| 07:11.53 | ChannelZ | well good luck/have fun, I need to toddle off to bed |
| 07:11.59 | kmyst | my ultimate goal is to virtualize this puppy and make it dummy proof enough i can offload the whole thing to support and let them deal with it but baby steps first |
| 07:12.07 | kmyst | yeah same here |
| 07:12.10 | kmyst | thanks! |
| 07:12.15 | ChannelZ | no prob |
| 07:12.38 | kmyst | night |
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| 16:56.12 | Cyford33_c | whts the best way to see how many calls your box can handle? |
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| 17:01.25 | [TK]D-Fender | Throw calls at it until you see failure |
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| 17:49.45 | carrar | Cyford33_c, setup SIPp traffic generator and test your box |
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| 20:26.50 | gmbs | I have about a dozen SFLphone clients connected to an Asterisk box using SIP. All are configured identically (with different extensions). A few are not receiving inbound calls. These same few are failing SIP channel prodding from the Asterisk server. They are OK outbound. Ideas? |
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| 20:30.46 | gmbs | SFLphone 1.3.0 Asterisk 11.17.1 |
| 20:31.25 | gmbs | ping times under 200ms |
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| 21:20.26 | lvlinux | gmbs: sounds like firewall issues, or a SIP ALG. |
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| 22:26.33 | antranigv | hay! in short- in which part of the doc/wiki is the configurations for the SIP? I'd just like to create 2 SIP accounts and make a test call. |
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| 22:28.43 | [TK]D-Fender | http://astbook.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html#DeviceConfig_id291081 |
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