| 00:05.34 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.89) |
| 00:05.42 | *** part/#asterisk bananapie (~david@76-10-155-177.dsl.teksavvy.com) |
| 00:14.23 | *** part/#asterisk repat (~repat@irc.pommepause.com) |
| 00:20.29 | carbinemonoxide | I just restarted my asterisk server and now I'm getting this: http://pastie.org/8299117 |
| 00:21.36 | WIMPy | Kaputt |
| 00:22.43 | igcewieling | carbinemonoxide: reboot, if that doesn't fix it, rebuild Asterisk |
| 00:22.50 | igcewieling | if that doesn't fix it, you have a hardware problem |
| 00:22.57 | carbinemonoxide | Ugh. Thanks. :P |
| 00:23.34 | WIMPy | Or a FS with random data. |
| 00:24.32 | carbinemonoxide | killall -9 safe_asterisk |
| 00:24.39 | carbinemonoxide | Whoops. |
| 00:25.28 | hjf | i tried using a modem over voip...and failed again hah |
| 00:25.55 | WIMPy | Did you expect anythign else? |
| 00:26.11 | hjf | well it's theoretically possible |
| 00:26.17 | hjf | using g.711 |
| 00:26.19 | hjf | and low bitrates |
| 00:26.26 | hjf | modem bitrates i mean |
| 00:26.47 | WIMPy | The modem bitrate probably doesn't matter much. |
| 00:26.56 | hjf | this is for a credit card terminal which works at 1200bps |
| 00:27.13 | WIMPy | 300 might work. |
| 00:34.09 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 00:35.42 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 01:02.16 | hjf | oh |
| 01:02.22 | hjf | there's more tunables to try |
| 01:02.32 | hjf | like disabling echo suppression and things like that |
| 01:02.36 | hjf | will try it tomorrow |
| 01:15.25 | *** join/#asterisk andrewyager (~andrewyag@175.103.28.98) |
| 01:20.15 | hjf | welp |
| 01:20.20 | hjf | i couldn't resist it |
| 01:20.24 | hjf | so i went and tried... no joy |
| 01:20.33 | hjf | it seems to connect (says CONNECT_1200) |
| 01:20.46 | hjf | but then it gives an error... |
| 01:21.23 | hjf | the saddest part is: this terminal HAS an ethernet jack, but visa/mc/amex/whoever, they only accept transactions over phone line |
| 01:21.36 | hjf | OR through 3G terminals |
| 01:21.56 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 01:22.41 | WIMPy | I haven't seen non-IP credit card terminals for a while. |
| 01:23.07 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:24.28 | hjf | i dont know what's keeping them |
| 01:24.46 | hjf | maybe it's a legal thing? |
| 01:25.57 | WIMPy | Maybe the guys that clear your transactions are afraid of the evil internet? |
| 01:26.09 | hjf | probably |
| 01:26.19 | hjf | but it's visa doing that |
| 01:26.30 | hjf | so... i guess they know a thing or two |
| 01:27.09 | hjf | i mean there's a phone number for each credit card. if you pay with visa it calls a number, if you pay mastercard it dials another number, etc |
| 01:27.23 | WIMPy | We always switch the CC terminals to IP to keep them frum using up telephony channels. |
| 01:27.49 | hjf | i printed the configuration for mine, it says "server address: 192.168.0.5" |
| 01:30.28 | WIMPy | If only it was as easy for fax. |
| 01:31.22 | hjf | i'll call visa tomorrow... ask if they support IP. most likely no, but who knows |
| 01:31.53 | hjf | i have my shop wired, it's easier for me to wire another cat5 than to add an analog line |
| 01:32.41 | hjf | their answer will probaly be "if you don't have a phone line you could use our 3G service". except 3G doesn't work most of the time here, and the rent for a 3G terminal is twice as much |
| 01:33.09 | WIMPy | And would obviousely use IP. |
| 01:34.26 | hjf | someone told me "yeah that's where the 3G module plugs in" (re: the rj45 port) |
| 01:35.04 | hjf | so maybe it really is a mini router and it listens on that 192.168.0.5 IP |
| 01:47.10 | hjf | what's the feature i'm looking for called? i have incoming calls ringing to an extension: exten => 108201,1,Dial(SIP/arcana,60) |
| 01:47.19 | hjf | but i want to ring several extensions at the same time |
| 01:47.22 | hjf | not only that one |
| 01:47.26 | hjf | ring groups? |
| 01:48.55 | Penguin | You don't ring several extensions. |
| 01:49.12 | Penguin | You call to one extension and make it dial several devices. |
| 01:49.39 | Penguin | The concept is known as a ring group. It is nothing more than a string of devices in a Dial(). |
| 01:52.26 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 01:55.57 | hjf | seems my voip provider uses a hardcore codec for calls hehe |
| 01:56.11 | hjf | i called from a POTS line to my voip number |
| 01:56.13 | Penguin | hardcore? |
| 01:56.16 | hjf | and put myself on hold |
| 01:56.26 | hjf | my music on hold was really distorted |
| 01:56.40 | hjf | prolly they use GSM or something as shitty |
| 01:57.45 | hjf | sip show channels shows that, at least between them and me it was alaw |
| 01:58.09 | hjf | i mean their server is in another city |
| 01:58.59 | hjf | i'm not good at explaining things |
| 01:59.30 | hjf | Penguin: i meant to say hardcore compression |
| 02:00.57 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 02:10.37 | igcewieling | Penguin: Hardcore VoIP is rated R |
| 02:11.46 | hjf | igcewieling: shouldnt that be rated X? |
| 02:12.15 | igcewieling | qactually should be NC-17. X is not a current official rating. |
| 02:38.45 | igcewieling | How the hell did I get a paper cut from a plastic bag? |
| 02:44.02 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 02:44.45 | *** join/#asterisk mnathani (~mnathani@arp-lax.gtapcsolutions.com) |
| 02:44.55 | mnathani | How do I place a call from the command line? |
| 02:45.19 | Penguin | channel originate |
| 02:46.14 | mnathani | thanks |
| 02:52.17 | *** join/#asterisk aruntomar (~Thunderbi@49.248.153.196) |
| 02:53.44 | carbinemonoxide | Hello. I've got my Asterisk server set up to use Google Voice via motif. I have set up a mailbox for my SIP phone on asterisk, it's showing that after 20 seconds it sends the call to voicemail but on the calling end it keeps ringing Google Voice and ends up at Google Voice's vm. When I check the voicemail that is on Asterisk it apparently records Google Voice. |
| 02:54.28 | Penguin | You forgot to make asterisk answer the call and accept it by pressing 1. |
| 02:54.46 | carbinemonoxide | Oh, thanks. :) |
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| 04:14.15 | mnathani | How can I fix these errors |
| 04:14.17 | mnathani | http://pastebin.com/7vEbuWqG |
| 04:14.27 | mnathani | I am trying to generate a call using a call file |
| 04:15.08 | mnathani | this is the call file I am using http://pastebin.com/uJ7aXMNG |
| 04:16.35 | hjf | mahahaha i converted my moh to g722 - sweet |
| 04:16.35 | Penguin | The channel tech is IAX2, not IAX. |
| 04:17.20 | Penguin | And you don't seem to have permission to the directory where the call file is. |
| 04:17.46 | *** join/#asterisk zerohalo (~zerohalo@c-174-62-142-83.hsd1.ma.comcast.net) |
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| 04:18.36 | Penguin | Did you use the built-in convertor for that? |
| 04:18.50 | hjf | yes |
| 04:18.59 | hjf | with a wav file at 16khz |
| 04:19.35 | hjf | i wouldn't be mad if i was put on decent music on hold at this quality |
| 04:19.48 | Penguin | I'm surprised it accepted that. Maybe the convert feature isn't as picky as regular playing. |
| 04:20.10 | hjf | yeah i tried today and it was complaining that it only took 8000hz files |
| 04:20.22 | hjf | sox /dk.mp3 -t raw -c 1 -r 16000 /dk.sln16 |
| 04:20.33 | hjf | thats the input i used |
| 04:20.40 | mnathani | I get this now: [Sep 5 00:19:57] NOTICE[21035]: pbx_spool.c:353 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy) |
| 04:20.48 | Penguin | That isn't asterisk's built-in convertor. |
| 04:21.46 | hjf | Penguin: file convert /dk.sln16 /usr/local/share/asterisk/moh/dk.g722 |
| 04:22.14 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 04:22.32 | Penguin | Yeah, okay. |
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| 07:08.18 | PLMg | hello, have some trouble running the register utility from asterisk. I download the file but it has no extension. I is just "register". Any adivce on how I can run it? |
| 07:10.29 | PLMg | nvm, figured it out. I should not work before drinking my coffee |
| 07:25.50 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:27.57 | bacobart | does asterisk CDR register the ringtime of call? (i.e. the time it took for someone to pick up the call) |
| 07:29.46 | bacobart | if i subtract the 'billsec' from 'duration' would that be the time spent ringing? |
| 07:34.21 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
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| 08:00.29 | *** join/#asterisk vbrinza (~vbrinza@188.138.253.238) |
| 08:01.41 | vbrinza | hello all. asterisk under nat. registering 2 extensions from public IP. calling each other. can't hear the voice. |
| 08:03.08 | *** join/#asterisk tallest_red (~CNZ@ip98-169-205-124.dc.dc.cox.net) |
| 08:03.20 | vbrinza | in rtp.conf i have rtpstart=10000 |
| 08:03.21 | vbrinza | rtpend=20000 |
| 08:04.14 | vbrinza | supposing that on router the port forwarding was done correctly (i don't manage this router), what else can it check to be sure that asterisk is not the issue? |
| 08:04.56 | vbrinza | i listen the port range using tcpdump and i see no package coming on any port from this range |
| 08:05.26 | bacobart | rtp is udp |
| 08:05.46 | bacobart | you could try sip set debug on to see what's going on in asterisk |
| 08:06.16 | bacobart | but if the call gets properly setup (via sip) but you don't hear anything it's probably a port forwarding issue on the rtp end |
| 08:06.47 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:10.01 | *** join/#asterisk pii3 (~void@unaffiliated/pii3) |
| 08:10.07 | kaldemar | vbrinza: sip debug will tell you if the media was set up directly between the phones. might be that one/both of them don't support it. the parameter name for that is directmedia (or canreinvite in older asterisk versions). |
| 08:14.27 | *** join/#asterisk Alex_h (~AlexHold@178.78.119.76) |
| 08:14.51 | vbrinza | <kaldemar> i have directmedia no |
| 08:14.53 | vbrinza | right now |
| 08:15.35 | Alex_h | howdy, can anyone think of any reason that asterisk would not generate a ringing tone to the caller when the callee provides a 180 ringing? |
| 08:16.14 | Chainsaw | Alex_h: So you want Asterisk to generate inband progress tones? Does your SIP client/hardphone not do that itself? |
| 08:16.36 | *** join/#asterisk onats (~onats@unaffiliated/onats) |
| 08:17.36 | Alex_h | Chainsaw: is that what its called, inband progress? I thought as a B2B asterisk was supposed to generate tones based on the signalling, such as ringing or busy, and inband would be a message from the network such as "this number is no longer in use", which you would see as a 183 + SDP |
| 08:17.54 | Alex_h | i may have that backwards though |
| 08:18.11 | Chainsaw | Alex_h: That's exactly inverted, yes. |
| 08:18.39 | Chainsaw | Alex_h: Out of band means less CPU resources expended on the Asterisk side. |
| 08:19.13 | Chainsaw | Alex_h: You may have to ensure that the device has a country/zone setting so that it generates the correct tones though. |
| 08:19.40 | Alex_h | i was thinking this, if the loadzone is set incorrectly could it cause silence? |
| 08:20.09 | Chainsaw | Generally it'll default to US if you set it wrong, but it is device-specific. |
| 08:21.42 | Alex_h | ok, in this specific case it is a meetme actually that is the device |
| 08:21.56 | Alex_h | all parties should be able to hear the progress tones |
| 08:22.14 | Chainsaw | Then it that specific case you probably want them generated in-band. |
| 08:22.24 | Chainsaw | in that specific case, even. Typing is difficult today. |
| 08:22.37 | Alex_h | lol, no problem i get what you mean |
| 08:23.05 | Chainsaw | In which case it'll default to what you have specified in indications.conf |
| 08:23.14 | Chainsaw | If that's not country-appropriate you may have to override it. |
| 08:23.58 | Alex_h | i was focusing on the SIP messaging as we have tried various combinations of progressinband and prematuremedia, to no avail, do you think that an incorrect setting in indications.conf could cause complete silence? |
| 08:24.05 | Alex_h | actually scratch that |
| 08:24.22 | kaldemar | vbrinza: then look at sip debug and rtp debug, they will show what's going on. |
| 08:24.45 | Alex_h | going on your earlier comment a 180 ringing should be accompianied by some RTP then, if asterisk is not generating the ringing tone itself (unless set to inband signalling) |
| 08:25.08 | Alex_h | this RTP would presumably be the ringing tone from the network |
| 08:26.51 | Alex_h | ah no wait thats wrong too |
| 08:26.58 | Alex_h | its the end device that generates the ringing |
| 08:27.03 | Alex_h | the callee device |
| 08:27.11 | Alex_h | ok its making more sense now |
| 08:28.36 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 08:29.13 | Chainsaw | Alex_h: You'd have to ensure that Asterisk stays in the media path, and that progress is generated in-band. |
| 08:29.27 | Chainsaw | Alex_h: So you can't have directmedia, and you must have in-band progress. |
| 08:29.48 | Chainsaw | Alex_h: Not to mention the resources to deal with this, I wouldn't want to do this with more than a few dozen participants. |
| 08:31.28 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
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| 08:31.48 | Alex_h | Sorry chainsaw i did get what you meant earlier, was just thinking out loud to try and understand the concept |
| 08:32.00 | Alex_h | thanks for the guidance |
| 08:32.23 | Chainsaw | Any time. |
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| 10:00.17 | cusco | hi |
| 10:00.26 | WIMPy | lo |
| 10:00.36 | cusco | Im testing out asterisk 11, and looking to get the call-id |
| 10:01.03 | WIMPy | 11 is stable. 12 is for testing. |
| 10:01.09 | cusco | well.. |
| 10:01.23 | cusco | yea.. but i'm not understanding why I don't see it in the verbose |
| 10:01.52 | cusco | nor in DumpChan(); |
| 10:01.56 | WIMPy | What call id are you missing and where? |
| 10:02.38 | cusco | ah.. it goes on the log files only? |
| 10:03.05 | cusco | I was looking at cli. is there any chance to have a variable with it? |
| 10:03.27 | cusco | I'm refering to http://blogs.digium.com/2012/07/19/asterisk-11-development-call-ids-for-asterisk-logs/ |
| 10:04.38 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:04.54 | WIMPy | Only for debug as far as I can see. |
| 10:10.37 | igcewieling | you mean like this: SIPCALLID=5214447-3587357252-264302@MSX2.nyigc.net |
| 10:11.49 | igcewieling | don'y know why your dumpchan isn't showing it. |
| 10:18.09 | kaldemar | not sip call id but the call id that shows in core debug for the channel. |
| 10:19.48 | igcewieling | not familiar with that. |
| 10:21.08 | igcewieling | ah, that. indeed. |
| 10:23.31 | bacobart | does asterisk support MSRP? |
| 10:45.36 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 10:55.33 | Greenlight | With the AMI it's possible to Originate a channel and put it direclty into an application, rather than a location in the dialplan. Am I correct in in thinking there is no way to redirect an existing channel to an application in a similar way, and it would need to be put into the dialplan if I'm doing a redirect? |
| 10:59.55 | WIMPy | yes |
| 11:00.41 | Greenlight | Cool, thanks :) |
| 11:11.27 | *** join/#asterisk petris (~petris@209.141.38.122) |
| 11:14.03 | skrusty | i know i've pimped this here once before, but wondered if anyone would give me some feedback on a website indexing free (and paid for) AGI scripts: http://theagigallery.azurewebsites.net/ (note, not the final URL) |
| 11:14.18 | skrusty | just want feedback on anything obvious i've missed :) |
| 11:23.24 | skrusty | deathly silence! :) |
| 11:24.21 | Greenlight | I've bookmarked it |
| 11:24.27 | skrusty | ;) |
| 11:25.09 | skrusty | thanks - once i am happy it's got the functionality needed, i'll post it to the users mailing list |
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| 12:50.43 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:00.50 | Blashyrkh | where do i configure what asterisk puts into his contact field at registration time? |
| 13:01.51 | igcewieling | Blashyrkh: the registration section of sip.conf.sample was not helpful? |
| 13:03.32 | Blashyrkh | i have someone trying to register with my opensips, he has an asterisk installation with freepbx, and his contact field in the registration has the wrong port. |
| 13:03.39 | Blashyrkh | but i will look into it thanks |
| 13:04.43 | Katty | good morning |
| 13:07.17 | igcewieling | Blashyrkh: wrong source port or wrong destination port? |
| 13:10.07 | Blashyrkh | dest |
| 13:10.56 | *** join/#asterisk aruntomar (~Thunderbi@49.248.159.144) |
| 13:10.59 | Blashyrkh | and in the register packet, the contact header is set gives the wrong contat URI Host POrt |
| 13:11.49 | *** join/#asterisk nicknam1232 (5c15c965@gateway/web/freenode/ip.92.21.201.101) |
| 13:16.25 | igcewieling | should be 5060, right? |
| 13:16.46 | Blashyrkh | yeah |
| 13:17.16 | igcewieling | sounds to me like the person is behind a NAT router with SIP SPI / ALG which would change the ports. |
| 13:17.34 | Blashyrkh | mhm |
| 13:18.06 | igcewieling | pastebin a sip debug and hope for the best |
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| 14:06.27 | cusco | hi |
| 14:06.41 | cusco | I seem to have trouble using ${LinkedID} ?? |
| 14:06.56 | cusco | it is actually not set in https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables |
| 14:07.30 | cusco | or must I use CHANNEL() ? |
| 14:08.54 | igcewieling | linkedid is not availabel in the dialplan as far as I know |
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| 14:09.00 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:09.09 | *** join/#asterisk [404] (~404]@12.179.117.114) |
| 14:09.16 | cusco | how about CDR(linkedid) ? |
| 14:09.18 | cusco | going to try that... |
| 14:09.42 | igcewieling | have fun |
| 14:10.44 | *** join/#asterisk aruntomar (~Thunderbi@49.248.158.90) |
| 14:11.14 | cusco | ok that works ! |
| 14:11.16 | cusco | :) |
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| 14:17.35 | kaldemar | man getenv |
| 14:17.59 | *** join/#asterisk aruntomar (~Thunderbi@49.248.156.106) |
| 14:18.05 | kaldemar | i have to learn to write in the correct window... |
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| 14:25.37 | zoid_ | Is there any difference in using headerless alaw files vs the ones with wav header? |
| 14:29.06 | cusco | If i'm not mistaken, playing audio files may require transcoding if not in the same format |
| 14:29.10 | cusco | so i use alaw for everything |
| 14:29.14 | Kobaz | Plan:Astricon All Access Pass |
| 14:29.16 | Kobaz | done and done |
| 14:30.01 | [TK]D-Fender | [10:28]cuscoIf i'm not mistaken, playing audio files may require transcoding if not in the same format <- no... it ALWAYS requires it when not in the same format |
| 14:30.20 | cusco | ok I was not sure.. but my perception is correct then |
| 14:30.33 | cusco | thanks |
| 14:30.55 | zoid_ | cusco: I doo too |
| 14:31.08 | zoid_ | but, alaw is just the codec |
| 14:31.21 | zoid_ | wav (or raw) is the container |
| 14:31.58 | igcewieling | zoid_: your question is so unimportant in the Asterisk world, I doubt many people have ever considered it. |
| 14:32.09 | igcewieling | things Just Work. |
| 14:33.07 | zoid_ | ok ok, I'll just go cry in the corner then :P |
| 14:33.52 | igcewieling | zoid_: just use the files |
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| 14:34.45 | igcewieling | headerless or not asterisk will handle it |
| 14:35.47 | zoid_ | Yes, it does. I tested it. I was just curious. |
| 14:36.08 | *** join/#asterisk duchman (~paulo@41.190.2.247) |
| 14:36.26 | zoid_ | I wrote a script some time ago, to check the format of the files before uploading them |
| 14:36.55 | zoid_ | and some of the files where failing to pass the check, but worked just fine in asterisk, as .alaw |
| 14:37.01 | zoid_ | and was wondering why |
| 14:37.22 | zoid_ | and then discovered that wav headers are "optional" |
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| 14:45.45 | dorphalsig | Hi. I'm running * 11 with 15 queues and 12 agents. However, some agents are shown as 'invalid' when I do queue show. I'm under the impression that these agents do not recieve calls once they are in this status. Here is the result of queue show and sip show peer of the agent thats invalid. http://pastebin.ca/2444073 |
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| 15:04.56 | mnathani | how can I fix this error message: [Sep 5 00:45:01] WARNING[21382]: pbx_spool.c:278 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/cell.call: Operation not permitted |
| 15:06.24 | *** join/#asterisk Vann (~manny@c-76-97-51-58.hsd1.ga.comcast.net) |
| 15:06.39 | igcewieling | mnathani: fix the ownership of the .call file. |
| 15:06.47 | cusco | permissions? is the file of the same owner of the user running asterisk?= |
| 15:07.20 | cusco | sorry for doubling igcewieling |
| 15:11.53 | dorphalsig | Hi. I'm running * 11 with 15 queues and 12 agents. However, some agents are shown as 'invalid' when I do queue show. I'm under the impression that these agents do not recieve calls once they are in this status. Here is the result of queue show and sip show peer of the agent thats invalid. http://pastebin.ca/2444073 |
| 15:12.29 | dorphalsig | why does this happen and how can I fix it? |
| 15:13.27 | dorphalsig | Yesterday somebody told me it was something of the position of the agent in the queue, but I dont actually get what that means |
| 15:13.56 | [TK]D-Fender | dorphalsig: "module reload app_queue.so" then look again |
| 15:15.53 | dorphalsig | [TK]D-Fender. It just added more agents in the invalid status :( Now instead of 2 I have 4 |
| 15:16.50 | Greenlight | Don't agents have underlying "real" devices that it bases the state on? eg an actual SIP device? |
| 15:16.56 | Greenlight | COuld this be set wrongly ? |
| 15:17.23 | dorphalsig | Greenlight: each agent is a SIP extension |
| 15:17.26 | [TK]D-Fender | dorphalsig: unload it, then reload in 2 steps |
| 15:17.31 | Greenlight | Oh, I see |
| 15:17.36 | [TK]D-Fender | [11:17]dorphalsigGreenlight: each agent is a SIP extension <- no, it isn't |
| 15:17.44 | Greenlight | I thought you mean you were using Agent channels |
| 15:18.00 | dorphalsig | Command 'module unload app_queue.so ' failed. |
| 15:18.43 | dorphalsig | [TK]D-Fender. Each agent is in its own extension and that extension gets added dynamically to the queue |
| 15:19.01 | dorphalsig | or at least thats what I understood |
| 15:19.31 | [TK]D-Fender | dorphalsig: Never through the word "extension" around like that. It is a horribly vague and misused term. You have LOCAL CHANNELS added as members |
| 15:19.37 | [TK]D-Fender | dorphalsig: Not "sip" |
| 15:20.21 | Greenlight | Device states of local channels have always been a bit funky. You should use the underlying SIP channel for the state information |
| 15:20.41 | [TK]D-Fender | Greenlight: He's using FreePBX.... |
| 15:20.55 | [TK]D-Fender | Greenlight: He is not doing anything... his GUI is. |
| 15:20.56 | Greenlight | Oh, I thought freepnx did that automatically |
| 15:21.19 | Greenlight | Oh perhaps it's optional from the GUI |
| 15:27.15 | *** join/#asterisk navaismo (~navaismo@189.191.251.129) |
| 15:31.59 | igcewieling | FreePBX users often get "wrong" advice when they ask on the wrong channel. |
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| 15:32.18 | tm1000 | igcewieling: +1 |
| 15:32.48 | igcewieling | Greenlight's recommendations are correct for Asterisk, and totally wrong for FreePBX |
| 15:43.04 | dorphalsig | [TK]D-Fender. Actually I'm monitoring everything using monast. When it tells me something is not quite right I go and look through the CLI |
| 15:43.45 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 15:45.52 | dorphalsig | But at the end of they day, I still dont understand why the agents'channels appear as members of the queue in the state 'invalid' |
| 15:45.59 | dorphalsig | *status not state, sorry |
| 15:47.07 | navaismo | hi dorphalsig, i just entered in the room didi you pb the queue show cmd? |
| 15:48.30 | Greenlight | dorphalsig: You should try and avoid using state information directly on local channels. Use the underlying SIP channel for your state information. |
| 15:49.15 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:49.18 | *** join/#asterisk stanreg (~stan@38.121.253.11) |
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| 15:49.24 | *** join/#asterisk stanreg (~stan@38.121.253.11) |
| 15:50.03 | stanreg | Check this out.. I just initiated a call from google voice to my sip phone. Both devies are on mute. I can hear multiple phone calls on the line. Recording as we speak. |
| 15:50.50 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 15:50.53 | Greenlight | NSA goofed up their tap ? |
| 15:51.25 | [TK]D-Fender | stanreg: I don't see Asterisk anywhere in that chain, nor anyhthing showing that Google Voice can support SIP directly... |
| 15:52.00 | igcewieling | navaismo: he is using freepbx |
| 15:52.07 | stanreg | [TK]D-Fender, right, right.. well this just seemed like the most sip-friendly channel to talk about this. Am I breaking some rule? |
| 15:53.22 | Blashyrkh | we should open a new SIP channel :D |
| 15:53.37 | [TK]D-Fender | stanreg: just as a start you've provided no details about what is doing this conversion for you, what SIP device you're using, where any of it is located, etc. You've given no technical details about something that also isn't directly supported here. You should be a lot more thorough if you expect proper insights if we can offer even that |
| 15:54.13 | navaismo | igcewieling, uh so issues with the hints I guess |
| 15:54.16 | igcewieling | I wish I could find a channel where Asterisk is discussed. |
| 15:54.21 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:54.37 | igcewieling | navaismo: no idea, I stopped listening when I found out. |
| 15:54.47 | Blashyrkh | does anyone here have experience with running asterisk on a raspberry pi? |
| 15:55.14 | navaismo | Blashyrkh, o/ |
| 15:55.14 | Greenlight | Rasp Pi makes a great media player.. .but running asterisk on it... you're nuts! |
| 15:55.30 | Blashyrkh | just as a small pbx for my home |
| 15:55.32 | dongola7 | Blashyrkh: i run asterisk on a pi for my home phone system. seems to work well enough, though i don't do a lot of calling on it |
| 15:55.54 | Greenlight | ^^ |
| 15:55.56 | navaismo | Blashyrkh, hundred of ppl use a Pi for asterisk |
| 15:55.58 | dongola7 | Blashyrkh: it only has one SIP device attached (an Obi100) |
| 15:56.03 | navaismo | and works fine |
| 15:56.13 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f5bf:817:afce:3c2) |
| 15:56.15 | stanreg | [TK]D-Fender, ok. All I'm saying is that I initiated a call from my browser (google voice) to my sip device. Both devices are on mute and I can hear multiple phone calls. Which might, I think, point to foul play. No asterisk here. Should I take my conversation elsewhere? |
| 15:57.16 | Greenlight | It's certainly interesting...however unrelated to Asterisk. Can you make out the other concerstations ? |
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| 15:59.07 | stanreg | Greenlight, there's a lot of noise on the line. I think it might be mostly (tapping?) foreign conversations, as this "sounds" chinese and I clearly heard something like "You have X minutes remaining" (long-distance card?). I'm recording the convo now. Mics on both of my devices are muted. |
| 15:59.29 | [TK]D-Fender | stanreg: We have no idea what sip device you're using, any of the networking involved, or descriptions of how you are getting from GV to SIP. You are offering us absolutely nothing to work with |
| 15:59.37 | igcewieling | ~ask |
| 15:59.37 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:01.05 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:02.26 | navaismo | ~answer |
| 16:02.26 | infobot | 42 |
| 16:02.33 | navaismo | ~42 |
| 16:02.33 | infobot | well, 42 is the answer to life the universe and everything, see also http://en.wikipedia.org/wiki/the_answer_to_life,_the_universe,_and_everything |
| 16:02.45 | boom^time | ~towel |
| 16:02.45 | infobot | ACTION wipes off boom^time with a towel. |
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| 16:03.05 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:03.30 | igcewieling | There is nothing wrong about playing with the bot as long as you do it in private and wash your hands after. |
| 16:03.37 | stanreg | [TK]D-Fender, ok. I'll try to be specific: From my browser, I used the dial feature in gmail to call my canada-based phone number assigned to me by my sip-provider (eg: 438-xxx-xxx). |
| 16:03.39 | stanreg | My phone (438-xxx-xxxx) is running csipsimple, which is a sip client for android. I picked up and started testing for echo. Then I noticed another conversation. I muted both devices and I'm now recording. |
| 16:03.47 | boom^time | igcewieling, that's why I requested the towel. |
| 16:04.06 | [TK]D-Fender | stanreg: Wait.. there's an ITSP involved now? |
| 16:04.27 | [TK]D-Fender | stanreg: Anyh more details we should know about? How about the answer to the rest of those questions? |
| 16:04.40 | stanreg | Jeez man, why are you being so offensive? |
| 16:05.07 | [TK]D-Fender | stanreg: I'm trying to verify how far the rabbit hole goes before starting on a wrong path for a solution. |
| 16:05.38 | stanreg | The "solution" is probably to hang up and hope I don't intercept that stuff again. This is the first time it happens, I'm not looking for "help" here. |
| 16:05.39 | [TK]D-Fender | stanreg: So, any other pieces involved? Describe the networking.... |
| 16:05.48 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:06.02 | stanreg | I'm simply describing how odd it is that I am hearing someone else's conversation. |
| 16:07.16 | Greenlight | My betting is on faulty hardware at the PSTN - ITSP side |
| 16:07.22 | [TK]D-Fender | stanreg: busted reinvite or port mixup could cross a call |
| 16:07.35 | navaismo | or you are an NSA employee |
| 16:08.08 | stanreg | Greenlight, gotcha. |
| 16:08.12 | igcewieling | [TK]D-Fender: sometimes it is better to simply give up. |
| 16:08.25 | [TK]D-Fender | igcewieling: Absolutely |
| 16:10.17 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:10.41 | navaismo | if [TK]D-Fender give up then where we can look for some hope? igcewieling keep out your pessimist thoughts from fender |
| 16:10.42 | stanreg | "You have 3 dollars remaining" |
| 16:11.47 | igcewieling | navaismo: I imagine The Book, "core show applications", "core show functions" and the .conf.sample files. 8-) |
| 16:12.15 | [TK]D-Fender | igcewieling: It does get better than in addition to "eavesdropping", he's relaying that proprietary information to everyone here. |
| 16:12.51 | igcewieling | [TK]D-Fender: *nod* I'm still trying to decide if I want to /ignore |
| 16:13.18 | igcewieling | I suppose I should because of CPNI, though I don't know if that applies here. |
| 16:13.36 | Greenlight | Oh, come on, it's aleast midly interesting to see how insecure phone calls really are |
| 16:14.10 | stanreg | I could probably stream this, I think? |
| 16:14.17 | Greenlight | Umm |
| 16:14.19 | igcewieling | Greenlight: mildly interesting to me is Asterisk |
| 16:14.35 | Greenlight | I would advise against streaming it... it may be quite illegal |
| 16:14.42 | igcewieling | among other things, but those would be inappropriate here. 8-) |
| 16:14.45 | stanreg | I see. |
| 16:14.51 | igcewieling | DO IT! DO IT! |
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| 16:15.24 | stanreg | like I said, tons of noise on the line, and mostly asian sounding. |
| 16:15.40 | Greenlight | Just 1 phone call ? |
| 16:15.55 | stanreg | For the last 10 or so minutes, yes, 1 phone call. It was a bunch earlier. |
| 16:16.00 | igcewieling | I could do random ranblings about this customer complaing about paging not working on their PBX, but we don't provide their PBX. Then 2 weeks later their sales rep opens a ticket with the same problem. |
| 16:16.03 | [TK]D-Fender | Asian-sounding noise? |
| 16:16.15 | stanreg | [TK]D-Fender, asian-sounding language. |
| 16:16.15 | ChannelZ | Dogs being cooked, that sort of thing |
| 16:16.16 | [TK]D-Fender | I wonder if I can EQ that out... |
| 16:16.27 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:16.50 | igcewieling | both times someone assigns the ticket to me. |
| 16:17.00 | stanreg | Greenlight, csipsimple should be recording the conversation. I'll see if I can upload it when I finally hang up. |
| 16:17.29 | stanreg | seems dead for the last 2 minutes. yeah, think I'll hang up. |
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| 16:47.58 | dorphalsig | Greenlight: So basically what I should do is just ignore the status on the queue? |
| 16:48.26 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:48.51 | Greenlight | Umm... I didn't say that |
| 16:49.15 | jmetro | hm |
| 16:49.23 | jmetro | how do i make a fake blf |
| 16:49.59 | jmetro | i have a extensions that set the user to In Office / Out of Office and want them to light up , kind of like a BLF for parking |
| 16:50.12 | jmetro | so if their status is out, it lights up and turns the "in" off. |
| 16:50.17 | navaismo | dorphalsig, maybe your issue is related with the HINTS that FreePBX generate, are you adding the memebers as A<peernumber,0. I.e: A100,0? |
| 16:50.37 | navaismo | if so try to change to S<peernumber>,0--> S100,0 |
| 16:51.49 | jmetro | hints would do that right <.< |
| 16:52.21 | paulc | jmetro: hints and custom device states. I did exactly what you described for DND and call forwarding in the dialplan |
| 16:52.43 | jmetro | can you provide a well-commented sample ? :3 |
| 16:52.52 | [TK]D-Fender | jmetro: It's in the book |
| 16:53.18 | paulc | jmetro: The DND button is green when off, red when on.. works a treat.. and for call forwarding I wrote some dial plan/IVR magic to allow changing of destination with smooth prompting (which we had from another project). It's niiiiice. |
| 16:53.25 | [TK]D-Fender | jmetro: https://wiki.asterisk.org/wiki/display/AST/Function_DEVICE_STATE |
| 16:53.29 | paulc | jmetro: what phones are you using? |
| 16:53.37 | jmetro | aastras |
| 16:54.04 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 16:54.26 | jmetro | hm now im trying to think how to set the aastra to watch that device |
| 16:54.34 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:54.40 | [TK]D-Fender | jmetro: BLF.... |
| 16:54.50 | paulc | I've got Cisco SPA508Gs but the Aastras are good too.. should work the same way - the only question is "what happens when you press that BLF button?". The Cisco's dial the extension that's being watched and it's the same every time.. so you have to write dialplan to do the toggling |
| 16:56.09 | jmetro | well i would setup like... Softkey 6 = BLF to extension 501 maybe, and that extension would be defined to [in psuedocode] set 502 devicestate=not_inuse and 501 devicestate=inuse |
| 16:56.20 | jmetro | and vice versa. |
| 16:56.31 | paulc | Yeah, that's pretty much it. |
| 16:56.36 | paulc | let me find my code and I'll pastebin it for you |
| 16:56.47 | jmetro | but hmm. |
| 16:58.15 | jmetro | Aastra is setup as a BLF to "501", how would that devicestate custom work |
| 16:58.16 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 16:58.20 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 16:58.35 | jmetro | like set(device_state(custom:501)=inuse) ? |
| 16:58.55 | jmetro | and then exten=>501,hint,Custom:501 |
| 16:59.17 | paulc | jmetro: see http://pastebin.com/RRehXFfx |
| 16:59.20 | [TK]D-Fender | jmetro: "like", but not |
| 16:59.43 | paulc | In my example, you'd set the BLF to "DND501" |
| 16:59.48 | [TK]D-Fender | jmetro: Do not forget your variable and function syntax rules |
| 17:00.02 | paulc | But you could use anything you like really |
| 17:00.15 | jmetro | Ahh |
| 17:00.17 | jmetro | i like it. |
| 17:00.30 | jmetro | that way it doesnt interfere with real device state either. |
| 17:00.35 | paulc | Your custom device states shouldn't clash with dialplan extensions or actual devices |
| 17:00.39 | paulc | Yup, exactly |
| 17:00.57 | jmetro | This is for a client but i'm sure this would score huge points internally too.. Lol |
| 17:01.22 | paulc | so I'm using custom device states AND the asterisk internal DB.. Call Forwarding is pretty similar too.. so people can see at a glance if they'er forwarding or not |
| 17:01.48 | paulc | and we have a button for forwarding helpdesk to the on call person.. ALL helpdesk phones have the same button.. so everyone can see if it's forwarding or not. Works really well. |
| 17:03.35 | paulc | I keep meaning to make a video of how the two options work.. especially with the smooth prompting.. it's nice.. "Calls to your extension will forward to.. external number.. six oh four.. two five seven.. five seven.. five seven.." with inflection. Much nicer than "six.. zero.. four... two.. five.. seven.." etc robotic style |
| 17:03.57 | paulc | Of course, you need a ton of prompts with all the digits combinations.. but we had that from another product, so it was an easy win :-) |
| 17:10.40 | jmetro | I've got a DB as well |
| 17:10.41 | jmetro | from realtime |
| 17:10.51 | jmetro | asterisk has saydigits |
| 17:10.52 | jmetro | btw |
| 17:11.12 | *** join/#asterisk bitflag (~o3www@adsl-29.109.242.92.tellas.gr) |
| 17:11.24 | bitflag | hello |
| 17:11.43 | ChannelZ-Wk | OH HAI |
| 17:11.56 | bitflag | straight to the roast |
| 17:13.23 | [TK]D-Fender | Straight outta Compton |
| 17:13.24 | ChannelZ-Wk | Mmmmm roaaaassttt |
| 17:13.30 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 17:13.44 | bitflag | cisco2821 <---sip--->asterisk<---sip--->7960 |
| 17:13.56 | dorphalsig | Greenlight: I've already added hints for all the channels that are members of the queue |
| 17:14.42 | paulc | jmetro: yeah, I know about saydigits :-) But when you custom string prompts together, it's way smoother sounding.. I should stick a demo up somewhere so you can hear it |
| 17:14.58 | bitflag | initiate call from the outside world to 2821 |
| 17:16.20 | jmetro | I have that actually |
| 17:16.23 | bitflag | a)call to ivr and then to extension : sound choppy :( |
| 17:16.43 | jmetro | Our voice recording company charged per paragraph so I had them record one line of our DID prompt and then record all our names in one paragraph |
| 17:16.45 | bitflag | as soon as the extension picks it up |
| 17:17.18 | bitflag | b)call goes straight to the extension : sound perfect :) |
| 17:17.37 | bitflag | ??? |
| 17:17.40 | bitflag | wtf |
| 17:17.43 | bitflag | :P |
| 17:18.38 | ChannelZ-Wk | Are you using directmedia? (when you said 'call goes straight to the extension' does it literally do so?) |
| 17:18.42 | bitflag | it plays well if there is an sccp device on the extension or if we use a softphone |
| 17:19.32 | bitflag | straight means that the inbound route has as destination the extension |
| 17:19.55 | ChannelZ-Wk | And what if the 2821 calls the 7960's extension directly, no outside call involved |
| 17:21.21 | bitflag | what do you mean 2821 calls the extension? how that could happen? |
| 17:21.35 | paulc | jmetro: http://blahdomain.com/CFwd_Smooth_Prompting.mp3 <-- Smooth prompting in action |
| 17:22.00 | jmetro | oh wow, thats -smooooth- |
| 17:22.03 | ChannelZ-Wk | oh sorry that's an appliance |
| 17:22.04 | jmetro | for dynamic numbers |
| 17:22.11 | ChannelZ-Wk | nevermind |
| 17:22.42 | paulc | We've got all the 3 digit combos from 000 to 999 that cover NPA/NXX combinations.. then all the 2 digit combos 00 through 99 as a start inflection (up) and an end inflection (down) |
| 17:23.36 | paulc | "Your membership number is.. two three four.. five six.. Your password is.. seven eight.. nine zero.." - way more human, way less robotic.. a little bit of polish on the customer experience :) |
| 17:23.46 | ChannelZ-Wk | In any event the choppiness is something about the network, the trick is to narrow down why or where. If you make a Playback/Echo extension and call it from the 7960 is it bad? |
| 17:24.50 | ChannelZ-Wk | What codec is being used? Is the server underpowered? |
| 17:24.57 | bitflag | no the 7960 sounds perfect if speaks internal internal |
| 17:24.59 | jmetro | paulc: thats a lot of sound files. |
| 17:25.26 | ChannelZ-Wk | Is the choppiness bidirectional? |
| 17:26.21 | paulc | jmetro: Yup.. but a few hours invested in it gives for a way better customer experience, over and over again.. so we figured it was worth it :) |
| 17:26.22 | bitflag | no very sure but i think is one way... |
| 17:27.20 | bitflag | sorry for not being very accurate but i very dizzy with all the testing |
| 17:27.38 | jmetro | paulc: its a good idea if youre selling that as a platform |
| 17:28.06 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 17:28.51 | bitflag | the strange for my understanding is that if there is no ivr the call sounds perfect |
| 17:28.54 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:29.16 | paulc | jmetro: Yeah, no doubt.. be nice to contribute that back to the community but not sure the higher ups would go for it (or the guy that voiced the prompts maybe). We've got 2 or 3 female versions too. Just lucky to be able to use stuff from core business product when we did the new PBX internally I guess :-) |
| 17:30.34 | jmetro | paulc: If my company went pure VOIP i'm sure we would invest in setting that up as part of our "new client template" |
| 17:31.17 | jmetro | we've managed to template everything up to the point of, enter in MAC address, extension, and company name, and you can have a 30 phone company done in 1 or 2 hours |
| 17:32.37 | bitflag | outgoing is working perfect as well when we call fro 7960 to the outside world |
| 17:32.50 | igcewieling | jmetro: then 5 days trying to make Asterisk emulate a Key System. 8-| |
| 17:34.16 | paulc | Haha yes, people do love their key system working don't they |
| 17:34.22 | paulc | "Hank! Call for you on line 3!" |
| 17:34.43 | igcewieling | "To pick up line 3, dial 703" |
| 17:35.01 | jmetro | Hank, call for you on park 1 , press the one with the red light :3 |
| 17:35.43 | paulc | I love when you go to Safeway and it comes over the PA.. "Produce Department.. two.. zero.. one.." |
| 17:39.44 | *** join/#asterisk aruntomar (~Thunderbi@49.248.156.135) |
| 17:39.58 | *** join/#asterisk serafie (~erin@nat/digium/x-lrjzrrktcrjhsuhh) |
| 17:42.20 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:43.56 | j4jackj | Anyone here enthusiastic about Opus? |
| 17:44.55 | [TK]D-Fender | Nope |
| 17:45.12 | [TK]D-Fender | No real carries support it, no real hardphones support it. |
| 17:45.22 | jmetro | asterisk 12 does |
| 17:45.36 | [TK]D-Fender | It's "noise" left largely for users willing to chain their heads to their PC's |
| 17:46.22 | ChannelZ-Wk | I love Directory Opus |
| 17:48.47 | j4jackj | [TK]D-Fender: Cmon, Opus is my internet radio station's codec! What's not to love? |
| 17:49.06 | j4jackj | It sounds transparent at 96k, too. |
| 17:49.16 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 17:49.42 | [TK]D-Fender | j4jackj: Asterisk is not a "radio station broadcast tool" |
| 17:50.04 | ChannelZ-Wk | It's not a matter of it not being good, it's a matter of support |
| 17:50.28 | *** join/#asterisk slidesinger (~slidesing@c-69-141-78-33.hsd1.nj.comcast.net) |
| 17:50.43 | hjf | really strange problem: calls on LAN work, but calls to a phone in another LAN, doesn't work - there's no nat or firewall involved, hosts are 100% visible. there's a router between both networks |
| 17:50.56 | hjf | i call the phone and they can hear me |
| 17:51.05 | ChannelZ-Wk | It will be more useful in coming years with WebRTC type apps but other than that.. |
| 17:51.07 | hjf | but all i hear is fshhh... and then silence |
| 17:52.41 | ChannelZ-Wk | Ah. You are using G.Fssshh |
| 17:53.05 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.141) |
| 17:53.11 | j4jackj | [TK]D-Fender: This we know |
| 17:53.22 | j4jackj | [TK]D-Fender: I don't use Asterisk. |
| 17:53.26 | hjf | Remotely bridging SIP/hjf-00000006 and SIP/jorge-00000007 |
| 17:53.43 | [TK]D-Fender | hjf: that doesnt' say anything meaningful really. Go look at the actual complete debug |
| 17:53.53 | ChannelZ-Wk | Do some RTP debug and make sure you have packets going to/coming from the right places. Fsshhh sounds a bit like "comfort noise" some phones like to do, depending on how loud it is. Maybe a device has lost its mind |
| 17:54.07 | hjf | but i have directmedia=yes directrtpsetup=yes nat=no in both peers :/ |
| 17:54.25 | hjf | and localnet=10.0.0.0/255.0.0.0 |
| 17:54.41 | hjf | are those settings supposed to allow direct communication between these two peers? |
| 17:55.06 | j4jackj | ChannelZ-Wk: what is that standard again? |
| 17:55.15 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:55.45 | ChannelZ-Wk | assuming the codecs match and Asterisk isn't recording or otherwise having to be in the media path. Perhaps the reinvites aren't working like you think |
| 17:56.07 | hjf | hm yea |
| 17:56.09 | jmetro | is out to lunch |
| 17:56.12 | ChannelZ-Wk | j4jackj: G.Fssshh? |
| 17:57.12 | *** join/#asterisk serafie (~erin@nat/digium/x-bzgxagrqegcgaoqv) |
| 17:59.21 | Penguin | Keep asterisk in the path and see if anything changes. |
| 17:59.50 | hjf | call trace http://pastebin.com/qY3PRP3U |
| 18:01.27 | Penguin | Also, what is hjf and what is jorge? Are they both asterisk? Is there a phone involved anywhere? |
| 18:02.25 | hjf | both are android (CSipSimple) |
| 18:02.57 | Penguin | Are they using the same asterisk? |
| 18:03.17 | hjf | yes both registered to the same server |
| 18:03.30 | Penguin | Are they both on the same LAN with asterisk? |
| 18:03.39 | j4jackj | ChannelZ-Wk: yes |
| 18:03.49 | hjf | no. there is lan A <--> ROUTER <--> lan B |
| 18:03.53 | hjf | lan a is 10.42.42.0/24 |
| 18:03.57 | hjf | lan b is 10.23.23.0/24 |
| 18:04.02 | hjf | there is no NAT between lans |
| 18:04.09 | Penguin | Is the public internet in between? |
| 18:04.13 | hjf | no |
| 18:04.54 | Penguin | Okay, so it's just two segments with a router between them. |
| 18:05.13 | hjf | yes |
| 18:05.38 | Penguin | Which segment is asterisk in? |
| 18:05.42 | hjf | 10.42 |
| 18:06.13 | hjf | ping with up to 1472 bytes works |
| 18:06.24 | hjf | mtu? |
| 18:07.08 | ChannelZ-Wk | Everything looks about right but it seems to be a problem between the phones communicating directly with each other. |
| 18:07.20 | ChannelZ-Wk | Turn off directmedia and see if you get different results |
| 18:07.48 | Penguin | I was still waiting on the results of that from nearly ten minutes ago. |
| 18:07.56 | Katty | ARE YOU OPEN ON THURSDAYS?!?! |
| 18:07.57 | j4jackj | ChannelZ-Wk: what is G.Fsh? |
| 18:08.09 | Penguin | You haven't heard of that codec? |
| 18:08.23 | ChannelZ-Wk | That was what we call in the industry "a joke" |
| 18:08.58 | hjf | ChannelZ-Wk: k i will try later i can't bother that extension anymore hah |
| 18:09.09 | [TK]D-Fender | [13:50]hjfbut all i hear is fshhh... and then silence [13:52]ChannelZ-WkAh. You are using G.Fssshh |
| 18:09.11 | Penguin | We don't care about the extension. We only care about the phones. |
| 18:10.51 | Katty | [TK]D-Fender: how's that clavical doing today? |
| 18:12.39 | j4jackj | ChannelZ-Wk: they should actually standardise G.792 as 'Low bitrate silence signaling for telephony applications' |
| 18:13.23 | *** join/#asterisk ttpearso (~ttpearso@gw.teamgleim.com) |
| 18:13.49 | [TK]D-Fender | Katty: SSDD. got chest/back load balancing stress for my lack of use o9f it which is annoying and sometimes bad... should start physio in another 2 weeks... |
| 18:14.09 | Penguin | I don't know if I've ever heard of g792. |
| 18:14.27 | Katty | nods |
| 18:14.37 | Katty | [TK]D-Fender: 2 weeks and that knot in your back will even out some |
| 18:14.40 | drmessano | It's G711 + G722 + G729 + ILBC |
| 18:14.45 | Katty | hi danny |
| 18:14.47 | drmessano | All on one channel |
| 18:15.01 | drmessano | Hello Katwoman |
| 18:15.11 | drmessano | How art thou? |
| 18:15.20 | Katty | good good. how's your wife doing? |
| 18:15.25 | j4jackj | Nope, that would be G811, without G729 and instead using Opus for that layer. |
| 18:15.34 | jmetro | Is it g722 quality with g711 bandwidth? |
| 18:15.37 | drmessano | She is quite fine. On the mend, as one would expect |
| 18:15.44 | Katty | nods |
| 18:15.45 | Katty | good! |
| 18:16.01 | drmessano | Oh crap, sorry, change my language preferences to en_GB |
| 18:16.11 | drmessano | Back to en_US now |
| 18:16.15 | [TK]D-Fender | [14:15]jmetroIs it g722 quality with g711 bandwidth? <- those are BOTH 64 kbps |
| 18:16.17 | drmessano | Yeah, she's koo, bro |
| 18:17.14 | ttpearso | Afternoon all, having an issue when reloading dahdi modules in asterisk (not restarts etc), all channels go to alarm and calls are dropped |
| 18:17.20 | Katty | it's cool, i speak en_GB and en_US ;) |
| 18:17.31 | [TK]D-Fender | I speak in tongues.... |
| 18:17.35 | [TK]D-Fender | I'm a cunning linguist ;) |
| 18:17.39 | Katty | and en_AU too |
| 18:17.42 | j4jackj | What about, using G722 to code Ulaw in 32kbps? |
| 18:17.53 | drmessano | Katty, I speak both as well, but my dunny closet and toilet get crossed, as well as the whole electric torch thing |
| 18:18.03 | Katty | grins |
| 18:18.45 | Katty | i also speak alabama ;) |
| 18:18.52 | drmessano | If I am using my phone as a flashlight, is that a "Blower Torch"? |
| 18:19.02 | Katty | not that i will mention names, of others in the channel speaking alabama. |
| 18:19.04 | j4jackj | Nooooooooooooooooo |
| 18:20.02 | Katty | drmessano: just don't forget....never leave the bubble and squeak alone while going to the lou |
| 18:20.10 | drmessano | If I use an electric cigarette as a light, is that a "Fag Torch"? |
| 18:20.17 | drmessano | These things MUST be addressed |
| 18:20.34 | Katty | in the lou. |
| 18:20.40 | drmessano | in the butt |
| 18:20.45 | Katty | what what?! |
| 18:21.19 | Katty | so i started my hipaa certification today. |
| 18:21.24 | drmessano | I had fun last night uploading all my goofy audio stuff to my personal hosted PBX |
| 18:21.27 | Katty | it's not nearly as dry as my other certs, which is nice. |
| 18:21.32 | jmetro | Isnt it Loo? |
| 18:21.39 | Katty | jmetro: probably. |
| 18:21.43 | Penguin | I think so. |
| 18:21.47 | drmessano | Like my TMDestroyer, Leekspin, Not the next caller, etc |
| 18:21.47 | Katty | i'm sure there's someone we can ask. |
| 18:21.53 | Penguin | The Lou would be St. Louis. |
| 18:21.57 | drmessano | I believe it's Loo as well |
| 18:22.02 | Katty | no one calls st. louis the lou |
| 18:22.08 | drmessano | I believe as well that it's Loo |
| 18:22.08 | Katty | well. i guess if you're a tourist maybe. |
| 18:22.09 | drmessano | Sorry |
| 18:22.10 | Penguin | They used to. |
| 18:22.21 | jmetro | HIPPA is interesting in how many ways it applies |
| 18:22.23 | Katty | none of my stl friends do. |
| 18:22.25 | jmetro | and what could violate HIPAA |
| 18:22.31 | Katty | yup. |
| 18:22.40 | Penguin | Back before they started calling it STL, it was The Lou. |
| 18:22.42 | Katty | jmetro: it is absolutely insane, but i've just started. don't know a whole lot just yet |
| 18:22.59 | jmetro | like in a school, if someone has a deadly peanut allergy, do you distribute that information? |
| 18:23.01 | Katty | criminal penalties and fines, mostly |
| 18:23.09 | drmessano | jmetro, if I am discussing HIPAA as it applied to Obamacare, could that be considered HIPAA POTUS? |
| 18:23.22 | Katty | you're such a clever boy, drmessano |
| 18:23.33 | drmessano | Katty, indeed |
| 18:23.58 | Katty | iiii want a hipaapotus for christmassss |
| 18:24.23 | drmessano | Katty, i'm just a geek that has what all the ladies want. Oh wait, that's kinda cool actually |
| 18:24.35 | Katty | pats drmessano |
| 18:24.45 | jmetro | sidenote: taco bell food is merely a conveyor belt for Lava Sauce. |
| 18:24.51 | Katty | yes, you're quite lovely dear. now don't be an arrogant ass. |
| 18:25.18 | Katty | jmetro: lava sauce? |
| 18:25.27 | Katty | jmetro: is that the spicey cheese sauce thing on the volcano tacos? |
| 18:25.32 | jmetro | Yes |
| 18:25.38 | jmetro | you can get it as a side for 25 cents |
| 18:25.38 | Katty | it's good :> |
| 18:25.52 | Katty | takes notes |
| 18:25.56 | jmetro | so basically, get 3 tacos, side of lava sauce, and you just had the best meal of your life. |
| 18:26.08 | Katty | i bet there's a ton of calories :/ |
| 18:26.14 | Katty | shakes fist |
| 18:26.18 | jmetro | maybe... |
| 18:26.32 | Katty | i should get to be scrawny AND eat processed crap food!!! |
| 18:26.47 | jmetro | 1 oz of lava sauce is probably the size of the 25 cent cup |
| 18:26.47 | Katty | Dear Taco Bell... plz to remove calories from lava sauce. kthx, love, Katty |
| 18:27.00 | jmetro | 140 calories in the whole cup, 14g of fat. |
| 18:27.07 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:27.12 | Katty | googles |
| 18:27.15 | Katty | 1 oz is 140 calories |
| 18:27.20 | Katty | oh right. |
| 18:27.27 | Katty | thanks for googling that for me. |
| 18:27.49 | Katty | i like how the taco is 150 (or 170ish?) and the sauce is nearly as much as the taco. |
| 18:28.01 | drmessano | Katty, i've given up arrogance. I've taken up "Constructive Egomaniacal Insightfulnessity". It might seem like the same thing but.. |
| 18:28.29 | Katty | but... i glare at you either way? |
| 18:29.08 | drmessano | Katty, yes, you've earned it |
| 18:29.37 | Katty | good thing you don't require regular glaring from me. |
| 18:29.43 | jmetro | You put far less of 1 oz on a taco though |
| 18:29.45 | jmetro | 1 oz might be 3 tacos |
| 18:29.55 | j4jackj | Katty: good thing you don't from me either |
| 18:29.59 | Katty | jmetro: unfortunately, i only consume about 350-400 per meal |
| 18:30.14 | drmessano | Katty, I probably do. I tried to train the wife to glare at me, but she's really not good at the whole "evil eye" thing |
| 18:30.36 | Katty | jmetro: so that kind of narrows the processed foods down. i can either eat real food and not be hungry, or i can eat taco bell and be hungry 2 hours later |
| 18:30.42 | Penguin | That would indicate to me a target weight of just 80 pounds. |
| 18:30.43 | drmessano | I wonder if the evil eye can be tweeted |
| 18:30.58 | Penguin | That is based on three 400 calorie meals and no snacks, though. |
| 18:31.13 | drmessano | I can't eat Taco Bell anymore. I am uber jealous of this convo |
| 18:32.45 | Katty | Penguin: i eat 4 times a day |
| 18:33.06 | Penguin | target of 107 is a little more realistic. |
| 18:33.12 | Penguin | Still seems pretty low. |
| 18:33.14 | Katty | target is actually 120 |
| 18:33.33 | Penguin | You can have 1800 calories a day. |
| 18:33.39 | Katty | probably. |
| 18:33.47 | drmessano | Katty, is there exercise involved or purely dietary modification? |
| 18:34.00 | Katty | drmessano: i know you've seen my marathon posts |
| 18:34.01 | Penguin | You just earned a bonus snack! |
| 18:34.11 | Katty | drmessano: ;) ;) |
| 18:34.22 | Katty | drmessano: i run twice a week, weather permitting. and lift weights twice a week as well |
| 18:34.44 | drmessano | Katty, i'm fairly certain I have not. I think Facebook's algorith has you in a spot where I don't see your posts in my feed |
| 18:34.46 | Katty | health permitting. |
| 18:34.58 | drmessano | I need to move you to close friends or something |
| 18:35.05 | drmessano | algorithm |
| 18:35.11 | Katty | idk, there's not much interesting i post. |
| 18:35.20 | Katty | endomodo reports. pictures of food |
| 18:35.24 | Katty | endomondo |
| 18:35.35 | Katty | maybe the cat. it's pretty boring really. |
| 18:36.09 | drmessano | I've been working out 4 times a week.. 30 mins of cardio, 45 mins of weight. My goal was "weight loss", but i've started ignoring the scale.. it's useless with the muscle i've packed on |
| 18:36.10 | *** join/#asterisk nightrid3r (~kvirc@62.205.64.154) |
| 18:36.25 | Katty | yup |
| 18:36.41 | Katty | i check the scale maybe a couple times a month |
| 18:37.03 | Katty | to confirm it's actually going down. not sure why i bother tho |
| 18:37.09 | drmessano | I think i've gained 5 pounds, after being pretty much neutral for 6 weeks or so |
| 18:37.15 | Katty | it's been going down consistently, at the same speed, for the last 12 months |
| 18:37.22 | drmessano | Thats awesome |
| 18:37.23 | drmessano | !! |
| 18:37.26 | jmetro | weight is different than fat though |
| 18:37.39 | jmetro | if i work out, i gain weight, i cant sem to burn the bit of fat i've got stuck on |
| 18:38.13 | Katty | drmessano: https://dl.dropboxusercontent.com/u/25951554/chart.png <- 1 year of data collection |
| 18:39.07 | Katty | drmessano: you can see it's pretty steady. |
| 18:39.13 | Katty | drmessano: based on averages. |
| 18:40.29 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 18:40.37 | Katty | smiles at igcewieling |
| 18:40.53 | Katty | igcewieling: how ya feeling today? |
| 18:41.06 | igcewieling | Katty: don't ask |
| 18:41.20 | Katty | nods and hugs igcewieling |
| 18:45.16 | drmessano | Katty, are you tracking this using Asterisk and storing the values in ASTDB or MySQL? |
| 18:46.36 | jmetro | call in to DID, dial your DTMF for calories |
| 18:47.13 | jmetro | iftime 1000-1400 its lunch, before is breakfast, after is dinner. |
| 18:47.28 | drmessano | I love using Bria on my phone for this.. I |
| 18:47.51 | drmessano | I love using Bria on my phone for this.. I've built up a lot of automation functions in the dialplan |
| 18:48.03 | drmessano | X10 stuff, tweeting, etc |
| 18:48.35 | jmetro | cat facts? |
| 18:49.17 | drmessano | Well, no.. and I realize that the modern "Hello, World" must involve some sort of cat related function |
| 18:49.22 | drmessano | So I fail miserably there |
| 18:49.25 | Katty | drmessano: lol, no. just an old fashioned google drive spreadsheet |
| 18:49.32 | jmetro | http://i1.kym-cdn.com/photos/images/newsfeed/000/241/756/e6e.png |
| 18:49.43 | drmessano | Maybe I can turn my Asterisk 8-Ball into a Grumpy Cat hate generator |
| 18:50.00 | Katty | drmessano: i carry a little notebook in my purse, which i usually fill out at dinner time regarding what i ate that day |
| 18:50.00 | drmessano | "As I see it, NO" |
| 18:50.04 | drmessano | "My reply is NO" |
| 18:50.11 | drmessano | "Better not tell you EVER" |
| 18:50.31 | drmessano | "Reply Hazy, Consider it a NO" |
| 18:51.03 | drmessano | OMG.. I could seriously Grumpy Cat all 20 8-Ball responses |
| 18:51.11 | drmessano | What have I been doing with my life until now |
| 18:51.51 | Katty | daydream about killing doctors? |
| 18:52.03 | *** join/#asterisk smirker (~x@101.162.79.153) |
| 18:54.13 | *** join/#asterisk bitflag (~o3www@adsl-249.46.190.3.tellas.gr) |
| 18:54.21 | igcewieling | Katty: that is what I'm starting to daydream about! |
| 18:54.33 | bitflag | bb ppl |
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| 19:57.15 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:57.24 | hjf | if i use for example an SPA3000 as a gateway to my POTS line, does it pick up the call when it rings, or will it only pick up when a sip phone picks up? |
| 19:59.33 | [TK]D-Fender | hjf: Depends how you configure it |
| 20:15.21 | igcewieling | From an RFO from XO/Verizon Outage Start Time: August 21, 2013 10:14 AM Resolution Time: August 26, 2013 8:36 PM |
| 20:20.55 | *** join/#asterisk BriGuy (~brian@74.115.41.6) |
| 20:23.52 | ttpearso | I'm having an issue when reloading dahdi modules in asterisk (not restarts etc), all channels go to alarm and calls are dropped, if anyone could take a look at http://pastebin.com/QnBGckRN I'd appreciate any help |
| 20:25.14 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 20:25.52 | navaismo | since you are in #asterisk i must say dont use reload, if you want to commit a chinge for a sip use: sip reload, but i guess this is the stuff comming from #freepbx |
| 20:26.55 | [TK]D-Fender | checkout time, BBIAB |
| 20:27.09 | igcewieling | ttpearso: you are doing a "module reload chan_dahdi" ? |
| 20:28.46 | ttpearso | igcewieling: correct |
| 20:29.01 | igcewieling | ttpearso: that should NOT drop calls. |
| 20:29.15 | ttpearso | igcewieling: that's why I'm here :-) |
| 20:29.42 | igcewieling | ttpearso: what version of Asterisk, what version of DAHDI, what card model? |
| 20:30.28 | ttpearso | Asterisk: 11.4.0 |
| 20:30.50 | ttpearso | DAHDI: 2.7.0 |
| 20:31.13 | igcewieling | ttpearso: downgrade your dahdi to 2.6 and see what happens. |
| 20:32.00 | ttpearso | igcewieling: will do. Has 2.7 has other issues? |
| 20:32.07 | ttpearso | *had |
| 20:32.28 | ttpearso | It's a Digium TE122 btw |
| 20:33.14 | igcewieling | ttpearso: I can't point to actual documented issues, but I get bad vibes from 2.7. you could try 2.7.0.1 first. |
| 20:34.53 | ttpearso | We previously ran 2.4.1 without issue, any reason I shouldn't start there? eg: does Asterisk 11 care? |
| 20:36.03 | igcewieling | ttpearso: since you should NOT be having this issue, most anything I recommend is speculation. If it was me I'd downgrade to the latest 2.6.x dahdi. I doubt the asterisk version matters. |
| 20:37.59 | LonghornWork | i thought module reloads to dahdi does drop calls since it reinitilizes the channels...given ive only worked on older dahdi versions :) |
| 20:39.21 | ttpearso | igcewieling: understand the speculation, but I'm in no position to suggest other fixes, I'll do that tonight |
| 20:39.47 | igcewieling | LonghornWork: it never did. dahdi_cfg does drop calls, but that is dealing with the kernel drivers, not Asterisk |
| 20:40.24 | ttpearso | The only thing I was thinking is maybe the card is going south - I've got a handful of identical cards I can test with, but since it's making/receiving calls, I'd prefer to start with software |
| 20:40.48 | LonghornWork | igcewieling: ahh, hmm i would have to test, but if it doesn't that would def be nice |
| 20:41.55 | igcewieling | ttpearso: HDLC abort messages can be thought of as the D-channel equiv of a crackle in the audio stream. |
| 20:49.09 | *** join/#asterisk bitflag (~o3www@adsl-34.79.107.162.tellas.gr) |
| 20:49.22 | bitflag | HELLo again :) |
| 20:50.06 | bitflag | i am running Asterisk 1.8.22.0... |
| 20:50.25 | bitflag | Created by Mark Spencer :p |
| 20:51.15 | bitflag | well i am wondering if the internal_timing = yes is applying to my version... |
| 20:51.44 | igcewieling | bitflag: see asterisk.conf.sample in the Asterisk source code. |
| 20:51.55 | bitflag | i don not have a timing source configured... do i need to do so or that is build into the core? |
| 20:52.41 | bitflag | actually is the freepbx distro |
| 20:52.44 | igcewieling | bitflag: "module show like tim" will show you if/what timing modules are loaded. if you have an active call the one in use will have a non-zero use count |
| 20:52.59 | igcewieling | bitflag: I wish you the best of luck. |
| 20:53.03 | bitflag | :D |
| 20:54.08 | bitflag | i have this load from what i understand >>> res_timing_timerfd.so Timerfd Timing Interface 1 |
| 20:54.42 | bitflag | is this working or is crapy? |
| 20:55.17 | bitflag | rtp disaster |
| 20:55.34 | igcewieling | we use timerfd |
| 20:55.47 | jmetro | RTP disaster sounds like two RTP streams colliding and exploding the internet. |
| 20:56.00 | igcewieling | res_timing_timerfd.so Timerfd Timing Interface 118 |
| 20:56.39 | bitflag | why yours is 118? what the number means? |
| 20:57.01 | bitflag | i have to dig more ...hmm |
| 20:57.14 | igcewieling | that is the number of calls using that timer |
| 20:57.26 | igcewieling | actually the number of channels I thingk |
| 20:57.44 | bitflag | ok you scared me |
| 20:57.47 | bitflag | :) |
| 20:57.59 | bitflag | because i have no calls |
| 20:59.47 | bitflag | what is your machine igcewieling? |
| 21:01.04 | igcewieling | t is a box on the end of an SSH connection in our NOC in Manhattan. Never actually seen it, think 16-core 16-gb |
| 21:01.29 | bitflag | 16-core sounds very good |
| 21:01.46 | bitflag | i have 16-gb as well |
| 21:02.32 | bitflag | the one i am trying is in egypt |
| 21:04.32 | jmetro | my gaming pc is an 8 core, wut wut |
| 21:04.36 | bitflag | cpuifo shows 1 X 4 core xeon @ 3.00GHz |
| 21:04.47 | jmetro | real 8 cores, not hyper-threading garbage |
| 21:06.31 | bitflag | could it a hardware prob my choppy sound, something like but timing? |
| 21:06.58 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 21:07.07 | bitflag | i am so lost ... |
| 21:07.52 | *** join/#asterisk luce (~luce@NYUFWA-NYUSHANET-01.NATPOOL.NYU.EDU) |
| 21:08.12 | bitflag | anyway.. thank you for the info :) |
| 21:09.06 | luce | hello all. I'm having issues using the chan_alsa.so module - i get an error when asterisk starts saying "expected rate 8000 but received 16000" - any thoughts/ |
| 21:11.43 | luce | does anyone have experience using ALSA with asterisk? |
| 21:13.51 | ChannelZ-Wk | from the alsa module I assume? |
| 21:14.22 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 21:15.21 | ChannelZ-Wk | dunno if your alsa config is required to be set to 8khz or what.. |
| 21:18.41 | luce | can i just edit the chan_alsa.so file? |
| 21:19.40 | luce | is there a way to configure alsa sample rate? |
| 21:22.46 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 21:29.34 | luce | Also, how do I connect or answer a call coming into asterisk directly to a usb headset or sound card? |
| 21:35.52 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 21:36.45 | kuku | I'm using Application: Bridge and in that diaplan I use Playback(), but the sound plays only on the side of the asterisk extension, not on the call recepient... |
| 21:42.57 | drmessano | Sounds like there is a crack in your bridge |
| 21:46.50 | kuku | :) |
| 21:46.52 | kuku | good one |
| 21:48.13 | luce | has anyone uses the chan_alsa.so module before? |
| 21:48.28 | navaismo | ~ask |
| 21:48.28 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 21:51.29 | luce | Is there a way to configure the sample rate in the chan_alsa.so? |
| 21:53.51 | *** join/#asterisk btcquant (~noahnoah@99-41-172-179.lightspeed.irvnca.sbcglobal.net) |
| 21:55.40 | btcquant | Hi - quick question about voicemail. I have a "messages" button on my polycom phone that calls voicemail, but I then need to enter mailbox and password every time. Is there a way to program a shortcut so that it just go directly into my box? |
| 21:56.16 | jmetro | i think you could "senddigits" but im not sure that works on itself. |
| 21:56.41 | btcquant | I know that I can set the phone "macro" to dial anything I want with that button. I just don't know what. |
| 21:56.57 | roderickm | btcquant: See the s option on VoiceMailMain https://wiki.asterisk.org/wiki/display/AST/VoiceMailMain+Application |
| 21:57.09 | btcquant | roderickm Awesome. Thanks!! |
| 21:57.22 | WIMPy | luce: Your hardware (or its driver) doesn't support the required samplerate. |
| 21:58.12 | WIMPy | luce: You can work around that issue by using dmix/(whatever the reverse direction was called). |
| 22:01.15 | luce | Thanks for the help. the alsa driver (i believe) is expecting 8000… while the logitech usb is sending 16k…hence the error "Rate not correct, requested 8000, got 16000"… what is dmix and reverse direction? |
| 22:01.48 | WIMPy | The ALSA internal software mixers. |
| 22:02.46 | WIMPy | I haven't tried the input part of it (which name just escapes me), but I think it should work just like dmix does for output. |
| 22:03.54 | luce | Thanks. I'll look into it |
| 22:08.23 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 22:08.27 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 22:11.26 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 22:16.10 | WIMPy | BTW: Would be nice if someone would give chan_alsa some TLC. We have CODECs that can make use of more than 8000s/s. |
| 22:21.27 | jmetro | i read that as 8000 seconds per second. |
| 22:21.46 | WIMPy | samples |
| 22:22.05 | eirirs | haha |
| 22:22.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:24.34 | gusto | hi |
| 22:24.40 | gusto | how are you WIMPy ? |
| 22:26.24 | WIMPy | smells some food. That should make things better :-) |
| 22:26.38 | jmetro | paulc: just added a bulk import tool for spreadshoots. AWW YEAHHH |
| 22:31.37 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 22:36.45 | igcewieling | Real men use hardphones. |
| 22:41.25 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 22:49.48 | ChannelZ-Wk | Spreadshoots? |
| 22:50.00 | ChannelZ-Wk | Like shotguns? |
| 22:50.39 | newtonr | like the gun in Contra, the one that fires three directions |
| 22:52.32 | newtonr | </old NES game reference> |
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| 23:09.10 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 23:15.45 | paulc | jmetro: haha nice.. gotta love spreadsheet importing eh |
| 23:16.10 | jmetro | Got my statuses working too |
| 23:16.24 | jmetro | i took a page out of your book and had it call my already existing *51 and *52 macros that set the status properly in the DB |
| 23:16.48 | jmetro | got it working as a 1 button with decision tree, also working as a 2 button static "set me in" or "set me out" |
| 23:17.07 | jmetro | aastras only do red light / off so the 2 button makes sense |
| 23:17.18 | jmetro | red/green would rock for one button though |
| 23:24.41 | [TK]D-Fender | jmetro: you can set for ringing as well |
| 23:25.04 | jmetro | On an aastra? is it just flashing red? |
| 23:25.27 | [TK]D-Fender | IIRC yes, not sure what other modes it supports. |
| 23:25.34 | [TK]D-Fender | There is one for "offline" as well IIRC |
| 23:25.56 | jmetro | flashing red would be quite annoying... but might be very useful for "after hours" so its flashing to remind the client to set themselves in office |
| 23:26.20 | jmetro | anyway, home i go |
| 23:27.10 | ChannelZ-Wk | BLF! Blinky Light Function! |
| 23:57.12 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 23:58.00 | ChannelZ-Wk | Meh. So I just went and did the free basic * certification thing on digium.com for fun. I passed, but it doesn't show by how much or show any answers I got wrong. |
| 23:58.36 | WIMPy | I got told how many % I got right. |
