| 00:00.35 | Ice_Strike | Yep |
| 00:01.03 | Ice_Strike | I was reading on StackOverFlow - they are saying PHP is bad choice to create a daemon |
| 00:01.46 | WIMPy | always thought PHP was a bad choice for everything. |
| 00:02.12 | Ice_Strike | Hah |
| 00:03.57 | Ice_Strike | Whats new WIMPy |
| 00:04.48 | WIMPy | Worldwide organised suicide progessing well within timeframe. |
| 00:05.12 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 00:05.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 00:05.25 | Ice_Strike | Hmmmm |
| 00:15.44 | *** join/#asterisk TimeRider (~steve@027bde06.bb.sky.com) |
| 00:22.39 | igcewieling | Ice_Strike: what is the link to the stackoverflow page? |
| 00:26.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 00:32.03 | Ice_Strike | igcewieling http://stackoverflow.com/questions/646928/is-it-wise-to-use-php-for-a-daemon/647801 |
| 00:32.17 | igcewieling | Ice_Strike thanks |
| 00:32.47 | Ice_Strike | igcewieling Do you use PHP? |
| 00:35.16 | igcewieling | Ice_Strike: yes. |
| 00:35.56 | igcewieling | It may not be the perfect tool for anything, but it is plenty "good enough" for almost anything I do. |
| 00:36.58 | Ice_Strike | igcewieling Oh have you created a daemon in PHP for AMI use? |
| 00:37.26 | igcewieling | Ice_Strike: yes. |
| 00:38.25 | igcewieling | PHP's forking and process control is most or less a thin wrapper around the C functions, so it is most helpful if you already know how to write daemons in C. |
| 00:38.29 | Ice_Strike | igcewieling Oh thats great - no issue with memory usage? or anything creashed. |
| 00:38.43 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 00:38.52 | igcewieling | Ice_Strike: my daemons seldom have to run for more than 24 hours. |
| 00:39.46 | Ice_Strike | number of concurrents call went through daemons you have targeted so far? |
| 00:39.48 | igcewieling | keep this in mind, no matter what language you write a daemon in, you should expect it to fail occasionally and design accordingly. |
| 00:39.58 | Ice_Strike | Yep ofcourse |
| 00:40.04 | igcewieling | Ice_Strike: I do not understand the question |
| 00:42.18 | Ice_Strike | Sorry, What the highest concurrents calls that connected to a daemon? |
| 00:43.36 | igcewieling | I do not connect calls to daemons |
| 00:43.57 | igcewieling | http://pastebin.ca/2316948 an example, though it doesn't fork, so might not be considered a "daemon" |
| 00:44.56 | Ice_Strike | Nice one |
| 00:45.37 | igcewieling | another daemon I wrote listened for UDP packets and resent the packet to a set list of hosts |
| 00:46.03 | Ice_Strike | Do you have a daemon that listen the call event, hang up, orginating call, etc? |
| 00:46.25 | igcewieling | Ice_Strike: no. that is complicated and time consuming to write |
| 00:46.40 | Ice_Strike | Yep |
| 00:46.49 | WIMPy | There's a lot of stuff you can listen for. |
| 00:47.35 | igcewieling | most of the "long running" php processes I write are not really daemons, they are just long running PHP processes used to monitor stuff from a web page via AJAX |
| 00:48.12 | Ice_Strike | Cool stuff! |
| 00:48.36 | igcewieling | actually, annoying, bug ridden, and overly complicated |
| 00:55.57 | artyx | I'm having a wierd issue, i am trying to access the voicemail from dahdi channel. i have a chan_dahdi.conf with a [151] and under that one of the options is mailbox=151@default ... when checking voicemail i get this in the cli mailbox=151@default |
| 00:56.24 | artyx | oops. [2013-02-23 23:18:18] ERROR[11546][C-00000027] app_voicemail.c: MAILBOX_EXISTS requires an argument (<mailbox>[@<context>]) |
| 00:57.10 | artyx | *97 results in it going to the comedian mail login prompt, specifying a user extension .. meaning it isn't reading this as valid, right? |
| 00:57.33 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 01:06.11 | igcewieling | artyx: show us the dialplan and the resulting CLI output on pastebin |
| 01:09.21 | artyx | igcewieling Your not goign to liek it, but cli output for the voicemail check is at http://pastebin.com/eGQGTVkq |
| 01:11.06 | artyx | http://pastebin.com/PVZXY9yi (dialplan for *97) |
| 01:11.11 | igcewieling | I'll like it better when I can see the dialplan, though this does look like a freepbx box. Remember god kills a kitten every time you ask a freepbx question on #asterisk |
| 01:11.41 | *** join/#asterisk taylorbyte2013 (~cyberninj@2001:470:1f05:12cc:1e6f:65ff:fec2:bec7) |
| 01:12.21 | igcewieling | Your VMCONTEXT=default but the mailbox is in the "device" vm context? |
| 01:12.36 | artyx | ah, i think your onto something |
| 01:12.37 | igcewieling | looks to be a freepbx issue. |
| 01:12.48 | igcewieling | I wish you the best of luck. |
| 01:13.50 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:13.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:59.57 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.100) |
| 02:09.07 | *** join/#asterisk Kyosh (~whoa@pool-72-89-93-13.nycmny.fios.verizon.net) |
| 02:09.57 | Kyosh | Are there any AMI events to tell you when a user SIP registration failed? if not, other than the CLI and log file, is there any way in real-time to see the SIP registration fail? Thanks. |
| 02:18.57 | DoYouKnow | [TK]D-Fender: can you take a look at a couple configs for google talk/jabber? |
| 02:19.04 | DoYouKnow | http://pastebin.com/WkwPJUPK |
| 02:19.06 | DoYouKnow | that's one |
| 02:19.12 | eirirs | http://web.archive.org/web/19961027001602/http://www.microsoft.com/IE/ |
| 02:19.14 | eirirs | that's other |
| 02:19.40 | DoYouKnow | lol |
| 02:19.42 | DoYouKnow | err |
| 02:19.44 | DoYouKnow | that's not a config |
| 02:19.47 | DoYouKnow | I meant a log |
| 02:19.56 | DoYouKnow | well, that's the main thing |
| 02:34.52 | *** join/#asterisk mintos (~mvaliyav@114.143.40.54) |
| 02:36.15 | *** join/#asterisk dolcea-xoom (~androirc@181.31.66.193) |
| 02:37.08 | *** part/#asterisk dolcea-xoom (~androirc@181.31.66.193) |
| 02:38.10 | igcewieling | Kyosh: see sip.conf.sample in the asterisk source dir, there is an option to generate a manager event on registration failure |
| 02:45.10 | *** join/#asterisk ruben231 (~OpenDial@112.198.79.22) |
| 02:54.05 | DoYouKnow | why is gtalk not talking to jabber and in vim type= is bolded red? |
| 03:20.15 | DoYouKnow | obviously I made some sort of error |
| 03:23.10 | DoYouKnow | well, I'm going to update my asterisk |
| 03:23.17 | DoYouKnow | I was using an old version |
| 03:25.03 | DoYouKnow | it may or may not help |
| 03:25.11 | DoYouKnow | well, it will help |
| 03:25.12 | DoYouKnow | I suppose |
| 03:26.12 | *** join/#asterisk edong23 (~quassel@mptc-dhcp-50-220.mptelco.com) |
| 03:29.30 | *** join/#asterisk coreyf1513 (~cfarrell@ws2.cfware.com) |
| 03:53.30 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 04:16.41 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:38.12 | *** join/#asterisk radic (~radic@dslb-094-216-239-124.pools.arcor-ip.net) |
| 04:42.00 | *** part/#asterisk fulcan (brads@2600:3c00::f03c:91ff:fe70:f0a9) |
| 04:48.02 | *** join/#asterisk phunguy (santas@will.one.day.hack-the-pla.net) |
| 04:53.27 | *** join/#asterisk DoYouKnow (~michael@75-28-180-206.lightspeed.bnsvil.sbcglobal.net) |
| 04:53.36 | DoYouKnow | hi. I am hearing one ring on asterisk then it hangs up |
| 04:54.17 | DoYouKnow | logs: http://pastebin.com/sFkt63Px |
| 04:55.25 | DoYouKnow | <PROTECTED> |
| 05:04.07 | Kyosh | igcewieling: i will look for that, thanks. |
| 05:09.49 | edong23 | DoYouKnow: try setting your debug and see if it gives you any more information |
| 05:20.36 | DoYouKnow | edong23: my phone isn't answering |
| 05:21.39 | DoYouKnow | or asterisk can't reach the phone |
| 05:22.19 | Kyosh | <igcewieling> Kyosh: see sip.conf.sample in the asterisk source dir, there is an option to generate a manager event on registration failure |
| 05:22.32 | Kyosh | igcewieling: do you know what the option is called? i could not find it. |
| 05:22.56 | edong23 | DoYouKnow: i understand that... |
| 05:41.33 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 05:41.33 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 05:42.49 | DoYouKnow | http://pastebin.com/Mu19zt8f |
| 05:42.54 | DoYouKnow | edong23: that's with debug enabled |
| 05:43.57 | DoYouKnow | I called an external # with console dial, then with a phone |
| 05:45.06 | DoYouKnow | doesn't it need to hang up? |
| 05:45.12 | DoYouKnow | so google can call back? |
| 05:45.17 | DoYouKnow | where do I put that in? |
| 05:45.30 | DoYouKnow | oh wait, that's something else |
| 05:45.42 | DoYouKnow | or no.. |
| 05:48.26 | DoYouKnow | google voice works fine through nat, right? |
| 05:58.17 | DoYouKnow | any ideas? |
| 05:58.18 | *** join/#asterisk taylorbyte2013 (~cyberninj@2001:470:1f05:12cc:1e6f:65ff:fec2:bec7) |
| 06:00.22 | DoYouKnow | there's just so much to look through |
| 06:00.34 | *** join/#asterisk taylorbyte2013 (~cyberninj@2001:470:1f05:12cc:1e6f:65ff:fec2:bec7) |
| 06:02.52 | BludSuckingFiend | Kyosh: ;authfailureevents=no ; generate manager "peerstatus" events when peer can't ; authenticate with Asterisk. Peerstatus will be "rejected". |
| 06:03.55 | DoYouKnow | edong23: are you still there? |
| 06:06.03 | BludSuckingFiend | DoYouKnow: That pastbin doesn't appear to have any debug enabled for the channel type you're using |
| 06:06.21 | BludSuckingFiend | I'm not that familiar with Motif, but you should enable debug for that module |
| 06:06.38 | *** join/#asterisk taylorbyte2013 (~cyberninj@2001:470:1f05:12cc:1e6f:65ff:fec2:bec7) |
| 06:09.41 | DoYouKnow | http://pastebin.com/9XJMg7m2 |
| 06:09.42 | DoYouKnow | motif debug |
| 06:09.49 | DoYouKnow | something is seriously wrong. this should've been easy |
| 06:10.37 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 06:11.59 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.135) |
| 06:14.35 | *** join/#asterisk taylorbyte2013 (~cyberninj@2001:470:1f05:12cc:1e6f:65ff:fec2:bec7) |
| 06:14.58 | BludSuckingFiend | <error code="501" type="cancel"><feature-not-implemented xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"/></error></iq> |
| 06:15.11 | BludSuckingFiend | That's in the response from google |
| 06:15.55 | edong23 | DoYouKnow: im barely here |
| 06:16.08 | edong23 | ive been sick for a few days, so im all medsed up |
| 06:16.23 | edong23 | either way, i havent ever actually done anything with motif or google voice |
| 06:16.32 | edong23 | but debug output is helpful in any deployment |
| 06:18.18 | *** join/#asterisk evilman_home (kvirc@95-29-217-225.broadband.corbina.ru) |
| 06:19.54 | *** join/#asterisk scottyob (~scottyob@27-33-131-215.static.tpgi.com.au) |
| 06:20.05 | scottyob | Howdy everyone. I'm failing at getting my SIP phone registered. |
| 06:20.15 | scottyob | is there a way I can view the password it's trying to register with? |
| 06:22.21 | DoYouKnow | edong23: yeah, I'm not feeling 100% either |
| 06:23.04 | DoYouKnow | I have several illnesses, mental/physics |
| 06:23.07 | DoYouKnow | *physical |
| 06:38.38 | DoYouKnow | there is an interesting thing going on here, and it may be from something I don't quite understand |
| 06:38.54 | DoYouKnow | whenever I place a call from asterisk to my phone, and I pick up the phone, I hear ringing |
| 06:39.02 | DoYouKnow | why would that be? |
| 06:42.07 | DoYouKnow | edong23: but I'm not trying to berate what you are experiencing |
| 06:42.10 | DoYouKnow | just saying |
| 06:45.38 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 07:00.26 | ChannelZ | scottyob: not really, it's encrypted |
| 07:04.48 | DoYouKnow | I'll try a reboot |
| 07:04.53 | DoYouKnow | actually hmm |
| 07:04.57 | DoYouKnow | I'll just reset asterisk |
| 07:05.03 | DoYouKnow | I shouldn't have to reboot |
| 07:10.05 | scottyob | ChannelZ: all good. Thanks :) |
| 07:17.11 | *** join/#asterisk FireAndIce (~FireAndIc@219.91.142.227) |
| 07:27.30 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 07:28.57 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 07:34.17 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 07:47.29 | DoYouKnow | most of the time I don't get through, but some of the time i do |
| 07:47.33 | DoYouKnow | Console/dsp is hanging |
| 07:47.54 | DoYouKnow | any ideas? |
| 07:57.08 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 08:16.45 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 08:28.01 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 08:41.59 | *** join/#asterisk bitwize (~bitwize@c83-253-251-219.bredband.comhem.se) |
| 08:46.59 | *** join/#asterisk phiscribe (~david@98.157.5.116) |
| 09:16.30 | drmessano | [2013-02-24 09:15:01] WARNING[3331] loader.c: Error loading module 'codec_silk.so': /usr/lib/asterisk/modules/codec_silk.so: undefined symbol: ast_unregister_file_version |
| 09:16.36 | drmessano | Any thoughts? |
| 09:41.27 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-148-199.pools.arcor-ip.net) |
| 09:45.33 | DoYouKnow | fascinating... |
| 09:45.57 | DoYouKnow | Console/DSP may not have been hung up properly after I comment out the code to hang it up :) |
| 09:46.06 | DoYouKnow | but it does hang in some way for quite some time |
| 09:46.12 | DoYouKnow | I'm trying to figure out what it is |
| 09:53.24 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 10:07.44 | DoYouKnow | Anyone here know why when I attempt to dial non-local telephone #'s, sometimes, but not always, it will time out, and it's usually very quiet? |
| 10:08.06 | DoYouKnow | forget time out, I mean "NOANSWER" |
| 10:08.15 | *** join/#asterisk af_ (~getsmart@78-134-99-27.v4.ngi.it) |
| 10:14.24 | ChannelZ | via what technology? |
| 10:32.47 | DoYouKnow | SIP and google voice |
| 10:33.14 | DoYouKnow | frequently, if I call the phone I'm placing an outbound call with first by a local phone, I can dial the outbound number |
| 10:33.26 | DoYouKnow | but otherwise it will give me a NOANSWER |
| 10:33.34 | DoYouKnow | from the phone itself, as it's handing over the call |
| 10:33.39 | DoYouKnow | (the pbx) |
| 10:37.23 | ChannelZ | GV is kind of a flake |
| 10:37.48 | *** join/#asterisk skirmisha (~vk@46.47.82.189) |
| 10:38.03 | skirmisha | guys, can someone help me with tigase server |
| 10:38.24 | skirmisha | server was moved and new ip was assigned, but now i can't get any node to auth |
| 10:38.30 | skirmisha | i got 401 auth error |
| 10:38.35 | ChannelZ | kind of the wrong venue |
| 10:38.51 | skirmisha | any ideas where the problem is? |
| 10:39.55 | skirmisha | ??? |
| 10:40.14 | ChannelZ | This is an Asterisk channel |
| 10:40.57 | skirmisha | yes, its about asterisk |
| 10:41.03 | skirmisha | trying to get blf working |
| 10:41.13 | skirmisha | i use res_jabber |
| 10:41.28 | skirmisha | but can't understand why ast can't get auth |
| 10:42.32 | ChannelZ | To the XMPP server, I guess you'd have to turn on debug there and see why |
| 10:42.37 | DoYouKnow | ChannelZ: are there any other free alternatives? |
| 10:43.02 | DoYouKnow | to a outward dial? |
| 10:43.36 | skirmisha | yes i got 401 on xmpp server |
| 10:43.42 | skirmisha | and can't understand why |
| 10:43.55 | DoYouKnow | skirmisha: what's blf? |
| 10:44.41 | skirmisha | Busy Lamp FieldTypically |
| 10:45.40 | ChannelZ | Well like I said you'd have to ask the server. You said its IP changed, is the DNS name the same as it was? |
| 10:46.10 | skirmisha | there is no DNS entry, only in etc/hosts and that one is changed |
| 10:46.43 | skirmisha | basically i can't get jabber to auth |
| 11:06.29 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 11:06.29 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.1 (2013/01/22), 10.12.1 (2013/01/22), 1.8.20.1 (2013/01/22), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 11:07.01 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-219-40.red.bezeqint.net) |
| 11:09.36 | DoYouKnow | ChannelZ: I removed google voice and get the same issue |
| 11:10.33 | ChannelZ | Which is what, exactly? You dial a number and just hear nothing? |
| 11:10.46 | DoYouKnow | <PROTECTED> |
| 11:10.56 | DoYouKnow | then it fails after that, repeatedly |
| 11:11.09 | DoYouKnow | I'm on a beta version of asterisk, but I have similar issues in the release version |
| 11:11.12 | DoYouKnow | let me double check |
| 11:12.27 | ChannelZ | Dunno about the console driver never really used it. |
| 11:13.27 | ChannelZ | brb all of my icons seem to have disappeared. |
| 11:18.37 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 11:30.02 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 11:40.23 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 11:41.24 | *** join/#asterisk TimeRider (~steve@027bde06.bb.sky.com) |
| 11:42.18 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.209.133) |
| 11:46.31 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.209.133) |
| 11:47.28 | *** join/#asterisk Ice_Strike (Ice_Strike@87.115.83.87) |
| 11:47.30 | Ice_Strike | Hi |
| 11:49.00 | DoYouKnow | hi Ice_Strike |
| 11:49.04 | DoYouKnow | I'm having trouble with sip |
| 11:49.06 | DoYouKnow | can you help? |
| 11:49.35 | DoYouKnow | I'm setting up 2 extensions, one tcp and one udp, and routing them through the pbx to a SIP account |
| 11:49.45 | DoYouKnow | a sip trunk |
| 11:50.19 | DoYouKnow | sometimes I'll call, and then hang up on the call from my phone, and the call with exit/hangup with a non-zero return status |
| 11:50.29 | DoYouKnow | visible from the asterisk console |
| 11:51.21 | DoYouKnow | then, subsequent calls will not work |
| 11:51.26 | DoYouKnow | until I reset everything |
| 11:52.48 | DoYouKnow | <PROTECTED> |
| 11:54.26 | Ice_Strike | I am not sure, but try adding nat=yes in sip.conf and reloa |
| 11:54.28 | Ice_Strike | reload |
| 11:56.27 | DoYouKnow | didn't work |
| 11:56.44 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 11:57.19 | Ice_Strike | pastebin your sip.conf and extention.conf |
| 12:03.18 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.209.133) |
| 12:04.34 | *** join/#asterisk vlad_sta_ (~vlad_star@178.177.209.133) |
| 12:05.00 | DoYouKnow | sip.conf: http://pastebin.com/df8uJ1E3 |
| 12:05.52 | DoYouKnow | http://pastebin.com/aBvVf7Y7 |
| 12:05.55 | DoYouKnow | extensions.conf ^ |
| 12:06.13 | DoYouKnow | oops |
| 12:06.16 | DoYouKnow | missed the sip.conf |
| 12:07.37 | DoYouKnow | http://pastebin.com/hQECaKtc |
| 12:07.39 | DoYouKnow | ^ sip.conf |
| 12:08.00 | ChannelZ | test with an actual sip device and get the console out of the equation |
| 12:13.35 | DoYouKnow | the remote sip server is sending a 500 internal server error |
| 12:13.47 | DoYouKnow | I got it dialing before |
| 12:13.48 | DoYouKnow | hmm |
| 12:15.15 | DoYouKnow | ChannelZ: I needed the register statement to connect |
| 12:16.15 | DoYouKnow | now it gives me the same thing |
| 12:25.57 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.11.130) |
| 12:26.52 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.11.130) |
| 12:33.34 | *** join/#asterisk ghost751 (~trechber@dslb-088-066-187-031.pools.arcor-ip.net) |
| 13:26.17 | *** join/#asterisk willryder (~tedryder@nc-184-3-102-84.dhcp.embarqhsd.net) |
| 13:32.29 | DoYouKnow | good morning |
| 13:32.45 | DoYouKnow | I've gotten a bit further |
| 13:32.59 | DoYouKnow | now it rings, and I hear a couple words, but it cuts off |
| 13:33.25 | DoYouKnow | the level of understanding is trailing behind the success rate at times, but there's no way of beating the statistics |
| 14:14.50 | DoYouKnow | it seems I am being transferred to the destination through another pbx |
| 14:15.02 | DoYouKnow | and it's messing up things |
| 14:15.13 | DoYouKnow | how do I add delay after a pick-up? |
| 14:15.19 | DoYouKnow | *answered call |
| 14:15.33 | WIMPy | Wait() |
| 14:15.38 | DoYouKnow | thnx |
| 14:31.20 | *** join/#asterisk v4x (~v4x@unaffiliated/v4x) |
| 14:39.00 | *** join/#asterisk salz212 (~chatzilla@182.185.231.80) |
| 14:58.12 | DoYouKnow | anyone know why after my SIP trunk's pbx is called on the other end, there is a short message, then I don't hear anything - until I press hold and release it, and I get a single word |
| 14:58.29 | DoYouKnow | if I keep pressing hold and releasing it on my phone, it will continue to spout out words |
| 14:58.47 | DoYouKnow | however, I am trying to get it to say the whole thing without it doing that :) |
| 15:22.14 | *** join/#asterisk FireAndIce (~FireAndIc@175.100.158.235) |
| 15:25.42 | salz212 | Hi all, I was wondering what is the best practice of keeping voicemail messages on a centralized server.. mounting? having a desperate node? replication- rsync ...? keeping them in DB.... .. community thoughts.. |
| 15:40.45 | file | <PROTECTED> |
| 15:48.14 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-213-040.pools.arcor-ip.net) |
| 15:48.41 | *** join/#asterisk salz212 (~chatzilla@182.178.193.153) |
| 15:50.10 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-213-040.pools.arcor-ip.net) |
| 15:51.59 | DoYouKnow | file: thanks |
| 15:56.50 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 15:57.39 | edong23 | Ok, i have been messing with this a few days... and i think there is a way i can do this, but it is dirty. Im here to see if anyone has a possibly cleaner way. I have a pri coming into an asterisk box. it is coming from a class 5 switch. when someone sends an international call to my asterisk box over the pri, i get TON as being International and 011 stripped off. Is there a channel variable or such i can use to match on international |
| 15:57.39 | edong23 | calls and prefix 011 before sending the call out? |
| 15:58.24 | edong23 | i have a possible alternative... which is to force teh class 5 switch to outpulse the 011, but this is non-standard. I want to keep it as clean as possible |
| 15:58.52 | WIMPy | It isn't being stripped of, it's not part of the number. |
| 15:59.40 | WIMPy | And chan_dahdi can do prefixed to number types. |
| 15:59.41 | edong23 | yah, you are exactly right. i shouldnt say "stripped". |
| 15:59.57 | edong23 | i tried internationalprefix=011 |
| 16:00.01 | edong23 | nothing happened |
| 16:00.24 | WIMPy | Did you reload it? |
| 16:00.42 | edong23 | restarted dahdi |
| 16:00.52 | edong23 | reload asterisk and restarted dahdi |
| 16:00.55 | WIMPy | chan_dahdi. |
| 16:01.18 | edong23 | hm... i thought restart dahdi would do that... but i can try that directly |
| 16:02.12 | WIMPy | Nop. and I wouldn;t be sure a reload works. |
| 16:02.26 | edong23 | i dont know either |
| 16:02.28 | WIMPy | Try to restart Asterisk or unload and load chan_dahdi. |
| 16:02.42 | edong23 | i started to think that maybe internationalprefix is for outgoing and not inbound |
| 16:02.43 | edong23 | ... |
| 16:02.46 | edong23 | but i could be wrong |
| 16:03.02 | edong23 | this is my first experience with international calling |
| 16:03.10 | WIMPy | No, that in. |
| 16:03.12 | edong23 | i can call international over any of my 12 providers |
| 16:03.24 | edong23 | but i havent set it up on my switch yet... |
| 16:03.33 | edong23 | hm.. WIMPy ok, ill give it a shot |
| 16:03.39 | edong23 | i dont have the ability to test from where i am |
| 16:03.44 | edong23 | not, a true test |
| 16:04.04 | edong23 | all i can do is simulate from here, but i need to actually dial the digits for a true test |
| 16:04.10 | edong23 | ill check it in a few hours |
| 16:04.15 | edong23 | but i believe i did all of this |
| 16:04.20 | edong23 | even stopped asterisk |
| 16:04.23 | edong23 | and restart |
| 16:04.37 | edong23 | ill check them again though |
| 16:11.59 | igcewieling | edong23: I had a problem with the internationalprefix and nationalprefix before. I because of this I decided to stop using them. |
| 16:12.22 | igcewieling | What specifically are you trying to do? |
| 16:12.28 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 16:12.46 | edong23 | determine if an inbound call (coming from the pri) is an international call or national |
| 16:12.59 | igcewieling | In most places you do not set international or national prefix and set the pridialplan to unknown |
| 16:13.20 | edong23 | there is that option... thats what i said, but it gets very dirty |
| 16:13.28 | edong23 | there are a few options |
| 16:13.40 | edong23 | but the correct way is to look at TON and then route according to that |
| 16:13.54 | edong23 | which, im fine with prefixing digits.. if that would work |
| 16:13.58 | igcewieling | edong23: ah. no really easy way. What we do is if the CID is 11 digits and starts with a 1 or is 10 digits and does not start with a 1 or 0 then it is domestic, all others are international |
| 16:14.02 | WIMPy | igcewieling: And then you fix the callerid in the dialplan? |
| 16:14.03 | edong23 | but i couldnt get it to work... |
| 16:14.35 | edong23 | igcewieling: im talking about the called party |
| 16:14.38 | edong23 | not the calling party |
| 16:14.44 | igcewieling | WIMPy: yup. We don't get many international calls so it has not been a big deal. I just like to "fixup" the CID so people can call back the TN from their phone's call history |
| 16:14.52 | WIMPy | Actually the prefixes don't work for me either. |
| 16:15.23 | WIMPy | I never really looked at it. But that loos quite bad actually. |
| 16:15.28 | WIMPy | looks |
| 16:15.32 | igcewieling | edong23: you want to know if the call coming into asterisk from the PRI is calling from outside your country? |
| 16:15.45 | edong23 | no, the called party |
| 16:15.56 | edong23 | i want to know if the call coming to the asterisk box is going to outside my country |
| 16:16.06 | edong23 | the called party number is international |
| 16:16.13 | edong23 | the calling party will always be national |
| 16:16.17 | edong23 | for me |
| 16:16.18 | igcewieling | edong23: heh, the prefix stuff will do notthing like that. |
| 16:16.21 | WIMPy | Yet another point on the list of why not to use dahdi. |
| 16:16.35 | igcewieling | you need to do it in your dialplan |
| 16:16.56 | igcewieling | WIMPy: hey now, PRIs are the BEST! |
| 16:16.59 | WIMPy | But that's what the prefix stuff is there for, isn't it? |
| 16:17.07 | WIMPy | I even think it used to work. |
| 16:17.07 | edong23 | if libss7 was better, i would use it |
| 16:17.15 | edong23 | but last time i tried it, it was bonched |
| 16:17.26 | igcewieling | WIMPy: they are for incoming calls to asterisk from the PRI, they add digits to the calling part number |
| 16:17.31 | WIMPy | igcewieling: Tehy are, but there are other drivers that I feel more comfortable about. |
| 16:17.34 | igcewieling | s/part/party |
| 16:17.52 | WIMPy | Yes, that's waht we want. |
| 16:18.01 | edong23 | igcewieling: do you have an idea of what i would set in my dialplan to determine this? |
| 16:18.03 | edong23 | cause i dont... |
| 16:18.03 | igcewieling | at least that is what we used them for. most of our PBXs use SIP w/POTS for backup/failover |
| 16:18.56 | igcewieling | edong23: what country are you in? normally you set the pridialplan=unknown and then your carrier figres out what the TON should be. Manually setting it to national or international causes exactly the issue you have. |
| 16:18.59 | WIMPy | This sucks. I think I need to take a closer look at that issue tomorrow. |
| 16:19.29 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 16:19.47 | WIMPy | Yes, always set them to unknown unless you have a good reason to do otherwise and know exactely what you're doing. |
| 16:20.06 | edong23 | igcewieling: im in US.... and though this is kinda an option, setting pridialplan to unknown will seriously break my other routing, but...i can manage my way around it if it is the only way |
| 16:20.07 | WIMPy | But this is about the caller-IDs or am I in the wrong movie? |
| 16:20.48 | edong23 | pridialplan unknown is absolutely not the best way to set up a pri |
| 16:20.53 | edong23 | so, noone can make that argument |
| 16:21.09 | edong23 | but if somehting is missing from libpri so i can determinethis information, then so be it |
| 16:21.19 | WIMPy | It's usually the only way. |
| 16:21.22 | igcewieling | edong23: it is the best way to set up a PRI in the USA. |
| 16:21.35 | *** join/#asterisk willryder (~tedryder@nc-184-3-102-84.dhcp.embarqhsd.net) |
| 16:21.35 | WIMPy | Anywhere |
| 16:21.59 | igcewieling | WIMPy: some carriers in other countries do not support unknown TONs on PRI, but all of them in the USA do |
| 16:22.16 | WIMPy | Seriousely? Where? |
| 16:22.24 | edong23 | no, um... |
| 16:22.33 | edong23 | WIMPy: igcewieling i think there is a detatchment here |
| 16:22.35 | igcewieling | WIMPy: It has been a long time, check the mailing list archives. |
| 16:22.40 | edong23 | im not getting a pri from some unknown |
| 16:22.47 | edong23 | i am providing the pri to myself |
| 16:22.51 | edong23 | from my class 5 switch |
| 16:23.08 | edong23 | in this situation, there is no reason i shouldnt be able to use the TON bit that actually do work |
| 16:23.09 | igcewieling | edong23: ah, then you should be fine since you control both ends 8-) |
| 16:23.37 | edong23 | i do control both ends, but i dont want it to get ugly still. I can have my class 5 switch outpulse the 011 |
| 16:23.42 | edong23 | that is teh cleanest way |
| 16:23.46 | edong23 | but not the correct way... |
| 16:23.52 | edong23 | if its the only way, then im ok with it |
| 16:24.09 | edong23 | i was just trying to see if i could use the TON bits to do something.. |
| 16:24.23 | igcewieling | edong23: this may help http://lists.digium.com/pipermail/asterisk-users/2005-May/102837.html |
| 16:24.33 | igcewieling | edong23: I'm sure you can set the TON but I can't find how at the moment. |
| 16:24.43 | edong23 | thats OUTGOING |
| 16:24.44 | edong23 | lol |
| 16:24.45 | *** join/#asterisk crazed1 (themrrober@cpe-76-90-21-3.socal.res.rr.com) |
| 16:25.07 | WIMPy | The information to set TON via cllaerid is in the chan_dahdi.conf.sample. |
| 16:25.09 | edong23 | im not dialing internation from asterisk over the pri |
| 16:25.20 | edong23 | im recieving them from my class 5 swtich over the pri to asterisk |
| 16:26.04 | igcewieling | ${CALLINGTON} * Caller Type of Number (PRI channels) |
| 16:26.21 | WIMPy | edong23: And that's what the prefixes should sort out. |
| 16:26.25 | edong23 | igcewieling: does that exist? |
| 16:26.44 | igcewieling | edong23: in your exact version of Asterisk? I don't know, but it is worth looking into |
| 16:27.02 | WIMPy | CALLERID(num-plan) |
| 16:27.02 | edong23 | igcewieling: link to that resource please? |
| 16:27.19 | edong23 | WIMPy: i dont know what callerid would do for me... |
| 16:27.40 | WIMPy | That function should give you the TON. |
| 16:27.53 | igcewieling | http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List |
| 16:28.23 | edong23 | igcewieling: this could work, i can test later |
| 16:28.35 | WIMPy | You know that voip-info contains lots of stuff that was valid in the Asterisk 1.2 times. |
| 16:28.43 | WIMPy | ~voip-info |
| 16:28.43 | infobot | it has been said that voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 16:28.48 | edong23 | this is almost literally what i asked for in my first question.. i ihave used TNS but not TON |
| 16:28.53 | edong23 | shoudl be pretty simle |
| 16:28.56 | igcewieling | WIMPy: looks like ANI-num-plan or num-plan to CALLERID may do it. |
| 16:29.21 | WIMPy | They should do it. |
| 16:30.04 | WIMPy | Noone cares, but I think I should fix that box using dahdi anyway. |
| 16:30.13 | igcewieling | manually setting the TON is like trying to commit suicide by putting a plastic bag over your head. Sure it may work, but is will be very unpleasant and not very effective in most cases |
| 16:30.26 | WIMPy | It has exactely the same issue. |
| 16:30.28 | edong23 | im not wanting to set it |
| 16:30.52 | edong23 | im wanting to just be able to determine, on the incoming call, if it is internation, so i can then route based on that |
| 16:31.12 | igcewieling | edong23: do the incoming calls come in over DAHDI? |
| 16:31.13 | edong23 | im starting to think having my switch outpulseasdialed might be better |
| 16:31.19 | WIMPy | Why not set it if you know what it is you're calling? Probably unneccessary work, but nothing wrong. |
| 16:31.19 | edong23 | yea |
| 16:31.44 | edong23 | WIMPy: because im not calling over the pri |
| 16:31.49 | edong23 | im recieving over the pri |
| 16:31.56 | igcewieling | WIMPy: exactly. IF the dialed number starts with 011 then it is international -- simple, trivial |
| 16:32.54 | edong23 | i suppose that is a way to do it... routing based on digits isnt the right way though. it may be the only way with dahdi and asterisk right now |
| 16:32.57 | WIMPy | edong23: I got that. I was just commenting igcewieling |
| 16:33.23 | WIMPy | It shouldn't be. |
| 16:33.43 | WIMPy | I'm off to a chilli test party in a moment, but I will take a look in to that tomorrow. |
| 16:33.44 | igcewieling | PSTNs have been routing based on dialed digits since the beginning of dialed digits |
| 16:33.50 | WIMPy | That needs to be fixed. |
| 16:34.14 | igcewieling | WIMPy: then we need "local" "toll" and "international" buttons on all our phones. |
| 16:34.35 | edong23 | um... pstns routing on cic codes |
| 16:34.44 | edong23 | and numerous other routing meathods |
| 16:34.57 | WIMPy | Various things, yes. |
| 16:35.03 | igcewieling | from the standpoint of the user and a PBX they route on dialed digits. |
| 16:35.08 | edong23 | dialed digits would the last on the list |
| 16:35.14 | WIMPy | The NP routing nightmare. |
| 16:35.21 | igcewieling | that is just translated into whatever other codes you need in your SS& |
| 16:35.44 | edong23 | this is no different... though |
| 16:35.52 | igcewieling | edong23: maybe you want to use chan_ss7 instead of chan_dahdi? |
| 16:35.54 | edong23 | over ss7 a call that is internation doesnt have 011 on it |
| 16:35.55 | edong23 | lol |
| 16:36.00 | edong23 | igcewieling: abasolutely not |
| 16:36.02 | WIMPy | Not neccessarily. If you receive a call and try to call back, you will proably send the raw number including the TON as received and not as you would display or dial them. |
| 16:36.06 | edong23 | unless they have fixed libss7 |
| 16:36.40 | igcewieling | no idea. ss7 handoff to our carriers would increase our costs by a hundredfold |
| 16:36.55 | igcewieling | so I don't keep on libss7 8-| |
| 16:37.14 | edong23 | yeah, im not sure |
| 16:37.19 | edong23 | it may have gone somewhere |
| 16:37.25 | edong23 | but, last time i used it, it was a bust |
| 16:37.35 | WIMPy | (with yoy = your phone/PBX. |
| 16:37.36 | edong23 | sangoma offers and ss7 stack they guarantee |
| 16:37.38 | WIMPy | ) |
| 16:38.01 | edong23 | but it is 5K |
| 16:38.04 | igcewieling | I love Sangoma but their support was less than great the last time we needed it. |
| 16:38.17 | edong23 | really? |
| 16:38.32 | edong23 | i have always (on the 2 times i needed it in 7 years) gotten good support |
| 16:38.40 | edong23 | but, i havent needed sangoma support in years |
| 16:38.53 | igcewieling | I used to always get great support from them, but not that last time (about 4 months ago) |
| 16:39.01 | edong23 | other than just rma t1 card someone plugged into a poe injector |
| 16:39.57 | edong23 | ok thanks igcewieling and WIMPy ill try to outpulse as dialed on my class5 switch |
| 16:40.16 | edong23 | its an easy change initially to test, but really hard to implement on my already build environment |
| 16:40.21 | edong23 | but i should be abel to do something |
| 16:40.34 | edong23 | i have a singel asterisk that is my tdm gateway |
| 16:40.38 | edong23 | 4 pris |
| 16:40.47 | edong23 | so, i can slice and dice how i see fit on it |
| 16:40.51 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 16:41.53 | edong23 | and igcewieling it makes sense that it would cost more for ss7 handoff |
| 16:42.00 | edong23 | its far greater risk |
| 16:42.09 | edong23 | you dont have the ability to control as much over ssh |
| 16:42.12 | edong23 | ss7 |
| 16:42.13 | edong23 | oop |
| 16:42.30 | edong23 | whereas, i can hinder your callerid spoofing, or such on a pri |
| 16:42.38 | igcewieling | edong23: apparently ss7 handoff is inexpensive in the EU. |
| 16:42.45 | edong23 | ss7 is a "you handle it" thing |
| 16:42.47 | edong23 | same here |
| 16:42.51 | edong23 | expensive |
| 16:42.51 | igcewieling | I always thought that is odd |
| 16:42.53 | edong23 | compared to a pri |
| 16:43.46 | edong23 | i mean, it is a much different system, for sure, but 100fold difference in price is pretty outrageous |
| 16:43.53 | edong23 | im assuming you are being generous |
| 16:43.57 | edong23 | but here, it is 19 |
| 16:43.59 | edong23 | 10x |
| 16:45.24 | igcewieling | moves the decimal point |
| 16:45.45 | edong23 | thats pretty outrageous |
| 16:46.16 | igcewieling | We have good luck with our IP/SIP handoff to our wholesalers, but tickets usually come back with NTF 8-| |
| 16:54.43 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:54.43 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:58.58 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 16:59.17 | jmls | hi guys, got a strange problem that I'm trying to fix on a customer site |
| 16:59.36 | jmls | I have register => foo:bar@host in sip.conf |
| 16:59.47 | jmls | I alos have a [foo] peer defined |
| 16:59.59 | jmls | but sip show registry is showing nothing |
| 17:00.32 | jmls | I was wondering what the causes of this could be ? |
| 17:00.40 | igcewieling | do a sip reload and see if there are any errors/warnings in the CLI |
| 17:01.10 | jmls | there are none - I've checked several times, core restarted (even restarted the box!) |
| 17:01.30 | jmls | sorry, should say this is asterisk-11 |
| 17:02.24 | Ice_Strike | How should a daemon process (AMI) be designed to listen multiple actions and events? For example: 50 agents currently on the calls and how should a daemon to monitor the Actions/Events from 50 agents? |
| 17:03.17 | igcewieling | Ice_Strike: the code I posted last night is a good example of that |
| 17:04.06 | igcewieling | Ice_Strike: you get all events over your one AMI connection |
| 17:05.05 | jmls | is SIP/2.0 405 Method Not Allowed a bad thing ... |
| 17:05.26 | jmls | after a <--- SIP read from UDP:<hostip> |
| 17:06.07 | igcewieling | jmls: is that the register of something like OPTIONS or INVITE? |
| 17:08.12 | jmls | options I think. trying to find it in the log |
| 17:08.25 | igcewieling | if options, then it is harmless and common |
| 17:08.39 | igcewieling | turn off qualify and the options messages will stop. |
| 17:08.42 | jmls | sorry, notify |
| 17:09.01 | igcewieling | NOFITY is likely MWI |
| 17:09.23 | jmls | customer can make outbound calls using this host, |
| 17:09.28 | igcewieling | do you have a mailbox=something under [foo]? |
| 17:09.36 | jmls | but no inbound (as there is no register) |
| 17:09.42 | jmls | igcewieling . yes |
| 17:09.45 | igcewieling | *nod* registration has nothing whatsoever to do with outbound calling |
| 17:10.04 | Ice_Strike | igcewieling I understand that, I meant what the best way to keep track of each agent in a daemon process |
| 17:10.53 | igcewieling | Ice_Strike: that is far beyond the scope of this channel. However, each message should have a uniqueid or linkedid (better) so you can figure out what events are for what call. |
| 17:11.23 | igcewieling | Ice_Strike: a "state machine" may be what you are looking for. It is a design which works well for events and stuff like that. |
| 17:12.19 | Ice_Strike | Great, i'll look into that |
| 17:12.52 | Ice_Strike | Should uniqueid or linkedid to be stored in a array or database for tracking purpose? |
| 17:16.50 | jmls | is there any problem woith ordering of externaddr, localnet, register => and [peer def[ ? |
| 17:28.47 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 17:36.18 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 17:56.23 | *** join/#asterisk gusto (~gusto@ppp-93-104-69-51.dynamic.mnet-online.de) |
| 17:59.42 | drmessano | Anyone using codec_silk ? |
| 18:11.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 18:15.51 | *** join/#asterisk salz212 (~chatzilla@182.185.204.69) |
| 18:37.56 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.6.201) |
| 18:40.42 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 18:41.24 | *** join/#asterisk Alagar (~helpdesk@vsusg1.vernalissystems.com) |
| 18:53.42 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 19:10.01 | ChannelZ | drmessano: I tried but don't have anything to use it with really. SFA was never made to support it |
| 19:11.06 | ChannelZ | I played with it briefly using CSipSimple on Android |
| 19:13.49 | *** join/#asterisk andy09usa (~Andrey@audotov.com) |
| 19:27.46 | drmessano | ChannelZ, I can't seem to get it to load on an OpenVZ VM. |
| 19:27.54 | drmessano | Works fine on two non-virtualized boxes |
| 19:28.05 | drmessano | NFI why |
| 19:31.03 | ChannelZ | hmm |
| 19:31.49 | ChannelZ | using -generic arch? |
| 19:35.13 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 19:45.43 | *** join/#asterisk garymc (~chatzilla@host86-174-143-125.range86-174.btcentralplus.com) |
| 19:55.27 | *** join/#asterisk hackish (b8a2b885@gateway/web/freenode/ip.184.162.184.133) |
| 19:56.59 | hackish | I haven't had any luck finding this one... I want to make automixmon record all calls coming in and going out. All the howtos cover hitting some buttons to initiate call recording... any ideas? |
| 19:59.17 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 19:59.39 | ChannelZ | use MixMonitor in your dialplan? |
| 19:59.41 | [TK]D-Fender | hackish: Because "auto" is for USER triggered only |
| 19:59.55 | [TK]D-Fender | hackish: Call Monitor YOURSELF in the dialplan |
| 20:00.16 | [TK]D-Fender | hackish: there is no setting for "always". * records when you tell it to. |
| 20:01.26 | hackish | ok, just learning this stuff, I'll see what the dialplan thing looks like. |
| 20:02.52 | [TK]D-Fender | hackish: It's only 95% of the job of configuring Asterisk |
| 20:03.29 | hackish | yep. I was looking at one of these no-brainer things like freepbx but they're all linux and myself specific. |
| 20:04.09 | hackish | old fashioned way for me unfortunately |
| 20:05.19 | [TK]D-Fender | hackish: Yes, and FreePBX generates thousands of lines of dialplan.... for you. It still has to be there |
| 20:06.36 | hackish | can you recommend any specific places to get a default dialplan? I'm not really doing anything special. 2 lines and voicemail |
| 20:07.03 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 20:08.47 | [TK]D-Fender | hackish: There is no such thing as "default" |
| 20:09.12 | hackish | understood but I'm sure it's a common recipe |
| 20:09.13 | [TK]D-Fender | hackish: It is your job to make yourss do whatever you want it to do. |
| 20:09.29 | [TK]D-Fender | hackish: Calling monitor is ONE line in your dialplan. Put it where YOU want. |
| 20:10.02 | [TK]D-Fender | hackish: There is no magic shortcut for this. Go read the book. |
| 20:10.04 | [TK]D-Fender | ~book |
| 20:10.04 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:10.06 | [TK]D-Fender | ^^^ |
| 20:10.33 | [TK]D-Fender | hackish: Maybe you'll have 5 places you'll have to put it. Maybe a hundred. Maybe ONE... we don't know what your dialplan looks like |
| 20:12.06 | hackish | so does every single person configuring the system write a 100 line dialplan themselves or are there config scripts or samples available somewhere you'd recommend? I've been quite unsuccssful at finding much in the way of howto as everyone seems to be only interested in selling their consulting services. |
| 20:12.28 | hackish | Getting something close, I can easily edit it as needed. |
| 20:13.21 | [TK]D-Fender | hackish: 100 lines is nothing big at all. |
| 20:13.33 | hackish | somehow I was afraid you were going to say that. |
| 20:13.50 | ChannelZ | and for "2 lines and voicemail" it could be only a dozen or less |
| 20:13.53 | [TK]D-Fender | hackish: And no there are no magic scripts. The concept of samples is almost worthless as everyone's needs may be different. |
| 20:14.29 | hackish | To me it seems like writing sendmail config files manually and compiling them with m4. Great, I program in assembly language all day long but not to write a hello world program. |
| 20:14.31 | [TK]D-Fender | hackish: There is no shortcut. There is no "how to" for your needs. You will have to learn how this all works. |
| 20:16.09 | hackish | So would you consider the GUI type config programs a waste of time then? |
| 20:16.10 | [TK]D-Fender | hackish: I could write a 5 line dialplan that takes all calls from SIP phones I wire up over here and lets them dial each other with voicemail if no answer, dial out an ITSP, and ring all for incoming calls. |
| 20:16.28 | [TK]D-Fender | hackish: GUI's have their place. |
| 20:16.51 | [TK]D-Fender | hackish: You are failing to comprehend that the persons NEEDS determines what is best. |
| 20:17.47 | hackish | All I'm trying to say is that my needs are very very simple. I'm going to assume that there are thousands of small home office guys who just need a line in with voicemail. |
| 20:17.59 | [TK]D-Fender | ~assume |
| 20:17.59 | infobot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav It makes an (ass) out of (u) and (me) |
| 20:18.34 | [TK]D-Fender | hackish: You want to record? Put Monitor() before your Dial()'s |
| 20:18.52 | [TK]D-Fender | hackish: This ain't Raw-Cat Sigh Hence |
| 20:18.58 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 20:20.44 | hackish | yes, the record part was just an additional thing. I really got screwed over hardcore by my phone supplier. I told them I wanted to switch so I'd do VOIP myself and they said well your contract expires next week. If you don't switch before then you're signed up for another 3 years or we can cancel it and you lose your number. Um... really? Ok f-u I guess I'll have to get this stuff working this weekend. |
| 20:23.08 | [TK]D-Fender | hackish: I can't tell where the line between hardware & phone service cross in that description |
| 20:24.30 | hackish | well the phone service is my cable company. They've been charging my business $60/mo for cid, voicemail and a phone line. So I bought an ATA and am in the process of getting a sip trunk. From there I just need to port the DID. It's the DID they're trying to hold ransom. |
| 20:24.52 | igcewieling | hackish: installing a PBX for the first time requires a huge learning curve |
| 20:25.13 | hackish | igcewieling, that's evident to me at this point. |
| 20:25.35 | hackish | I got quoted $2500 from 2 local companies who do it. |
| 20:25.57 | igcewieling | You may want to consider FreePBX. You still have a big learning curve, but for a basic PBX it isn't that bad |
| 20:26.18 | hackish | freepbx unfortunately only runs on linux and mysql |
| 20:26.22 | igcewieling | hackish: where are you located? |
| 20:26.30 | hackish | Ottawa, Canada |
| 20:26.35 | igcewieling | hackish: same as Asterisk |
| 20:26.42 | [TK]D-Fender | [15:26]hackishfreepbx unfortunately only runs on linux and mysql <- and why is this bad? |
| 20:27.07 | hackish | not allowed to run linux or mysql. |
| 20:27.07 | igcewieling | If you can't use linux and mysql then you should abandon all hope of making Asterisk work for you. |
| 20:27.21 | [TK]D-Fender | hackish: Because...? |
| 20:27.23 | *** join/#asterisk fakhir__ (~fakhir@unaffiliated/fakhir) |
| 20:27.46 | igcewieling | [TK]D-Fender: I don't think it matters. 8-| No linux == no asterisk pbx |
| 20:27.59 | hackish | icewieling, I got astresk working on BSD without any issues. FreePBX on the other hand is so full of linux-isms it won't even compile |
| 20:28.38 | hackish | TK, not allowed to run linux or mysql. Policy. |
| 20:28.43 | igcewieling | hackish: Ah, I'd forgotten about BSD support. I didn't know those crazy people who make sure it work on BSD were still around. 8-) |
| 20:29.02 | Ice_Strike | I dont think asterisk available on Windows |
| 20:29.04 | [TK]D-Fender | hackish: And the reason you can't run your obviously important phone services on a proper dedicated machine as is recommended is.....? |
| 20:29.18 | igcewieling | [TK]D-Fender: Set up to fail. 8-) |
| 20:29.28 | hackish | it is a dedicated machine |
| 20:29.44 | [TK]D-Fender | igcewieling: Yes. His own rules are fucking him over. |
| 20:30.03 | igcewieling | [TK]D-Fender: yup. set up to fail, so no point in discussing it further. |
| 20:30.06 | hackish | not my rules unfortunately |
| 20:30.23 | [TK]D-Fender | Bake me a cake. NOW. Caveat : not allowed to use any flour or sugar of any kind. |
| 20:30.34 | [TK]D-Fender | \hackTime to learn Asterisk then |
| 20:30.36 | igcewieling | gets up and leaves the room |
| 20:30.41 | [TK]D-Fender | hackish: Time to learn Asterisk then |
| 20:30.44 | igcewieling | there, problem and solution |
| 20:31.15 | hackish | sucks don't it. |
| 20:31.16 | Ice_Strike | hackish You have two options ... Learn Asterisk or Hire someone cheap from elance |
| 20:31.44 | igcewieling | Ice_Strike: He may have trouble finding people with experience with Asterisk and BSD |
| 20:31.55 | hackish | yep, the approved vendors know they have you by the short and curlies. |
| 20:32.09 | [TK]D-Fender | hackish: "Approved" by whom? |
| 20:32.10 | hackish | I can't imagine that BSD has anything to do with it. |
| 20:32.22 | hackish | by the org I work for. |
| 20:32.30 | igcewieling | My advice? Get them to sign a 2 year contract and use that time to come up with a real plan and solution. |
| 20:33.30 | [TK]D-Fender | hackish: You are basically setting up a who house of card between internal "rules" as to what you can run, who your "approved" vendors are. I'm curious how many other "made to fail" conditions you are willing to impose before you realise that you completely fucking yourself over for no good reason. |
| 20:34.35 | [TK]D-Fender | hackish: https://www.youtube.com/watch?v=v77SF4TFUoM |
| 20:38.34 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 20:46.17 | hackish | TK, don't quite know why it's such a big deal what OS they shove down my throat. |
| 20:47.27 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 20:47.37 | hackish | I could probably tolerate a windoze config tool. |
| 20:49.45 | [TK]D-Fender | hackish: The bigger deal is your list of "approved" vendors |
| 20:50.20 | [TK]D-Fender | hackish: But if you wanted an easier solution, you've already cut all of those off. |
| 20:51.41 | [TK]D-Fender | hackish: So sure, you can use BSD. It'll be all basic from-scratch config, and apparently ha to be provided from somebody's massively restricted vendors unless you can learn how to do it yourself extremely quick |
| 20:52.03 | hackish | The companies allowed to do this work need certain pre-requisites that are hard to get. I have the necessary qualifications, I'm just not a telecom expert. |
| 20:52.06 | [TK]D-Fender | hackish: Mind you, you also havent told us what you're even using right now as phones go |
| 20:52.43 | *** join/#asterisk serafie (~erin@24.214.158.242) |
| 20:58.41 | hackish | tk, it's just a plain old phone hooked to a cisco psa122. I've managed to config the cisco and get it talking to the astrisk. |
| 20:58.52 | hackish | spa122 rather |
| 21:00.14 | igcewieling | hackish: now comes the hard part. |
| 21:00.42 | [TK]D-Fender | hackish: "From there I just need to port the DID." <- have you picked your vendor yet? |
| 21:01.12 | [TK]D-Fender | hackish: I was asking about what you were using BEFORE BTW |
| 21:01.15 | hackish | yes. Once again, on an approved list |
| 21:01.21 | [TK]D-Fender | hackish: Not what you may be testing * with now |
| 21:01.36 | [TK]D-Fender | hackish: I didn't ask if you had a list. I asked if you made the choice |
| 21:02.04 | [TK]D-Fender | hackish: Stop being a generic secret-squirrel |
| 21:02.07 | hackish | the answer is still there. |
| 21:02.20 | [TK]D-Fender | hackish: And every term is vague |
| 21:03.05 | igcewieling | which carrier/vendor are you using? |
| 21:03.21 | hackish | primus SGS. |
| 21:03.42 | igcewieling | do you have a DID with them already which you can use for testing? |
| 21:03.54 | [TK]D-Fender | hackish: So what was your previous phone setup? |
| 21:04.00 | hackish | they are providing one for me, then I port over to it later. |
| 21:07.46 | hackish | the previous phone setup consisted of a normal business line through a cable company. |
| 21:08.36 | hackish | plus a number of different phones for making outbound calls. |
| 21:10.00 | hackish | my objective is to get the business line side of things up ASAP. |
| 21:11.17 | hackish | Other than that I will later get some sort of codec that's already been compiled for the BSD I use. |
| 21:11.37 | [TK]D-Fender | hackish: So you had NO PBX at all before and just a buch of phones plugged in no smarter than with your home line? |
| 21:12.03 | [TK]D-Fender | hackish: At which point then you might as well port the number over, point your SPA direct to your new provider and plug your phones into THAT instead and be done with it |
| 21:12.20 | [TK]D-Fender | hackish: You'll have what you have now with a new provider. The End. |
| 21:12.38 | [TK]D-Fender | hackish: And then if you actually to be able to do something smart with it... go learn Asterisk |
| 21:12.52 | [TK]D-Fender | hackish: But at leasrt you'll be off your cable provider |
| 21:13.16 | hackish | The ultimate destination is to have more than 1 transport for a call depending on the destination. |
| 21:13.39 | [TK]D-Fender | hackish: Fine. go learn Asterisk. In the meantime, get off your contract before you're locked in any further |
| 21:17.23 | hackish | Do you think it would be a big deal to switch from the SPA box to asterisk? While I do need to get out of the stupid contract fast I also have a box that the SPA has to be plugged into to make some calls. This can be replaced with an asterisk codec that just takes a USB device and already runs on the BSD machines. |
| 21:22.59 | *** join/#asterisk corretico (~luis@190.211.93.38) |
| 21:31.46 | [TK]D-Fender | hackish: "asterisk codec" <- this is not a real term as-used |
| 21:32.08 | [TK]D-Fender | hackish: "also have a box that the SPA has to be plugged into to make some calls" <- clarify this |
| 21:32.58 | [TK]D-Fender | hackish: "This can be replaced with an asterisk codec that just takes a USB device and already runs on the BSD machines" ,- actually... clarify this whole phrase. What is this "USB thingy" you're talking about? |
| 21:34.57 | hackish | I don't know what the module is called. |
| 21:35.49 | [TK]D-Fender | hackish: So far all I see is "USB" and "runs on BSD machines". No description of function or at what point you believe Asterisk is expected to be able to support it. |
| 21:35.54 | hackish | It's called a ZRTP AES Codec |
| 21:36.50 | [TK]D-Fender | hackish: I'm all but certain that no Asterisk code supports that thing |
| 21:38.39 | [TK]D-Fender | ZRTP is a protocol, not a CODEC, and AES is an encryption algorithm. Two pieces out of a triad that is missing the third piece and carriying the name of the missing piece. That is a very broken term |
| 21:40.42 | hackish | The third piece is just a card that plugs in via USB. |
| 21:41.44 | [TK]D-Fender | hackish: And I doubt that Asterisk has any code to make use of it |
| 21:42.03 | *** join/#asterisk hesco (~hesco@c-174-48-250-91.hsd1.fl.comcast.net) |
| 21:42.12 | [TK]D-Fender | hackish: I know there are ZRTP patches for *. Unsure how easy it will be to bring those in if it hasn't been merged. |
| 21:42.30 | hackish | I don't have to worry about that part. It's already done for me and they just provide it as a .so file |
| 21:42.59 | [TK]D-Fender | hackish: a ".so" for Asterisk? |
| 21:43.08 | hackish | yes |
| 21:43.43 | [TK]D-Fender | hackish: Hope it works for whatever version you'll be running. Keeping in mind this is a purely SOFTWARE thing for I dunno ... everyone else. |
| 21:44.15 | [TK]D-Fender | hackish: Relying on some USB dongle for this is jsut another layer of ridiculous dependency in this house of cards |
| 21:44.25 | hesco | I need to do a hard reset to factory defaults (including losing the IP address to the boot server) on a Polycom SIP501. Can anyone help me know how, please? I tried the 4-6-8-* combination but it simply purged the config and pulled the same corrupted configuration down from the wrong boot server. I need to change the boot server IP, but am locked out by a non-default password. |
| 21:46.05 | hackish | The USB dongle is just a key. Without it the system is useless. |
| 21:46.57 | [TK]D-Fender | And software could have whatever values are in there all on it's own. |
| 21:47.06 | hackish | The whole thing works with 1.8.20 so that's what I installed. |
| 21:47.07 | [TK]D-Fender | And people have already done this "straight' |
| 21:47.29 | [TK]D-Fender | But hey, best of luck with it anyway |
| 21:49.22 | Ice_Strike | Polycom is nice phone, I have them at work.. about 60 Polycom phones! |
| 21:50.00 | Ice_Strike | I've set Polycom to connect to Boot Server (TFTPD) |
| 21:51.08 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 21:57.36 | lorsungcu | cool story Ice_Strike |
| 21:58.16 | [TK]D-Fender | I LIKE TURTLES! |
| 21:59.12 | [TK]D-Fender | hesco: " I tried the 4-6-8-* combination" -> when? What did you do right after? |
| 22:05.15 | lorsungcu | anyone tried building an OSS EPM template for Snom VIsion? |
| 22:21.23 | *** join/#asterisk micdobro (~mic@0305ds4-vby.2.fullrate.dk) |
| 22:26.12 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 22:29.47 | hesco | [TK]D-Fender: I unplugged the phone, waiting a minute and then plugged it back in. It grabbed its boot rom from the old boot server and left me locked out with its non-default password. I was not sure what to do after that so I put off the project until I posted this report in this channel. |
| 22:31.58 | hesco | In searching about, all the advise I found seems to assume I have a password to reset the defaults using the menus. |
| 22:37.39 | hesco | I read some advice about using the mac address as a password after the 4-6-8-* sequence, but that did not work for me. |
| 22:39.14 | lorsungcu | hesco: 456 does not work? |
| 22:39.54 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 22:40.50 | hesco | no, the config pushed down from the tftp server sets a non-default password |
| 22:41.32 | hesco | that is where I get stopped. If 456 worked, I would have simply pointed the phone at my new server and be back in business. |
| 22:41.55 | hesco | As it is, the phone is registering to the old server, rather than to my server. |
| 22:43.50 | lorsungcu | and you don't have access to the old server/it's config files? |
| 22:50.37 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 22:55.56 | hesco | afraid not. |
| 22:56.15 | hesco | lorsungcu: ^^^^^ |
| 23:01.17 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:02.38 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 23:02.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@bas1-montreal08-1279584386.dsl.bell.ca) |
| 23:14.03 | lorsungcu | is the phone local to you? |
| 23:14.30 | lorsungcu | or do you have control of DHCP in the phones network |
| 23:15.28 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.4.243) |
| 23:21.10 | *** join/#asterisk serafie (~erin@24.214.158.242) |
| 23:23.42 | WIMPy | edong23: I revert my statement. The prefix options do what they're supposed to do. It was the IAX trunk that had wrong IDs. |
| 23:25.14 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:27.49 | edong23 | WIMPy: no idea... i havent had a chance to test yet |
| 23:35.26 | *** join/#asterisk Phoebus (~Phoebus@pdpc/supporter/active/phoebus) |
| 23:44.53 | hesco | lorsungcu: yes, the phone is on my desk and the dhcp server is in the appliance router on my desk as well. This is an interesting direction of inquiry. However I just discovered that my washing machine is not anymore. I have to run out to a laundromat if I am going to be ready for the new work week in the morning. I will return to this channel, and if you are still around this conversation in about two hours. Feel free to fill in th |
| 23:45.41 | lorsungcu | all you need to do |
| 23:45.50 | hesco | ok, still here |
| 23:45.50 | lorsungcu | is set a tftp server toy serve new config files |
| 23:45.54 | lorsungcu | even just the defaults |
| 23:46.06 | lorsungcu | and set DHCP options 66 and 150 to point to that server.. |
| 23:46.48 | hesco | where do I set dhcp options 66 and 150? on the phone where I lack the password? or on the appliance router? |
| 23:46.53 | lorsungcu | on your router. |
| 23:46.57 | lorsungcu | whatever is doing DHCP |
| 23:47.57 | hesco | That is worth investigating when I am back. Thanks. Not that familiar with dhcp, will study up to see what I can learn and whether this appliance device allows me that level of control. |
| 23:48.41 | [TK]D-Fender | You don't need DHCP options for that phone to pick up configs |
| 23:49.40 | lorsungcu | what would you recommend? |
| 23:51.05 | hesco | Its a D-Link EBR-2310 and does not seem to let me configure Options 66 and 150. But I have another router I could plug it into. I'm running out of time on this laundromat though. Forgive me, [TK]D-Fender, if you have other ideas I look forward to reading them on my return. Back in two hours. |
| 23:51.14 | [TK]D-Fender | Point to the phone to whatever server you want. |
| 23:51.31 | [TK]D-Fender | hesco: Oh, you've moved on from the Polycom issue? |
| 23:51.45 | lorsungcu | yeah, i was basing that on him being unable to get into the admin menu |
| 23:51.54 | [TK]D-Fender | hesco: What do you have that IS working? :p |
| 23:52.22 | lorsungcu | if he resets it and is still unable to access that menu |
| 23:52.30 | lorsungcu | and DHCP options aren't set |
| 23:52.34 | lorsungcu | not sure what's going on |
| 23:52.46 | lorsungcu | unless he isn't trying to get in directly after the reboot |
| 23:53.08 | lorsungcu | assume that's what you were asking and didn't get an answer on before. |
| 23:54.57 | *** join/#asterisk Phoebus (~Phoebus@pdpc/supporter/active/phoebus) |