| 00:00.20 | leifmadsen | paulc: if the channel goes away, I would indeed expect 'h' to be triggered every time |
| 00:00.25 | leifmadsen | if it didn't before, that was probably a bug |
| 00:02.39 | paulc | leifmadsen: sure, I would do.. but no channels went away (or did they?).. Imagine A is talking to B, C is talking to D, and E is connected to music on hold. AGI redirects E to ChanSpy on A's channel. All is well. AGI then redirects E to ChanSpy on C's channel. E triggers the h extension at this point. |
| 00:02.52 | *** part/#asterisk WWGD (~WWGD@208.79.14.130) |
| 00:02.52 | paulc | Maybe I'm missing a nuance of the low level internals of chanspy? |
| 00:03.00 | paulc | s/do/too/ |
| 00:03.32 | paulc | in fact, I rely on h to trigger a web service to say "this person doing monitoring has gone away" so the web control panel thing gets updated |
| 00:03.35 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 00:04.18 | paulc | I like the h extension :) I'm just not sure why it's triggering in this scenario, when E is still in progress and is an active call/channel.. does chanspy touch it (in bad places?) when the spying starts? |
| 00:06.09 | *** join/#asterisk sourcode (~code@ppp-58-8-220-83.revip2.asianet.co.th) |
| 00:09.23 | *** join/#asterisk AlcornTech (chris@pool-71-171-143-33.chi01.dsl-w.verizon.net) |
| 00:11.03 | AlcornTech | I've got a fresh CentOS 5.5/Asterisk 1.4 install (via yum). This is my first time attempting to use Zaptel/DAHDI and I'm getting "No channel type registered for 'Zap'". I know I'm missing something simple. Can someone throw me a bone? |
| 00:14.39 | *** join/#asterisk bjhaid (~abejide@41.220.69.9) |
| 00:21.49 | pabelanger | AlcornTech: Zaptel is no more, use DAHDI. |
| 00:21.53 | *** join/#asterisk coppice (~chatzilla@190.131.92.116.dyn.pacific.net.hk) |
| 00:22.19 | shido6 | hey what happens when I enter sip set debug ip 1.2.3.4 and do another ip in another session will asterisk console show both? |
| 00:22.29 | shido6 | or just the last ip? |
| 00:22.35 | Tozz_ | why dont u just try it? |
| 00:27.07 | AlcornTech | @pabelanger: I've got DAHDI installed (not zaptel)...do I need to use a different prefix instead of 'Zap' to specify DAHDI channels? |
| 00:27.22 | pabelanger | AlcornTech: Yes |
| 00:27.24 | pabelanger | DAHDI |
| 00:27.34 | AlcornTech | @pabelanger: will try. thanks much. :) |
| 00:28.37 | *** join/#asterisk ghostmediapro (~Trupsalms@71.23.182.164) |
| 00:29.05 | ghostmediapro | any reason why after the upgrade to 1.8 I wouldn't have the gtalk.conf or jabber.conf files under /etc/asterisk ? I also appear to be missing the chan_gtalk and res_jabber |
| 00:29.35 | bmoraca_work | ghostmediapro: you didn't select to build them in your "make menuselect"? |
| 00:32.55 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 00:33.17 | *** join/#asterisk shadowapex (~William@adsl-99-107-163-194.dsl.lsan03.sbcglobal.net) |
| 00:33.20 | ghostmediapro | bmoraca_work: how can i do i remake config |
| 00:33.42 | bmoraca_work | having the config files there does nothing for you if the modules didn't get built |
| 00:33.44 | ghostmediapro | Bmoraca_work: how can i instill them now sign that step has been missed |
| 00:33.59 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-170-15.dsl.stlsmo.sbcglobal.net) |
| 00:34.23 | ghostmediapro | Bmoraca_work: please assist |
| 00:34.24 | bmoraca_work | run "make menuselect" again, select the modules, run "make" then run "make install" then copy the sample config files out of the ./config/ directory in the asterisk source folder |
| 00:34.51 | bmoraca_work | any more help than that and i'll need a prepayment for my consulting service |
| 00:35.05 | ghostmediapro | bmoraca_work: am i running this from just root or do i have to be inside a directory |
| 00:35.17 | bmoraca_work | facepalms |
| 00:36.22 | shadowapex | Hey everyone, I'm trying to get the "call-limit" parameter working with Asterisk using a realtime database configuration, but it doesn't actually work. If I set "rtcachefriends=yes", then the call-limit works flawlessly, but unfortunately working in cached mode forces us to have to do a "sip reload" everytime there is a change to the database. Has anyone gotten "call-limit" to work with realtime? |
| 00:36.26 | ghostmediapro | bmoraca_work: i'm running it from the etc/asterisk directory with no result |
| 00:36.41 | bmoraca_work | that's not where you're supposed to run it... |
| 00:37.15 | bmoraca_work | shadowapex: you can use "sip prune realtime peer" to reset just a single realtime peer |
| 00:37.31 | bmoraca_work | shadowapex: you are, however, running in to a limitation of realtime sip. |
| 00:37.38 | bmoraca_work | i believe it's well documented. |
| 00:37.58 | ghostmediapro | bmoraca_work: the usr/src/asterisk directory does not exist |
| 00:38.07 | hardwire | palmfaces. |
| 00:38.16 | bmoraca_work | then you must not have installed asterisk via source, in which case I cannot help you |
| 00:38.30 | bmoraca_work | contact whoever built the packages you installed from and ask them to add jabber and gtalk |
| 00:39.00 | ghostmediapro | asterisknow |
| 00:39.05 | ghostmediapro | iso |
| 00:39.08 | hardwire | haha |
| 00:39.10 | hardwire | screwed |
| 00:39.13 | hardwire | *ahem* |
| 00:40.19 | shadowapex | bmoraca_work: That might be something we can use... but there's no way to get the call-limit to work with a realtime configuration? |
| 00:40.55 | hardwire | just use groups.. it will make you happy. |
| 00:41.01 | hardwire | groups + func_odbc + joy. |
| 00:41.02 | bmoraca_work | shadowapex: like I said, I think it's a well documented limitation. what are you trying to use call-limit for? hints? or do you actually want to limit the number of calls? |
| 00:41.16 | shadowapex | bmoraca_work: I've read a lot online, but people have been mainly referring to asterisk 1.4/1.6. It's still not available for 1.8? |
| 00:41.38 | shadowapex | bmoraca_work: I just want to be able to limit the numeber of simultaneous calls on a per user basis. |
| 00:41.41 | bmoraca_work | not sure about 1.8. i've had some problems with 1.8 |
| 00:41.47 | bmoraca_work | shadowapex: then use groups. |
| 00:41.53 | bmoraca_work | shadowapex: much more flexible |
| 00:42.16 | shadowapex | bmoraca_work: We're also using this in conjunction with a2billing. |
| 00:42.24 | bmoraca_work | ick |
| 00:42.27 | bmoraca_work | no help here, then |
| 00:42.29 | LemensTS | Anyone mess with the Kirk Dect phones that use the Kirk Wirless server with asterisk? |
| 00:42.32 | hardwire | about the only time I've had use for call-limit was when the limit was 1 and I was accessing a single channel on a GSM gateway |
| 00:42.44 | hardwire | LemensTS: rolling on dough are we? |
| 00:42.47 | hardwire | on/in |
| 00:43.03 | bmoraca_work | those are the polycom ones, right? |
| 00:43.22 | *** join/#asterisk MrTelephone (~MrTelepho@S01060018e7e1c1b7.ls.shawcable.net) |
| 00:43.25 | LemensTS | hardwire: I'm not sure....looking at building systems at larger houses with wireless WIFI phones (cant run new cabling) |
| 00:43.30 | shadowapex | bmoraca_work: I'm not really familiar with Asterisk groups, but do you think it'd be possible to implement with a2billing? |
| 00:43.38 | hardwire | LemensTS: buy everybody iphones/androids :) |
| 00:43.39 | hardwire | cheaper. |
| 00:43.41 | MrTelephone | Are there any programmers at the keyboard tongiht? |
| 00:43.46 | bmoraca_work | shadowapex: anything's possible. |
| 00:44.03 | bmoraca_work | shadowapex: the question is how much of a2billing do you have to rewrite |
| 00:44.14 | shadowapex | bmoraca_work: Except realtime w/ call-limit apparently :( |
| 00:44.28 | bmoraca_work | shadowapex: yeah, but there are workarounds. |
| 00:44.29 | shadowapex | bmoraca_work: Yeah, I guess that'll have to be something I look into. |
| 00:44.37 | hardwire | shadowapex: a2billing has support to limit to 1 or infinite.. nothing between |
| 00:44.44 | hardwire | however you could easily hack it to support arbitrary amounts |
| 00:44.50 | hardwire | uses a2billing (*sigh*) |
| 00:44.57 | bmoraca_work | a2billing never impressed me much |
| 00:45.02 | shadowapex | bmoraca_work: Yeah, I've noticed that setting in a2billing. |
| 00:45.05 | hardwire | I had to get something up quick. |
| 00:45.17 | paulc | leifmadsen: I added a note to https://issues.asterisk.org/view.php?id=18675 regarding that issue we were talking about - you'd updated the ticket earlier today.. maybe that helps clarify some more.. |
| 00:45.19 | MrTelephone | what does a2billing do? |
| 00:45.31 | hardwire | shadowapex: using func_odbc you can easily have outbound calls limited based on another table related to account numbers. |
| 00:45.52 | bmoraca_work | i use func_odbc extensively in my asterisk-based softswitch |
| 00:45.55 | bmoraca_work | it's loverly |
| 00:45.58 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 00:46.15 | hardwire | juse make an odbc call to your database to figure out the limits.. then do another lookup in the a2billing call detail records to see how many channels are up via a count(). |
| 00:46.18 | hardwire | then you're done |
| 00:46.34 | hardwire | freaking apple.. now I can't jailbreak a 3gs untethered. |
| 00:46.36 | hardwire | annoying. |
| 00:47.13 | shadowapex | hardwire: Yeah, it looks like I'll need to do something like that in order to get the functionality I need. |
| 00:47.15 | MrTelephone | is there a command line in asterisk that tells you what the max channel usage was? like how many channels were used on your PRI? |
| 00:47.31 | hardwire | MrTelephone: groups works well for that |
| 00:47.45 | MrTelephone | yeah my outgoing circuit is g1 |
| 00:47.57 | hardwire | so for the first priority... Set(GROUP=pri) |
| 00:47.58 | bmoraca_work | MrTelephone: not really. you can use asterisk-stat-v2 to do some CDR analysis though |
| 00:48.02 | MrTelephone | I never know if my pipe gets congested.. |
| 00:48.05 | shadowapex | hardwire: I'll look into func_odbc, asterisk groups, and realtime pruning and see which will be the most efficient solution. I just wanted to see if there was an easier route. |
| 00:48.12 | hardwire | then the second NoOp(PRI IS NOW AT {$GROUPCOUNT() |
| 00:48.14 | hardwire | or whatever |
| 00:48.19 | shadowapex | hardware/bmoraca_work: Thanks for your input guys. |
| 00:48.19 | hardwire | then scan your logs |
| 00:48.20 | bmoraca_work | MrTelephone: if you're looking for "right now" readings, "core show channels" works pretty well |
| 00:48.22 | MrTelephone | ok |
| 00:48.34 | MrTelephone | yeah I just want it to alarm if more than X channels are used. |
| 00:48.49 | bmoraca_work | meh, just tell the customers to deal with it |
| 00:48.59 | bmoraca_work | or buy you a second PRI |
| 00:49.38 | paulc | MrTelephone: I have a PHP script somewhere that connects to the manager port and gets the channel/call count, then outputs the number in a format that Cacti likes.. we can then graph call counts over time, as well as alert at certain levels.. could probably dig it out if you were interested.. |
| 00:49.48 | MrTelephone | for sure. My over subscription is pretty high. 10:1 probably |
| 00:50.35 | MrTelephone | paulc, nice. Did you make it? |
| 00:50.50 | MrTelephone | GROUPCOUNT sounds interesting |
| 00:51.05 | paulc | Yup - we needed to monitor activity on a click to call type server, see how much we were pumping through it |
| 00:51.22 | MrTelephone | my bell rep said there is no such thing as PRI REVERSE CHARGE. Does anyone know about these indication? |
| 00:52.02 | MrTelephone | paulc, if you want to send it I'll take a look at it |
| 00:52.26 | paulc | MrTelephone msg me your email addy and I'll shoot it off now before I head out |
| 00:53.44 | *** join/#asterisk Freeaqingme (~Freeaqing@83.160.112.249) |
| 00:54.13 | MrTelephone | that adtran company sure does business well. I told them what I wanted to do and they gave me estimates, advice, and had a salesperson in my area call me the next day. You rarely get service like that anymore. |
| 00:56.01 | *** part/#asterisk shadowapex (~William@adsl-99-107-163-194.dsl.lsan03.sbcglobal.net) |
| 00:56.49 | ghostmediapro | bmoraca_work: would reinstalling asterisk from source do the trick |
| 00:57.30 | LemensTS | Besides the iphone/android, what is your favorite IP WIFI phone? |
| 00:58.29 | ghostmediapro | hardwire: could installing from source work |
| 00:59.06 | MrTelephone | eyebeam on a laptop |
| 00:59.36 | ghostmediapro | eyebeam i believe to be a excellent app |
| 01:00.13 | LemensTS | Got to be handheld phones for the customer. |
| 01:00.42 | ghostmediapro | is it sip base |
| 01:01.04 | MrTelephone | wifi phones for the office or for public roaming? |
| 01:01.09 | ghostmediapro | android and iphone has sip apps i believe |
| 01:01.27 | MrTelephone | what about ipod touch? |
| 01:01.31 | *** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net) |
| 01:02.10 | MrTelephone | this whole technology thing on the go is getting way out of hand |
| 01:02.23 | ghostmediapro | correct me if i am wrong be i believe that the skype phone has a hack to use it as a sip phone without paying supcribtion, but of course you have to buy it full price for find referbished ones |
| 01:02.43 | ghostmediapro | almost all apps for iphone can be used on each other |
| 01:03.10 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-170-15.dsl.stlsmo.sbcglobal.net) |
| 01:04.12 | ghostmediapro | @hardwire and @bmoraca_work no to spam the room, but would recompeling from soucre help, be for i jump in start the process |
| 01:04.44 | hardwire | ghostmediapro: I'd recommend just not using asterisknow and using say.. debian squeeze. |
| 01:05.12 | hardwire | get you going pretty quick so you can think about what you want to do instead of worrying about asterisk installation |
| 01:05.21 | ghostmediapro | hardwire i had issues youing piaf without touching anything, sh** JUST DIDN'T WORK |
| 01:05.34 | hardwire | ? |
| 01:05.39 | hardwire | not sure I understand |
| 01:05.48 | ghostmediapro | PBX IN A FLASH |
| 01:05.54 | ghostmediapro | THE RECOMMENED INSTALL |
| 01:05.55 | hardwire | youing? |
| 01:05.58 | MrTelephone | INSTA PBX |
| 01:06.03 | hardwire | using! |
| 01:06.07 | CoffeeIV | version 1.6.2 (ubuntu) here - I am trying to track down a bunch of failed dns lookups of "dynamic" in my logs, and I think it is host = dynamic in my users.conf, when it should be dynamic = yes or something like that. Where can I read documentation about what is allowed in users.conf ? |
| 01:06.21 | ghostmediapro | I HAD ISSUES WITH |
| 01:06.30 | hardwire | with blank? |
| 01:06.36 | hardwire | :P |
| 01:06.41 | ghostmediapro | PIAF |
| 01:06.44 | MrTelephone | 5000 for a 16 phone nortel system, 10,000 for an avaya 48 phone system, INSTA PBX by Asterisk (C) - PRICELESS - |
| 01:06.46 | hardwire | pee in a fire? |
| 01:06.47 | MrTelephone | literally |
| 01:06.48 | hardwire | what's going on! |
| 01:06.49 | hardwire | haha |
| 01:07.01 | hardwire | erm.. what does debian have to do with PIAF? |
| 01:07.17 | ghostmediapro | SORRY, SOMEWHAT OF A NEWBIE SO I NEED SOME SORT OF GUI, WHAT DABIAN DO YOU REFER |
| 01:07.24 | hardwire | the debian kind of debian. |
| 01:07.33 | ghostmediapro | UBUNTU |
| 01:07.36 | MrTelephone | DABIAN? |
| 01:07.36 | hardwire | no gui. un newb yourself. |
| 01:07.43 | hardwire | it'll be worth it |
| 01:07.47 | MrTelephone | try hatred |
| 01:08.01 | MrTelephone | hat red linux enterprise |
| 01:08.38 | hardwire | ghostmediapro: the asterisk book works really well in the newb circles. |
| 01:09.12 | hardwire | the guis are almost impossible to support in this channel |
| 01:09.23 | hardwire | so many problems in so many different ways. |
| 01:09.25 | MrTelephone | wouldn't the gui's be easier? |
| 01:09.36 | MrTelephone | Plug and Play |
| 01:09.45 | hardwire | Point of view I guess.. I dislike using the GUI because it doesn't do anything I want. |
| 01:10.01 | hardwire | I have to spend days in code juts to make it do what I want it to do just to make it easier on someone else that will never use it. |
| 01:10.29 | hardwire | bbl |
| 01:10.35 | MrTelephone | yeah even command line config editing still won't get you what you want :( |
| 01:10.51 | MrTelephone | source code crunching seems to be the only way :( |
| 01:10.58 | hardwire | I think saying things like "command line config editing" is strange as well. |
| 01:11.24 | MrTelephone | I guess it could be said a differnet way |
| 01:11.54 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 01:14.47 | ghostmediapro | I CAN DO CLI, BUT COULD I STILL YOU FREEPBX WITH IT |
| 01:14.48 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 01:19.23 | diemos | No need to yell. |
| 01:27.23 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 01:27.51 | ghostmediapro | @hardwire & @bmoraca_work configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 01:31.50 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 01:32.32 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:43.21 | *** join/#asterisk toresbe (toresbe@simula.gunkies.org) |
| 01:43.23 | toresbe | hey guys |
| 01:43.51 | toresbe | quick question - I'm guessing it's more of a Cisco question, but it's worth a shot |
| 01:44.20 | toresbe | anyone have any experience making a FXS module on a Cisco box talk to Asterisk? It only seems to support H.323, not SIP. |
| 01:47.24 | pabelanger | ~book |
| 01:47.24 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 01:48.12 | pabelanger | ghostmediapro: ^ I suggest you read chapter 3 |
| 01:51.46 | ghostmediapro | GOT IT @channel, then i'm Gone For This Post http://michigantelephone.wordpress.com/2010/12/21/how-to-use-google-voice-for-free-calls-on-an-asterisk-1-8freepbx-2-8-system-the-easy-way/ what iso would anyone recommend |
| 01:53.04 | leifmadsen | ugh, something is happening on my server downtown... |
| 01:54.40 | *** join/#asterisk tyman_ (~tyler@173-12-219-179-Fresno.hfc.comcastbusiness.net) |
| 01:54.49 | AlcornTech | any idea what could cause "dahdi show status" to show 2 cards as "OK", but "dahdi show channels" only lists the pseudo channel? |
| 01:54.51 | leifmadsen | oh well, I'll just hope something is getting overloaded in the rack that is affecting my server. Going to enjoy the rest of the evening and will deal with that later.... |
| 01:58.00 | leifmadsen | ouch... oomkiller for vmware |
| 01:58.01 | leifmadsen | nice |
| 02:00.48 | *** join/#asterisk Kumbang (~kumbang@61.94.142.212) |
| 02:07.40 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 02:11.59 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 02:16.39 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 02:39.51 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 02:39.58 | IsUp | hello |
| 02:40.28 | IsUp | i am trying to do "voice reminder" thing, basicly i am using call files to call my cellphone |
| 02:40.49 | IsUp | and playing files, they are not music, just speech |
| 02:40.59 | IsUp | but i cant decide on codec, gsm is really bad |
| 02:42.12 | IsUp | i am recording my voice on Windows, 44100Hz Mono Wav |
| 02:43.39 | p3nguin | I would use wav, but 44100 Hz will not work. |
| 02:43.54 | AlcornTech | can anyone spare a few minutes to help me with PRI? "pri show span 1" yields "Status: Provisioned, In Alarm, Down, Active"... not sure how to come out of alarm or out of down status. |
| 02:44.17 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 03:08.38 | bdfoster | hey uh anyone around here? |
| 03:08.54 | bdfoster | just tried to recompile asterisk 1.8.2.3 |
| 03:09.35 | bdfoster | now i have a screen that says "Game over. You Lose!" |
| 03:10.41 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 03:12.25 | IsUp | bdfoster: whats going on |
| 03:13.14 | bdfoster | i think the tarball is either screwed up or your site was hacked |
| 03:13.32 | bdfoster | http://endigotech.com/Screenshot.png |
| 03:14.00 | bdfoster | s/your/the |
| 03:14.38 | bdfoster | IsUp, any suggestions other than to call it a loss and redo my whole server? |
| 03:15.18 | IsUp | LOL |
| 03:15.19 | IsUp | wtf |
| 03:15.35 | bdfoster | yep |
| 03:15.51 | IsUp | gimme exact url of tarball |
| 03:16.10 | bdfoster | http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.2.3.tar.gz |
| 03:17.31 | bdfoster | this is my exact steps |
| 03:17.36 | bdfoster | amportal stop |
| 03:17.36 | bdfoster | cd /usr/src/libpri* |
| 03:17.37 | bdfoster | make uninstall |
| 03:17.37 | bdfoster | make clean |
| 03:17.37 | bdfoster | make install |
| 03:17.37 | bdfoster | cd /usr/src/ss7* |
| 03:17.39 | bdfoster | make uninstall |
| 03:17.41 | bdfoster | make clean |
| 03:17.43 | bdfoster | make install |
| 03:17.45 | bdfoster | cd /usr/src/as*1.8.2.3* |
| 03:17.47 | bdfoster | make uninstall |
| 03:17.49 | bdfoster | make clean |
| 03:17.51 | bdfoster | ./configure |
| 03:17.53 | bdfoster | make menuconfig |
| 03:17.55 | bdfoster | contrib/scripts/get_mp3_source.sh |
| 03:17.57 | bdfoster | contrib/scripts/get_ilbc_source.sh |
| 03:17.59 | bdfoster | make |
| 03:18.01 | bdfoster | make install |
| 03:18.04 | Freeaqingme | bdfoster, please, dont paste here |
| 03:18.05 | bdfoster | amportal start |
| 03:18.12 | IsUp | bdfoster: use pastebin please |
| 03:18.13 | Freeaqingme | use something like pastie.org, and just paste the link here |
| 03:18.44 | drmessano | bdfoster, I would try again with a water type like wartortle or psyduck. Sounds like you chose the wrong Pokemon |
| 03:18.52 | bdfoster | haha |
| 03:20.25 | bdfoster | not funny seeing my whole server that i've spent a great deal of time getting perfect is going down the drain |
| 03:29.11 | AlcornTech | bdfoster: rm -rf / ? |
| 03:31.42 | mzb | AlcornTech: grow up |
| 03:31.59 | mzb | bdfoster: make distclean |
| 03:32.01 | bdfoster | AlcornTech, don't paste commands that will screw up one's system |
| 03:32.04 | mzb | (or clean) |
| 03:32.16 | bdfoster | mzb, still booting, and i need a cig |
| 03:32.52 | AlcornTech | wow. I'm an adult. I just assumed everyone knew what that command does. And if a system is really that screwed up, sometimes you just have to start over. no offense. |
| 03:33.05 | mzb | bullshit |
| 03:33.25 | bdfoster | omg |
| 03:33.44 | bdfoster | have you seen some of the people that come in to #freepbx? |
| 03:34.09 | bdfoster | have you seen some of the people that go into #fedora, #centos, #ubuntu and the like? |
| 03:34.25 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
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| 04:03.23 | ghostmediapro | icbsearch.c :138: warning: operation on 'pp' may be undefined |
| 04:10.18 | ChannelZ | try plunger then flush again |
| 04:11.02 | ghostmediapro | ChannelZ: Huh |
| 04:11.30 | p3nguin | "Huh," indeed. |
| 04:12.20 | ChannelZ | It's a PP joke. You people have no sense of humor whatsoever |
| 04:12.59 | ghostmediapro | awww good one |
| 04:13.11 | ghostmediapro | taking mental note |
| 04:13.51 | *** join/#asterisk kuku (~kuku@c-24-13-139-34.hsd1.il.comcast.net) |
| 04:15.50 | ChannelZ | I'm so misunderstood. |
| 04:16.52 | maxagaz | I have asterisk 1.8.2 installed and I've just download 1.8.2.3, is it safe to make install over 1.8.2, or should I uninstall 1.8.2 first ? |
| 04:17.17 | ChannelZ | it's pretty safe |
| 04:17.30 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 04:17.45 | ChannelZ | if you have any extra modules you built separately the installer will notice warn you |
| 04:18.01 | ChannelZ | s/warn/and warn/ |
| 04:22.41 | kuku | If I dont see dahdi in my newly compile 1.8.2.3 ( and I compiled dahdi ) - where did I mess up ? |
| 04:23.02 | maxagaz | ChannelZ: thanks |
| 04:23.32 | p3nguin | Maybe you didn't install dahdi, or maybe you didn't install dahdi before you ran configure on the asterisk source. |
| 04:25.02 | kuku | p3nguin: I make, make isntall, make config dahdi. then I make clean asterisk, ./configure, make, make install. |
| 04:33.23 | ChannelZ | you mean chan_dahdi wasn't built? |
| 04:34.52 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 04:36.25 | kuku | I restarted the box and it started working :) |
| 04:36.46 | kuku | Thanks. Now I can't get the phones to pickup tftp. I never have luck with tftp |
| 04:36.49 | ChannelZ | universal fix |
| 04:44.52 | kuku | Any help on this tftp |
| 04:45.10 | kuku | netstat shows its running, phones are pointed to it, but no request come in |
| 04:45.21 | kuku | ( when the phone restarts ) |
| 05:01.08 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
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| 05:43.41 | ChannelZ | kuku: What phones? if they ain't askin', it's hard to debug the tftp server |
| 05:47.52 | citywok | hmm. so i need to be able to execute a system command via a call file. how the heck do i do this? |
| 05:50.06 | goddva | citywok: do know how to create a call file, then just use System as application..!? |
| 05:50.41 | goddva | dial you could simply dial sleep extension with local |
| 05:50.48 | citywok | goddva: never used call files before. but everything i find wants to be bound to a channel... i don't really need a channel to do anything, and teh basic callfile i made didnt do anything. |
| 05:51.15 | citywok | i tried to bind to channel that would call my cellphone, and then the command would run system, and my command, but it didnt do anything lol |
| 05:52.02 | goddva | console output? |
| 05:52.17 | goddva | did asterisk try to do something? |
| 05:52.25 | goddva | is the call file still there (in the folder) |
| 05:52.34 | citywok | ait, i had a typo. maybe that was why. |
| 05:52.38 | citywok | s/ait/wait |
| 05:53.45 | citywok | oooh, excellent itw orked! |
| 05:53.58 | citywok | man, i can't believe i just hacked my own server via PHP & Asterisk. lolol. |
| 05:53.58 | goddva | perfect - nothing better... |
| 05:54.38 | citywok | ssh died on a colo'd box for some reason, so i wanted to bump it. i can upload php files via a web app, and run them which can exec crap, which can spool call files. lol. |
| 05:54.44 | citywok | i would have used the ami but the manager isn't turned on. |
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| 06:03.01 | goddva | citywok: nice solved :) |
| 06:03.19 | goddva | citywok: try to do that with a MS box :P |
| 06:03.19 | citywok | sort of. it hasn't gone down completely yet. something may be stuck. |
| 06:05.57 | Chewie71 | Need a little help getting my call routing working... I have a Digium AEX410 w/ 2xfxo 2xfxs ports... I'm setup using the AsteriskNOW image and FreePBX. Have a pots line plugged in to one of the fxo ports. When I call in it rings a few times...but then a female voice comes on and says "bye" and the call drops. It doesn't ring through to any extensions. |
| 06:11.26 | citywok | it finally rebooted. |
| 06:20.40 | Chewie71 | getting this error in the /var/log/asterisk/full log... |
| 06:20.52 | Chewie71 | [Jan 31 12:18:09] VERBOSE[32264] pbx.c: == Starting DAHDI/4-1 at from_analog,s,1 failed so falling back to exten 's' |
| 06:21.55 | WIMPy | Analog is evil. |
| 06:22.04 | WIMPy | And you don't seem to have an s extension. |
| 06:22.33 | WIMPy | But falling back to s from s seems an interesting idea. |
| 06:23.05 | drmessano | Will it go round in circles? |
| 06:23.08 | Chewie71 | No 's' extension....I have a SIP extension labeled '111' |
| 06:23.11 | drmessano | (Billy Preston) |
| 06:24.07 | drmessano | Chewie71, your ANY/ANY route in Inbound Routes needs to go somewhere |
| 06:24.21 | drmessano | It wont magically route to any extension from an Analog line |
| 06:24.23 | WIMPy | You need a WaitExten at s in order to be able to dial. |
| 06:28.31 | Chewie71 | drmessano: ok...I have a route "from ANY DID/ ANY CID" and it's destination is set to extension 111 |
| 06:29.27 | Chewie71 | but I get the "bye" message and hangup |
| 06:30.02 | *** join/#asterisk Corydon76-home (four@c-69-137-80-31.hsd1.tn.comcast.net) |
| 06:30.02 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 06:31.27 | drmessano | Chewie71, which context= do you have set in chan_dahdi.conf |
| 06:31.42 | drmessano | or chan_dahdi_groups.conf if you're using the DAHDI module |
| 06:32.30 | Chewie71 | "from_analog" |
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| 06:43.34 | drmessano | should be from-pstn or from-trunk to route into the FreePBX dialplan |
| 06:43.46 | drmessano | from-pstn is preferred for DAHDI/ZAPTEL |
| 06:43.54 | drmessano | (though it goes to the same place) |
| 06:49.10 | Chewie71 | drmessano: ah! didn't know that name mattered. :P It rang through to the softphone extension on my laptop that time. Yea! But no audio in the call.... Had the same problem calling between two SIP extensions. The calls worked, but no audio. |
| 06:50.02 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:52.24 | drmessano | ~sipnat |
| 06:52.24 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 06:52.29 | drmessano | Try that |
| 06:52.36 | drmessano | Of course |
| 06:52.40 | drmessano | You have FREEPBX |
| 06:52.49 | drmessano | So you need to use the SIP Settings Module for these settings |
| 06:54.34 | Chewie71 | OK....let me play with that for a bit... |
| 06:59.25 | *** join/#asterisk jhardy (~jveezee@S010600137214c323.vf.shawcable.net) |
| 06:59.49 | *** join/#asterisk Siblor (~putthibon@58.136.24.13) |
| 06:59.54 | Siblor | Hello, does it possible to stream live conversation in asterisk channel to Java web application while Originate to application Record? |
| 07:00.15 | jhardy | Anyone running asterisk in a virtual machine while using vlan interfaces on the host os? |
| 07:00.34 | jhardy | (virtualbox) |
| 07:01.20 | Chewie71 | drmessano: must be getting tired. I installed the Asterisk SIP Settings module but now can't find it in FreePBX... |
| 07:01.40 | Chewie71 | nevermind..... |
| 07:05.02 | Chewie71 | drmessano: tks...worked great... :) Now I can go to bed. |
| 07:05.38 | drmessano | Sweet |
| 07:05.58 | *** join/#asterisk mayfield (~mayfaild@76-250-152-203.lightspeed.snantx.sbcglobal.net) |
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| 07:36.52 | *** join/#asterisk shapr (~shapr@nat/digium/x-ldxfzocnumzmkmqp) |
| 07:37.18 | shapr | What's the right way to configure Asterisk if both phones and server are behind different NAT firewalls? Does that require a STUN server? |
| 07:38.59 | p3nguin | ~sipnat |
| 07:38.59 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 07:39.19 | p3nguin | The voip-info page should cover your case. |
| 07:39.23 | *** join/#asterisk Kumbang (~kumbang@180.245.174.79) |
| 07:39.38 | p3nguin | And no, it usually does not need STUN. |
| 07:40.48 | shapr | The problem I'm seeing is that SIP Via header for the phones instead shows the externip for the Asterisk server. |
| 07:41.23 | shapr | reads the link |
| 07:41.33 | p3nguin | Make sure you configure the nat setting appropriately for both the general section AND the peer definition in your sip.conf. |
| 07:41.53 | p3nguin | typically nat=yes and canreinvite=no |
| 07:42.22 | shapr | aha, I bet that's it |
| 07:42.48 | p3nguin | The nat setting in the general section is for when your system is behind a NAT... |
| 07:43.09 | p3nguin | and the nat setting in each peer entry is for when those peers are behind their own NAT. |
| 07:43.39 | p3nguin | They are actually exclusive. |
| 07:44.00 | shapr | exclusive? as in, they don't affect each other? |
| 07:44.13 | p3nguin | I mean mutually exclusive. |
| 07:44.42 | p3nguin | They do not affect each other. You can set the peer's nat to yes and your own to no or your own to yes and the peer to no... etc. |
| 07:45.13 | shapr | That makes sense, thanks. |
| 07:46.15 | p3nguin | But if your own system is behind NAT, you need to configure more than just the nat setting. You have to configure the localnet and the externaddr/externip or the externhost as well. |
| 07:46.48 | p3nguin | Those additional parameters are not for peers. |
| 07:50.58 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 07:53.55 | b14ck | yo |
| 07:56.54 | ChannelZ | yo-yo |
| 07:57.09 | b14ck | how goes it |
| 07:57.28 | ChannelZ | not too terrible. You? |
| 07:59.39 | b14ck | same |
| 07:59.39 | b14ck | :) |
| 07:59.52 | b14ck | Just doing some coding (asterisk, actually). |
| 07:59.55 | b14ck | Refactoring some code |
| 08:00.03 | b14ck | Been a while =p |
| 08:00.08 | shapr | b14ck: agi script? or what? |
| 08:00.15 | b14ck | no, some dialplan |
| 08:00.26 | b14ck | We have some pretty extensive dialplan in use. |
| 08:00.31 | b14ck | With thousands of users :o |
| 08:01.15 | ChannelZ | swooty |
| 08:01.28 | b14ck | I wanna open source the base for it. |
| 08:01.38 | b14ck | We developed like an asterisk framework for dialplan. |
| 08:01.41 | shapr | Yow, wouldn't it be easier to do some external scripting? |
| 08:01.43 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 08:01.44 | b14ck | So that you can just write little modules and they plug into it. |
| 08:01.57 | b14ck | Well, we do external stuff too, it's a complex system. |
| 08:02.26 | b14ck | i wrote a c agi library a while back |
| 08:02.29 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:02.30 | b14ck | but i dont use it anymore |
| 08:02.39 | b14ck | https://github.com/rdegges/cagi |
| 08:03.00 | shapr | I'm just getting starting with Haskell AGI... I know Haskell, and some Asterisk, but not AGI. |
| 08:03.03 | b14ck | most of our stuff now is python/django/dialplan/ami |
| 08:03.10 | b14ck | agi is really simple |
| 08:03.21 | b14ck | you just do read/writing via STDIN/STDOUT |
| 08:03.27 | b14ck | and there's only a few commands you can use |
| 08:04.05 | shapr | What about other external scripting APIs like AEL or so? |
| 08:04.42 | b14ck | Well, if you want to use a language that is not asterisk-specific, then you should probably use the AMI. |
| 08:04.52 | b14ck | You can run your code somewhere else, and talk to asterisk via TCP. |
| 08:04.55 | shapr | Is AEL sort of like AppleScript such that each module can export commands? |
| 08:05.03 | shapr | aha, that sounds nice |
| 08:05.04 | b14ck | ael is like dialplan, you just write some basic logic |
| 08:05.16 | b14ck | most people use dialplan for code that runs locally on the server |
| 08:05.20 | b14ck | like core call routing, etc. |
| 08:05.29 | b14ck | and then they use ami for code that runs remote that needs to interact with asterisk |
| 08:06.04 | shapr | That makes sense. Still seems weird to me that you can't script Asterisk in Python or some other common extension language. |
| 08:06.14 | b14ck | you can (using agi) |
| 08:06.23 | b14ck | But there's overhead of launching the interpreter everytime you break off. |
| 08:06.34 | b14ck | In my situation, it's too much load on the server to do that consistently. |
| 08:06.45 | b14ck | And it's also kinda annoying, since dialplan can do everything agi can. |
| 08:06.49 | b14ck | Except simpler =p |
| 08:06.56 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:07.11 | shapr | So there must be FastAGI or something that keeps an interpreter in memory? |
| 08:07.31 | b14ck | Well, you could do some IPC stuff to hack it. |
| 08:07.36 | b14ck | But it's not really worth the trouble IMO |
| 08:07.49 | b14ck | dialplan is a simple language, and it's really straightforward to learn and use |
| 08:07.56 | b14ck | and asterisk uses it natively, which is nice |
| 08:09.27 | shapr | I've done lots more Python than dialplan, but I can see that dialplan makes more sense for asterisk-only use. |
| 08:09.37 | b14ck | yeah |
| 08:09.40 | b14ck | I do python too ^^ |
| 08:09.48 | b14ck | Are you doing asterisk stuff for fun, or work? |
| 08:10.03 | shapr | both |
| 08:10.11 | shapr | What about you? |
| 08:10.11 | b14ck | oh cool, what are you building? |
| 08:10.14 | b14ck | same |
| 08:10.25 | shapr | At work I just support asterisk |
| 08:11.33 | shapr | I'd like to build something nifty with asterisk... |
| 08:12.11 | b14ck | oh cool |
| 08:12.13 | b14ck | where do you work? |
| 08:12.20 | b14ck | i used to do that |
| 08:12.24 | b14ck | that's how i was introducted to it |
| 08:12.28 | shapr | The biggest frustration I have with Asterisk is that video codec patents prevent it from being a general purpose media/transcoding application. |
| 08:12.30 | b14ck | *introduced |
| 08:12.35 | shapr | I work at Digium. |
| 08:12.38 | b14ck | oh, hah |
| 08:12.45 | shapr | :-) |
| 08:12.55 | b14ck | I used to work for fonality a couple years ago =p |
| 08:13.02 | shapr | How was that? |
| 08:13.05 | b14ck | meh |
| 08:13.09 | b14ck | stressful |
| 08:13.10 | b14ck | heh |
| 08:13.31 | shapr | Digium is really fun... lots of work, but awesome people. |
| 08:13.46 | b14ck | cool |
| 08:13.56 | diemos | Ah, you work there too? |
| 08:14.06 | b14ck | nope |
| 08:14.13 | b14ck | err |
| 08:14.15 | b14ck | whoops |
| 08:14.19 | shapr | is confused |
| 08:14.25 | shapr | diemos: Where do you work? |
| 08:14.26 | diemos | I meant shapr; I went back to Huntsville myself for SV training. |
| 08:14.58 | shapr | diemos: Do you work at SV? |
| 08:15.03 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 08:15.15 | diemos | No, I'm a reseller. |
| 08:15.45 | shapr | I passed the dCAP after taking the Asterisk Advanced class, that was fun... and LOTS of brain stuffing. |
| 08:16.19 | b14ck | what's dcap? |
| 08:16.34 | shapr | digium certified asterisk professional, iirc |
| 08:16.41 | diemos | ^ |
| 08:16.43 | b14ck | oO |
| 08:16.47 | b14ck | Digium is pretty big now |
| 08:17.14 | shapr | And yet, a fun workplace |
| 08:17.14 | diemos | You should see their building, they're planning on adding wings onto it to make it look like an actual asterisk. |
| 08:17.23 | shapr | looks at the building |
| 08:17.23 | b14ck | woa |
| 08:17.26 | b14ck | i had no idea it was that big |
| 08:17.41 | diemos | And their lunch room overlooks basically the entire town haha. |
| 08:17.59 | carrar | PICS or it never happen! |
| 08:18.07 | carrar | PICS of Katty Eating Cake |
| 08:18.09 | diemos | I have pics on my old phone :( |
| 08:18.43 | shapr | Pics taken now would be dark :-P |
| 08:19.23 | shapr | carrar: http://www.flickr.com/photos/kpfleming/1525446306/ |
| 08:19.23 | diemos | shapr: Tell that indie chick at the coffee shop I said hi. |
| 08:19.47 | shapr | diemos: When did you do your SV training? |
| 08:19.49 | diemos | The small cute one. You know what I'm talking about. |
| 08:19.53 | shapr | heh, I do |
| 08:20.02 | diemos | A few months back, Steve did it. |
| 08:20.44 | shapr | The downside of working at Digium is that I support Asterisk, but I haven't actually *used* it that much. |
| 08:21.01 | b14ck | you'll get the hang of it after a few months |
| 08:21.17 | carrar | People payfor SV training? :) |
| 08:21.24 | diemos | no |
| 08:21.29 | diemos | it's free, you just have to get there. |
| 08:21.31 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 08:21.32 | shapr | I have lots of Linux experience and software dev experience, but telephony is whole 'nother thing. |
| 08:21.35 | carrar | heh |
| 08:22.07 | diemos | shapr: Yeah, now try and learn Asterisk and Freeswitch at the same time while boss is having you do random things every day on top of it. |
| 08:22.13 | shapr | yuck |
| 08:22.17 | diemos | Story of my life. |
| 08:23.25 | shapr | The only thing I truly dislike so far is that fax just hurts, whether analog or sip. |
| 08:23.52 | diemos | shapr: See, that's when we implement a basic POTS line. Headache gone. |
| 08:24.24 | shapr | I'd prefer email attachments... faxing is pain. |
| 08:24.25 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:5d2f:b62a:2eb9:5e64) |
| 08:24.40 | carrar | he shapr you should come here and remodle my bathroom |
| 08:25.02 | shapr | Does your bathroom have a fax machine? |
| 08:25.08 | carrar | is kpfleming your pics? |
| 08:25.30 | carrar | lots of bathroom remodling pics there |
| 08:25.39 | shapr | Nah, that's kpfleming's pics. He's like, head software guy at Digium. |
| 08:26.02 | carrar | oh, tell him to come remodle my bathroom ;) |
| 08:26.03 | shapr | He's a smart guy. |
| 08:26.15 | diemos | carrar: see that little skylight thing at the top? I believe that's where the lunchroom is. |
| 08:26.25 | carrar | always taking vacations |
| 08:26.28 | shapr | diemos: Yup, that's it. The lunchroom is really cool. |
| 08:26.39 | shapr | carrar: Actually, I think he does working vacations. |
| 08:26.48 | carrar | haha |
| 08:26.52 | carrar | no such thing |
| 08:27.18 | carrar | i do that in Japan every year |
| 08:27.36 | carrar | spend a few months there |
| 08:27.41 | carrar | and "work" |
| 08:27.50 | carrar | in the "remote office" |
| 08:27.56 | diemos | Great thing about voip is that you're an extension on an internal system. Bad thing about voip... you're an extension on an internal system -.- |
| 08:27.59 | Tozz_ | I've upgraded from 1.4.18 to 1.4.39.1 last night, and now 3 phones (Linksys SPA942) are registerred according to *, but the phones display an amber light indicating their not registered |
| 08:28.01 | Tozz_ | any ideas? |
| 08:28.35 | diemos | Tozz_: It sounds like they tried to re-register while the service was down. Tried rebooting them? |
| 08:28.40 | Tozz_ | yes |
| 08:28.50 | Tozz_ | the lights go green for about 2s |
| 08:28.58 | Tozz_ | then they go amber, but asterisk shows them as registered |
| 08:29.02 | Tozz_ | when you dial them I get 410 Gone |
| 08:29.03 | diemos | What do the phones' status pages say? |
| 08:29.46 | diemos | http://ip.of.the.phone/admin/basic |
| 08:30.09 | Tozz_ | let me ask, mom |
| 08:30.09 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 08:30.31 | shapr | blinks |
| 08:30.41 | diemos | shapr: We got free lunch when we went there :D |
| 08:30.49 | diemos | sammiches and all! |
| 08:30.55 | Tozz_ | Registration State:FailedNot Reachable |
| 08:30.57 | shapr | Yah, that's one advantage of Digium training, you get awesome local food. |
| 08:31.53 | diemos | Tozz_: Hm, try factory resetting and re-provisioning or re-registering the phones. |
| 08:32.01 | Tozz_ | I dont think its the phones |
| 08:32.06 | Tozz_ | since I've upgraded asterisk last night |
| 08:32.17 | Tozz_ | * is the only change, so I would think the problem lies in there |
| 08:32.18 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 08:32.28 | diemos | But how can you say it's Asterisk if the rest of your phones work (assuming you have more than three). |
| 08:32.52 | Tozz_ | good point |
| 08:33.01 | Tozz_ | maybe its some fucked up SIP ALG feature in $cust router |
| 08:33.42 | Tozz_ | because 'Reg. Contact' shows the external IP, which shouldnt be the case if hes behind nat |
| 08:34.01 | diemos | Tozz_: I have weird things like that happen all the time on both Freeswitch and *. |
| 08:34.13 | shapr | sip aware routers, generally aren't |
| 08:34.13 | diemos | I just toy with it for a while until it works, really. |
| 08:37.06 | wdoekes2 | Tozz_: sip dump? if he does do a poll with e.g. notify after the reg, he may not like the (new) result code |
| 08:40.07 | Tozz_ | wdoekes2 / diemos: it seems disabling SIP ALG in de router solves it |
| 08:40.13 | Tozz_ | a 'feature' which can only be set in the telnet interface |
| 08:40.13 | Nugget | telnet is eeeeeeevil! |
| 08:40.19 | wdoekes2 | hehe |
| 08:40.34 | Tozz_ | oh, didn't fix it ;( |
| 08:40.43 | Tozz_ | it turned the phone into a christmas tree |
| 08:40.47 | *** join/#asterisk _zoom_ (~zero@41.218.35.221) |
| 08:40.59 | wdoekes2 | wow |
| 08:41.30 | _zoom_ | hello, what is the default timeout for ringback |
| 08:41.32 | _zoom_ | ? |
| 08:42.26 | shapr | diemos: You should come back for the Asterisk Advanced class, Derek is an awesome teacher. |
| 08:43.06 | diemos | shapr: Convince my boss to pay for it, I just got a lecture today on how we need to be more productive -.- |
| 08:43.11 | Tozz_ | wdoekes2: subscribe returns 404 |
| 08:43.16 | kaldemar | _zoom_: define ringback. |
| 08:43.16 | Tozz_ | but the phone is registered according to * |
| 08:43.25 | shapr | diemos: Decreases the cost of a support subscription |
| 08:43.41 | wdoekes2 | do you have hints? or should it return 404? |
| 08:43.46 | Tozz_ | no hints |
| 08:43.46 | _zoom_ | time until gets NOANSWER from dial apps |
| 08:44.37 | wdoekes2 | spijtig.. syslog op de phone? |
| 08:44.48 | kaldemar | _zoom_: 136 years until you specify otherwise. |
| 08:44.53 | wdoekes2 | (niet dat ik daar ooit nuttige info uit gehaald heb) |
| 08:45.14 | diemos | shapr: Well, boss isn't exactly happy with digium right now either lol |
| 08:45.16 | kaldemar | s/until/unless/ |
| 08:45.21 | shapr | het spijt mig? |
| 08:45.49 | shapr | diemos: oh? how so? As far as I can tell the HSV peeps completely kick ass. |
| 08:46.43 | shapr | wdoekes2: I didn't get the second one, my Dutch is terrible. |
| 08:46.52 | wdoekes2 | "a shame" |
| 08:47.12 | wdoekes2 | "not that I ever got any useful info out of it" |
| 08:47.16 | diemos | Oh, it's not the people really. I try and stay out of it but we were quoted two SV systems for a certain price then when it came down to it, weren't getting that price. |
| 08:47.29 | shapr | oh |
| 08:47.38 | shapr | I don't know much about the SV guys. |
| 08:47.57 | diemos | Yeah, that's what we deal with mostly. |
| 08:48.19 | diemos | But I've met a few of the * people as well, they're all really cool. |
| 08:48.44 | shapr | wdoekes2: Ah, thanks |
| 08:49.00 | shapr | spijtig makes sense when I think about "het spijt mig" |
| 08:49.05 | wdoekes2 | and mig is swedish, it's mij |
| 08:49.07 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 08:49.31 | shapr | talar du svenska ocksa? :-/ |
| 08:49.34 | wdoekes2 | visst |
| 08:49.56 | shapr | varför? Jar är mycket förvånad och förvirrad! |
| 08:50.04 | wdoekes2 | haha, farsan är svensk |
| 08:50.08 | shapr | jaså |
| 08:50.23 | shapr | Dutch and Swedish seem pretty close to me anyway. |
| 08:50.44 | shapr | Not that I can find any Swedish speakers here in Alabama :-/ |
| 08:50.54 | wdoekes2 | both of germanic descent indeed |
| 08:51.11 | *** join/#asterisk rethus (~suther@p5087D0C5.dip.t-dialin.net) |
| 08:51.31 | rethus | in manager.cof the setting bindaddr: whatfor are this option? |
| 08:51.39 | rethus | asterisk listen only on this ip? |
| 08:52.15 | Tozz_ | yes |
| 08:52.17 | plundra | For the manager-interface, yes. |
| 08:52.23 | kaldemar | rethus: it is for the manager interface. 0.0.0.0 makes it listen to all addresses. |
| 08:52.23 | plundra | Not asterisk as a whole. |
| 08:53.34 | rethus | kaldemar: k, what if i'm on vpn? 10.8.0.1 should i enter this vvpn-address, or 0.0.0.0 or 127.0.0.1 ? |
| 08:54.04 | Siblor | Hello, does it possible to stream live conversation in asterisk channel to Java web application while Originate to application Record? |
| 08:55.16 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:55.20 | kaldemar | rethus: depends on how you want it to function. do you want to enable connections to the manager interface from outside your asterisk box? |
| 08:55.56 | rethus | thats a good question. Let me shortly think about it. |
| 08:56.35 | ChannelZ | <jeopardy theme> |
| 08:56.36 | rethus | i have a webapplication which should use asterclick... this lay on the same server as asterisk. |
| 08:56.45 | ChannelZ | </jeopardy theme> |
| 08:57.13 | rethus | so i think its not necessary |
| 08:57.20 | plundra | If it's on the same machine, then bind 127.0.0.1. |
| 08:57.27 | rethus | so i set it to 127.0.0.1 |
| 08:57.42 | kaldemar | rethus: then 127.0.0.1 should be enough, if you have no software outside the box that needs to connect the interface. |
| 08:57.46 | plundra | To avoid any risk of remote connections, if there are bugs (in asterisk or your passwords :-P) |
| 08:58.06 | plundra | And if you DO have remote software, consider wrapping those connections in something. |
| 08:58.09 | rethus | no, only the sip-clients connect |
| 08:58.10 | plundra | SSH for example. |
| 09:02.34 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 09:02.40 | *** part/#asterisk giany (~giany@shifu.x83.org) |
| 09:05.38 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:05.40 | schmidts | good morning |
| 09:05.49 | *** join/#asterisk a_m_y (~IceChat7@121.96.33.130) |
| 09:07.20 | Tozz_ | it seems that * tries to contact the phone on its internal IP |
| 09:07.24 | Tozz_ | even thought nat=yes |
| 09:07.42 | Tozz_ | any ideas what might cause that? |
| 09:08.02 | shapr | Tozz_: look at the SIP traffic with wireshark and check the Via header? |
| 09:08.13 | kaldemar | Tozz_: lack of externip/externaddr/externhost in your config. |
| 09:08.52 | shapr | kaldemar: Can you explain why that is? |
| 09:09.02 | a_m_y | hi is it normal to get this when issuing make install in libpri. http://pastebin.com/Cszp2GGR |
| 09:09.42 | shapr | a_m_y: looks normal to me |
| 09:09.42 | kaldemar | shapr: asterisk needs to be told what public address it should be using. there is no automated way of finding it out. |
| 09:09.58 | Tozz_ | kaldemar: my * box has an external IP |
| 09:10.08 | Tozz_ | kaldemar: the problems arised after I upgraded to 1.4.39.1 last night |
| 09:10.12 | Tozz_ | but only on 3 phones (behind 1 router) |
| 09:10.27 | shapr | suggests 1.8.2 |
| 09:10.53 | Tozz_ | yes well, last time I upgraded to 1.6 everything went fubar. We have quite alot of SIp accounts for all kinds of equipment from customers |
| 09:10.53 | shapr | Digium won't be offering commercial support for 1.4 or 1.6 in a few months, they'll be security fix only... basically end of life. |
| 09:11.01 | kaldemar | Tozz_: where are the phones? |
| 09:11.08 | Tozz_ | everywhere in the country ;) |
| 09:11.22 | Tozz_ | but behind nat, draytek 2700 router |
| 09:11.27 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 09:11.36 | kaldemar | Tozz_: as in relation to your asterisk box. |
| 09:11.51 | Tozz_ | i dont understand the question? |
| 09:12.12 | kaldemar | Tozz_: are they in the same LAN or somewhere else? |
| 09:12.20 | Tozz_ | elsewhere |
| 09:12.46 | Tozz_ | the phones are at customer location, behind draytek 2700 DSL router. Our * box has a public IP |
| 09:13.11 | kaldemar | and you have defined nat=yes, an external address and localnets under [general]? |
| 09:13.45 | Tozz_ | mm, no. But we never had |
| 09:13.52 | Tozz_ | in 1.4.18 this config worked like a charm ;) |
| 09:14.06 | schmidts | tozz_ maybe your draytek routers use sip alg? |
| 09:14.16 | Tozz_ | schmidts: I verified that, SIP ALG is disabled |
| 09:14.23 | Tozz_ | but, the 'Reg. Contact' still shows the external IP |
| 09:14.34 | Tozz_ | from what I've seen that usually means SIP ALG is enabled |
| 09:14.46 | Tozz_ | disabling NAT keepalive on the phones seem to have solved it for now |
| 09:14.52 | kaldemar | wait, asterisk used its internal address as the source address or the phone's internal address as the destination? |
| 09:15.11 | Tozz_ | I see ARP Requests for the phone's internal IP on my box |
| 09:15.17 | Tozz_ | so asterisks tries to talk to the internal IP |
| 09:16.21 | Tozz_ | but it seems to be solved with disabling NAT keepalive on the phones |
| 09:17.02 | schmidts | tozz_ this makes no sense, nat keep alive keeps the ports open. without this asterisk will not have a way back through the nat |
| 09:17.12 | schmidts | only if you have qualify=yes |
| 09:17.22 | Tozz_ | which we have |
| 09:17.40 | Tozz_ | perhaps it brakes if the NAT session times out |
| 09:17.43 | Tozz_ | we have to wait and see |
| 09:19.17 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 09:22.12 | *** join/#asterisk endigo-ceo (~bdfoster@99-153-227-32.lightspeed.gnwdin.sbcglobal.net) |
| 09:24.20 | endigo-ceo | got a problem, i upgraded to 1.8.2.2 on jan 23, and just realized my cdr stopped working. i just updated today to 1.8.2.3 and now all my cdrs are gone. is this a bug? is there a workaround? |
| 09:24.37 | Tozz_ | just regular cdr file or mysql? |
| 09:24.48 | endigo-ceo | mysql |
| 09:25.05 | Tozz_ | (i have no experience with 1.8) but with 1.4 u need the asterisk-addon package |
| 09:25.40 | kaldemar | the addons are included in the main package in 1.8. |
| 09:25.50 | endigo-ceo | ah, maybe. not sure if 1.8 |
| 09:25.55 | endigo-ceo | yea, that above lol |
| 09:26.21 | kaldemar | but they don't build automatically, you need to select them in the menu. |
| 09:26.47 | endigo-ceo | so, what do i need to select? |
| 09:27.13 | endigo-ceo | ive always been on 1.8 (with this system anyway) |
| 09:31.13 | endigo-ceo | someone else here is also having the same issue, but he compiles from svn |
| 09:31.15 | kaldemar | in the "Add-ons" section, cdr_mysql. |
| 09:31.34 | kaldemar | in the menu that you get with "make menuselect". |
| 09:31.45 | endigo-ceo | yea, ok ill take a look |
| 09:32.18 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 09:32.24 | diemos | I'm loving this song: http://dancesauce.com/wp-content/uploads/2011/01/Freefire-Dataloss-Darth-Vader-Remix.mp3 |
| 09:43.13 | endigo-ceo | kaldemar, thank you that solved the issue |
| 09:45.42 | *** join/#asterisk Siblor (~putthibon@202-176-88-55.static.asianet.co.th) |
| 09:48.04 | *** join/#asterisk manji (~manjiki@tribute.physics.upatras.gr) |
| 09:51.43 | *** join/#asterisk drmessano^ (~nonya@pdpc/supporter/active/drmessano) |
| 09:53.34 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 10:10.03 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 10:10.25 | shamelessn00b | Hi guys, I wanted to know if that issue on channel redirect on AMI call to asterisk 1.8 was fixed |
| 10:11.02 | shamelessn00b | I'm setting up a new asterisk server right now, about to download, need to decide whether I should download 1.6.x or 1.8.x |
| 10:14.56 | *** join/#asterisk manji (~manjiki@penicuik.physics.upatras.gr) |
| 10:16.10 | shamelessn00b | Ok, I did a quick check and I think the issue has been resolved |
| 10:18.15 | creativx | is there anything speshul I need to do to make 1.4 playback .sln files? |
| 10:18.49 | shamelessn00b | no |
| 10:18.53 | shamelessn00b | slin |
| 10:18.56 | shamelessn00b | you mean |
| 10:19.11 | shamelessn00b | .slin |
| 10:19.30 | creativx | no I mean .sln |
| 10:20.23 | creativx | oh. does not exist in any format |
| 10:20.36 | shamelessn00b | signed linear |
| 10:20.41 | shamelessn00b | is .slin |
| 10:20.48 | shamelessn00b | .raw |
| 10:20.51 | shamelessn00b | rather |
| 10:21.18 | creativx | playback() ignores the audio language setting |
| 10:21.25 | creativx | so it didnt look in sounds/no/... but in /sounds/ |
| 10:22.05 | shamelessn00b | did you set the language in the dialplan |
| 10:22.15 | creativx | no, I thought I had set it in sip.conf |
| 10:24.33 | creativx | but oh well, set(channel(language)=no) fixed it |
| 10:26.55 | *** join/#asterisk daxt (~daxt@112.135.65.111) |
| 10:27.08 | daxt | Hello does anybody who use VDP ? |
| 10:28.19 | diemos | I don't even know what VDP is. |
| 10:28.33 | shamelessn00b | visual dialplan? |
| 10:29.42 | shamelessn00b | I have my own tool to write dialplan, each context is a node and the lines that direct from 1 context to the other are edges |
| 10:30.05 | shamelessn00b | its totally awesome :) |
| 10:34.13 | daxt | diemos ya its visual dial plan |
| 10:34.36 | daxt | i am trying to connect it with my EC2 instance and unfortunately it does not connect :( |
| 10:44.41 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 10:54.08 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 10:54.57 | *** join/#asterisk benlangfeld (~ben@unaffiliated/benlangfeld) |
| 10:57.55 | *** part/#asterisk rethus (~suther@p5087D0C5.dip.t-dialin.net) |
| 11:04.09 | *** join/#asterisk yenkee (~yenkee@217.75.94.46) |
| 11:10.22 | yenkee | Hi I would like to register sip trunk to my provider. I have 100 extensions 100-199 and numbers from sip provider xxxxxxxxx00-99. Is it possible to made separate registrations for each extension using only one trunk. If not how to bind ext to real numbers. Provider is probably using kamailio |
| 11:10.49 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:19.29 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 11:36.57 | *** join/#asterisk geemee (~ocs@mailhost.exterity.com) |
| 11:39.17 | geemee | Hi All. We have an issue with an old asterisk box that occasionally ( i havent been able to find a pattern yet) when transferring a call to another extension the person recieving the transferred call hears loud white noise / distortion. I am looking through the logs but I am struggling to see what may be the problem. Has anyone experienced this before>? |
| 11:39.46 | geemee | I should add sorry that we are using IDSN lines with misdn as the initial inbound route for calls. |
| 11:42.57 | *** join/#asterisk shazaum (~shazaum@unaffiliated/shazaum) |
| 11:43.09 | shazaum | asterisk source make in openbsd (fail) |
| 11:43.10 | shazaum | :( |
| 11:46.39 | *** part/#asterisk shazaum (~shazaum@unaffiliated/shazaum) |
| 11:48.12 | *** join/#asterisk shazaum (~shazaum@unaffiliated/shazaum) |
| 11:53.33 | *** join/#asterisk BlackBishop (dexter@d3xt3r01.tk) |
| 11:53.56 | BlackBishop | hmmm .. I'm using chan_datacard, any way I can store in ${SMS} a multiline sms ? |
| 11:54.03 | BlackBishop | I seem to only get the first line ... |
| 11:54.42 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 12:03.48 | *** join/#asterisk joel_oliveira (~chatzilla@alpes.nortenet.pt) |
| 12:08.20 | *** join/#asterisk Weazel (~Weazel-@213.8.83.6) |
| 12:08.39 | Weazel | hey guys, is it possible to use Alert Info to change a Call waiting tone in a specific phone ? |
| 12:09.46 | Weazel | <PROTECTED> |
| 12:11.46 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 12:16.45 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:16.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:19.22 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 12:19.57 | shazaum | uia |
| 12:21.05 | shazaum | anonymouz666, nao vi mais vc no asterisk-br :) |
| 12:21.28 | anonymouz666 | há um bom tempo já |
| 12:23.18 | shazaum | eh, nao mudo nada :P |
| 12:24.30 | anonymouz666 | shazaum: como assim? |
| 12:24.57 | shazaum | anonymouz666, o canal continua a mesma coisa |
| 12:24.58 | shazaum | :) |
| 12:25.10 | anonymouz666 | ah sim |
| 12:25.28 | anonymouz666 | e você gostaria que mudasse? |
| 12:26.06 | yenkee | Hi I would like to register sip trunk to my provider. I have 100 extensions 100-199 and numbers from sip provider xxxxxxxxx00-99. Is it possible to made separate registrations for each extension using only one trunk. If not how to bind ext to real numbers. Provider is probably using kamailio |
| 12:26.32 | *** join/#asterisk tobi- (~tobi@85.220.135.170) |
| 12:31.06 | *** join/#asterisk BlackBishop (dexter@d3xt3r01.tk) |
| 12:31.24 | shazaum | anonymouz666, um dia quem sabe |
| 12:35.39 | BlackBishop | I have a 123 number defined in extensions.conf ... with some stuff in it .. but if I put Dial(SIP,123) in another part of the extensions.conf ( an incomming-datacard ) .. it doesn't work .. |
| 12:35.48 | BlackBishop | anyone can enlighten me on how this thing works ? |
| 12:37.56 | schmidts | BlackBishop take a look at Goto or try a Dial(LOCAL/123) thats what you want |
| 12:38.46 | BlackBishop | ow |
| 12:38.50 | BlackBishop | tries |
| 12:38.53 | schmidts | blackbishop dial(SIP/123) trys to dial peer 123 not extension 123, or you can use DIAL(SIP/123@localhost) but its not what you want at all ;) |
| 12:39.03 | BlackBishop | so I want LOCAL |
| 12:39.35 | *** join/#asterisk drcode (~user1@bzq-84-111-91-102.red.bezeqint.net) |
| 12:39.38 | drcode | hi all |
| 12:39.48 | schmidts | Local opens a new channel which acts like a new call to this extension and bridge your call and the local call together, a goto just moves your call to the extension you want |
| 12:40.05 | drcode | can asterisk have good support (opensource) for video confrence? |
| 12:40.20 | drcode | I mean SIP and Video |
| 12:40.20 | BlackBishop | schmidts: what if I don't have 123 in @default |
| 12:40.28 | BlackBishop | I have it in [robotel] |
| 12:41.00 | schmidts | blackbishop a goto would look like this: Goto(robotel,123,1) or Dial(LOCAL/123@robotel) |
| 12:41.16 | BlackBishop | notes everything down |
| 12:41.48 | drcode | any one using video and sip with asterisk? |
| 12:42.28 | leifmadsen | drcode: there is no video conferencing in Asterisk, but you can do end-to-end video |
| 12:42.43 | drcode | I see |
| 12:43.00 | drcode | no , I am looking for MUC for 10 pepole at same time |
| 12:43.10 | leifmadsen | then Asterisk will not satisfy that dependency |
| 12:43.20 | leifmadsen | try Skype |
| 12:43.31 | leifmadsen | or LifeSize |
| 12:43.37 | drcode | I need server |
| 12:43.49 | drcode | local server I mean |
| 12:43.50 | leifmadsen | you have a lot of needs that may not be satisfied :) |
| 12:43.55 | drcode | k |
| 12:44.09 | leifmadsen | you could try Logitech |
| 12:44.09 | drcode | I read some about videomixer |
| 12:44.35 | schmidts | drcode you can use asterisk to connect your sip video calls to the conference system but you will not be able to use only asterisk |
| 12:44.39 | leifmadsen | that would be a third-party application if it does exist |
| 12:44.41 | BlackBishop | schmidts: the drinks are on me ! |
| 12:44.58 | Weazel | hey guys, is it possible to use Alert Info to change a Call waiting tone in a specific phone ? i mean i can change the CW tone from the phone itself in general, but i want it to play only from a certain extension calling into it |
| 12:44.59 | *** join/#asterisk Akiraa (~Akira@92.81.213.75) |
| 12:45.25 | leifmadsen | Weazel: you'll have to look at the documentation for the phone to determine if you can do that |
| 12:45.43 | leifmadsen | you can certainly set a variable in Asterisk to anything you want :) |
| 12:47.31 | anonymouz666 | where's [tk]d-fender? |
| 12:48.30 | anonymouz666 | "paste your sip.conf and extensions.conf" :P |
| 12:48.36 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:51.04 | *** join/#asterisk devon_hillard (~Akira@unaffiliated/devon-hillard/x-7250961) |
| 12:57.10 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
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| 12:59.27 | shriven | Hello. Sorry if this information is somewhere, but I can't find it. Are there any upgrade instructions for going from 1.6 > 1.8? |
| 13:00.06 | wdoekes2 | there us usually a CHANGES file in the root of the source package that lists the differences |
| 13:00.25 | shriven | ahh ok I'll have a look in there |
| 13:01.18 | shriven | wdoekes2: UPGRADE.txt in the package. Thanks! |
| 13:17.23 | BlackBishop | anyone can help set up asterisk to be able to use smsq right ? :/ |
| 13:17.32 | BlackBishop | I don't seem to understand how it actually works |
| 13:17.41 | BlackBishop | I need to devine a context or something in asterisk ? |
| 13:17.46 | BlackBishop | then pass it as a variable to smsq ? |
| 13:22.35 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 13:23.30 | BlackBishop | if I do smsq -d number message I get [Feb 1 14:48:27] NOTICE[314]: chan_local.c:692 local_call: No such extension/context 17094009@default while calling Local channel :| |
| 13:24.40 | BlackBishop | all I'm trying to do is replicate the "datacard sms datacard0 number message" cli command.... |
| 13:24.49 | BlackBishop | is it possible to do that with smsq ? |
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| 13:43.02 | leifmadsen | BlackBishop: that error says there is nothing in [default] with the extension 17094009 available |
| 13:43.24 | leifmadsen | your [default] context doesn't have the right extension number or pattern match in it for the Local channel to call |
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| 13:49.55 | BlackBishop | leifmadsen: yeah, but why is it trying to call that context ? and how should that context look like for my service center number .. ? |
| 13:50.51 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 13:52.15 | leifmadsen | BlackBishop: that's how you've configured smsq I guess -- I have never used that application |
| 13:52.28 | leifmadsen | so I don't have enough information to provide you any additional help |
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| 14:19.24 | BlackBishop | leifmadsen: well, a more simple solution is asterisk -r -x 'datacard sms datacard0 number message' |
| 14:19.29 | BlackBishop | I guess I'll just use that |
| 14:20.09 | yenkee | Hi I would like to register sip trunk to my provider. I have 100 extensions 100-199 and numbers from sip provider xxxxxxxxx00-99. Is it possible to made separate registrations for each extension using only one trunk. If not how to bind ext to real numbers. Provider is probably using kamailio |
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| 14:28.55 | schmidts | yenkee it would be much easier to only register 1 trunk and just let your provider sent you the dialed extension |
| 14:33.18 | yenkee | schmidts: but I want each ext have their exact real number 111 -> xxxxxxxx11 |
| 14:35.02 | yenkee | schmidts: and how the providers server would know where to send incoming calls. If i asume it right and they are using Kamailio, i have to register each number to receive these calls. |
| 14:36.17 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 14:36.57 | schmidts | yenkee if your provider knows which numbers is owned by which account (normally it should be) it will be no problem |
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| 14:40.43 | leifmadsen | yenkee: ya I have a customer with 1000 DIDs and only 1 registration |
| 14:41.09 | leifmadsen | the incoming call is a request to a particular number and then the dialplan does whatever it does for each of the numbers (based on the route lookup table) |
| 14:41.23 | yenkee | leifmadsen: and how your provider knows where to send incoming calls. |
| 14:41.32 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 14:41.42 | leifmadsen | yenkee: they just setup like schmidts said -- all 1000 numbers are associated with my account |
| 14:41.47 | leifmadsen | which is common |
| 14:41.50 | tzanger | yeah right... a customer with a thousand DIDs... just like you have a fiancee... your lies are getting more and more unbelievable. |
| 14:41.55 | leifmadsen | why would you register every single DID number from the same location? |
| 14:42.11 | leifmadsen | tzanger: you're right... my love in my heart is for you |
| 14:42.18 | leifmadsen | tzanger: I can't deny it any longer! |
| 14:42.33 | tzanger | there's no need to deny it leifmadsen |
| 14:42.38 | tzanger | society is accepting of these things now |
| 14:42.53 | leifmadsen | nice |
| 14:42.58 | tzanger | how goes/ |
| 14:43.02 | *** join/#asterisk daxt (~daxt@112.135.93.102) |
| 14:43.16 | leifmadsen | oh not bad! I'm about to burn a DVD and reinstall Ubuntu on my server downstairs |
| 14:43.30 | tzanger | nice, I was stupid and decided to play with the 11.04 alphas |
| 14:43.36 | tzanger | my dependency tree is in a sad sad state |
| 14:43.44 | leifmadsen | :( |
| 14:43.46 | tzanger | I'm surprised when the computer actually boots :-) |
| 14:43.50 | leifmadsen | ya I'm just going to install 10.10 I think |
| 14:43.55 | tzanger | yeah that's smart |
| 14:44.00 | tzanger | 10.10 didn't cause me any pain |
| 14:44.10 | tzanger | and my workstation is still 9.10 or 9.04 I think |
| 14:44.14 | leifmadsen | I have no interest in trying 11.04 at all while they try a new GUI interface |
| 14:44.14 | Janos | hi there, have 2 asterisk interconnected using sip (A and B), but right now only have calls in one direction, asterisk B can call asterisk A but not the other way around, sip show peers in asterisk A shows the peer as unreachable, how can i tell what's wrong ? |
| 14:44.28 | tzanger | well I'm running KDE so it's not so abrupt :-) |
| 14:44.32 | leifmadsen | nice :) |
| 14:44.35 | leifmadsen | I use Gnome |
| 14:44.42 | leifmadsen | I knew we'd never work |
| 14:45.15 | tzanger | it's how things work sometimes... we're just too different for each other |
| 14:45.21 | tzanger | the differences were exciting at first |
| 14:45.24 | tzanger | but now they get in the way |
| 14:46.48 | yenkee | leifmadsen: Now I have 1 number from my operator (username:password), this works in 1 trunk. But now i want 100 numbers. If it should be your way they have to change configuration of their server. So I have to deal with them to send my those numbers. |
| 14:48.37 | *** join/#asterisk NirS (~androirc@2.54.13.27) |
| 14:48.57 | yenkee | leifmadsen: For example they are offering voicemail if I am not registered. This functionality wouldn't be possible because they wouldn't know if the number is connected. |
| 14:49.02 | *** join/#asterisk daxt (~daxt@112.135.72.142) |
| 14:50.16 | schmidts | yenkee do you really want to let your provider sent voicemails to your customers directly? |
| 14:50.55 | NirS | <PROTECTED> |
| 14:51.09 | *** join/#asterisk pepe_lucho (~pepe_luch@unaffiliated/pepe-lucho) |
| 14:52.00 | Janos | so any idea how to find out why an asterisk is not getting registered on another ? |
| 14:52.39 | Dovid | leifmadsen: I know your not at work yet. put in a bug report for the issue that we spoke about yesterday |
| 14:52.50 | NirS | start with a ngrep trace |
| 14:53.07 | tzanger | Janos: take a look at the sip debug on each box. My initial guess is that one is not using a register where the other is thinking it should. |
| 14:53.12 | NirS | make sure the packets are getting there |
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| 14:56.25 | leifmadsen | Dovid: yes and I will see it when I do my daily bug triage :) |
| 14:56.31 | leifmadsen | you don't need to point them out to me :) |
| 14:56.36 | Janos | tzanger: i have the register on both sides and it was working until i rebooted yesterday after applying an update unrelated to asterisk |
| 14:57.51 | tzanger | Janos: take a look at the sip debug logs |
| 14:58.53 | tzanger | I have seen situations where an office phone which connects to both A and B "wins the race" and manages to register using port 5060, leaving the asterisk box itself to use a different port and cause the far-end server to reject the registration |
| 14:59.00 | tzanger | that was a real bitch to figure out |
| 15:00.07 | Katty | i can haz caffeines now plz?! |
| 15:00.56 | tzanger | Katty: I'll be going to brunch in 30-45, we could share all kinds of caffeine |
| 15:01.15 | Katty | you haven't had mono in the last 18 months have you? :< |
| 15:01.39 | tzanger | Katty: I've never had mono ever |
| 15:01.40 | tzanger | EVAR |
| 15:01.51 | Katty | ^_________________^ |
| 15:01.54 | Katty | lesgo! |
| 15:02.16 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 15:02.26 | tzanger | I think the only truly infectious thing I ever had aside from colds/flus was chicken pox |
| 15:02.42 | Katty | you've never had strep? |
| 15:02.58 | tzanger | not that i'm aware of. I've still got both my tonsils and my appendix too |
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| 15:03.37 | Katty | ya you'd know if you had strep, you lucky ducky |
| 15:03.39 | tzanger | and 3 of my 4 wisdom teeth... doc pulled one when I was 16 and I didn't go to another dentist until I was 30 |
| 15:04.23 | tzanger | dentist looked at me, said my teeth had a little more plaque buildup than most people, but no cavities and nothing out of the ordinary |
| 15:04.39 | shriven | tzanger: lucky |
| 15:04.42 | Katty | do you brush twice or once a day? |
| 15:04.52 | Leddy | wow, that was pretty painless, just changed my apple id |
| 15:04.56 | Leddy | "it just works" |
| 15:05.05 | tzanger | which is kind of funny... as a kid I had cavities all over the place. Just this past year I had my first adult filling, and it was just a surface thing, he did it without anaesthetic or anything |
| 15:05.16 | Katty | aww :< |
| 15:05.21 | Katty | i guess that's good tho. idk. |
| 15:05.30 | Katty | i'm very lucky and don't have toof problems. |
| 15:05.34 | tzanger | Katty: usually once but if I feel grungy I'll brush them again at the end of the day |
| 15:05.43 | Katty | electric? |
| 15:05.45 | Katty | or regular |
| 15:05.51 | tzanger | Katty: well as I said it was a surface thing, it wasn't a deep cavity so no need to drill, I think he sanded it off to be honest |
| 15:06.00 | Katty | nice |
| 15:06.20 | tzanger | Katty: regular. I cannot stand electric anything for hygeine... regular toothbrush, grandpa-style safety razor... I'm all-manual. :-) |
| 15:06.40 | tzanger | something bugs me about having to rely on electricity to shave or brush my teeth |
| 15:06.42 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 15:06.52 | Katty | mkay |
| 15:07.07 | Katty | i tried an electric toothbrush and it did so much better i wouldn't go back to regular |
| 15:07.19 | Katty | but i understand that you don't like battery operated vibrating things ;) |
| 15:08.17 | leifmadsen | ya, they work way way better than manual |
| 15:08.38 | leifmadsen | I have one with a small round head because the larger heads don't get my wisdom teeth |
| 15:09.00 | Katty | what kind do you have? |
| 15:09.59 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 15:11.09 | Janos | tzanger: i'm debugging with "sip set debug ip <remote ip>" and all i'm getting is this on asterisk A, not getting enything on asterisk B |
| 15:11.09 | tzanger | I duno... my wife uses an electric one and she has way more problems with her teeth than I do. and Volkerding apparently got some nasty fungal infection in his lungs from breathing the "cloud" that an electric toolbrush generates |
| 15:11.11 | Janos | http://pastebin.com/4XQnpSLc |
| 15:11.52 | Janos | that is getting retransmitted around 4 times |
| 15:12.02 | tzanger | Janos: looks to me like the firewall isn't doing what you expect it to |
| 15:12.39 | Janos | tzanger: meaning i have a communication problem ? |
| 15:12.50 | tzanger | Janos: yeah, could be something as simple as what I described |
| 15:13.06 | tzanger | do you have any SIP phones behind the firewall that connect directly to the far-end asterisk box? |
| 15:13.26 | Janos | tzanger: asterisk IS the firewall on both sides |
| 15:13.42 | tzanger | Janos: doesn't matter |
| 15:13.56 | tzanger | Janos: do you have any SIP phones being A that connect directly to B? |
| 15:14.03 | tzanger | behind |
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| 15:14.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:16.10 | Janos | tzanger: no, let me check comm and see if everything is fine, why is there no "can't connect error" or something, i would have never thought about comm with that sip debug |
| 15:16.38 | tzanger | "retransmitting" generally means it was not getting the (a) response so it's trying again |
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| 15:22.58 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 15:24.51 | Janos | tzanger: i think i found the problem, but don't know how to fix it, like i said both asterisk servers are the firewall/router on both ends, and they both have an internet link and a private link, according to tcpdump asterisk A is trying to register with asterisk B but using it's public address as source, this is not good |
| 15:27.50 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:28.58 | drmessano-lt | bindaddr |
| 15:30.25 | Janos | i had bindaddr in sip.conf on each side bound to the internal ip address of each server, i just changed it to 0.0.0.0 but still getting the same issue |
| 15:33.18 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 15:35.46 | Janos | shutdown my internet interface and restarted asterisk, everything is working know, this is deff not good |
| 15:36.57 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 15:38.07 | ariel_ | ping tzafrir you have a few more minutes this time just for some questions? |
| 15:38.57 | tzafrir | ariel_, yes |
| 15:39.34 | ariel_ | how do I auto start on the first system drbd, dahdi and twinstar after a reboot? |
| 15:41.25 | *** join/#asterisk daxt (~daxt@112.135.95.88) |
| 15:47.37 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 15:48.41 | teloniusz | hi, guys. I've just accidentally removed my startup script for blackfin asterisk distribution (/etc/init.d/asterisk). Could anyone guide me to the repository where I can get it? |
| 15:48.50 | Katty | needs blood pumping, adreneline rushing music. suggestions? |
| 15:50.03 | *** join/#asterisk tyman (~tyler@173-12-219-178-Fresno.hfc.comcastbusiness.net) |
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| 15:50.58 | carrar | Do some Air Traffic controlling Katty |
| 15:51.10 | carrar | or sky diving |
| 15:51.32 | carrar | or drink coffee! |
| 15:51.36 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 15:52.14 | drmessano-lt | Katty: Jane's Addiction |
| 15:52.27 | Katty | mkay |
| 15:52.35 | Katty | i could totally do air traffic control |
| 15:52.36 | drmessano-lt | I recommend the entire "Nothing's Shocking" disc |
| 15:53.05 | drmessano-lt | or perhaps Vanilla Ice's 1990 hit CD, "To the Extreme" |
| 15:53.50 | drmessano-lt | Particularly the whimsical musings of the tracks "Ice, Ice, Baby" and the 3 seconds of silence at the beginning of the CD |
| 15:53.58 | drmessano-lt | The rest, sadly, was forgettable |
| 15:55.22 | WIMPy | suggests "Living on vanilla ice" by DJ Brokenwindow then. Also the only good track on his CD. |
| 15:55.39 | teloniusz | Katty: http://flightradar24.com/ ;) |
| 15:57.33 | drmessano-lt | WIMPy, I once consumed an entire container of "Vanilla Ice", attempting to go "To the Extreme". I spent 3 days in the bathroom squinting like a Ninja Turtle. |
| 15:58.49 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 16:02.01 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 16:03.32 | BlackBishop | I'm trying to do something like: if(something || something || something){ set(); dial(); hangup() } ; if(something || something || something){ set(); dial(); hangup() } ; if(something || something || something){ set(); dial(); hangup() } ; |
| 16:03.42 | BlackBishop | what would be the best way to do something like this ? |
| 16:09.56 | russellb | BlackBishop: if it were me, AGI. |
| 16:10.08 | russellb | or if not that, AEL perhaps. |
| 16:10.45 | leifmadsen | doesn't like AEL |
| 16:11.44 | Janos | well problem is solved, but no idea why it happened and prolly will happen again |
| 16:12.09 | *** join/#asterisk barcef (~fbc@200.77.29.71) |
| 16:12.13 | Katty | russellb: ohai |
| 16:12.17 | Katty | hugs russellb |
| 16:12.27 | russellb | hi! |
| 16:13.01 | barcef | Every morning I need to resave my extension list so that my dahdi port begin to answer. Is there a way to troubleshoot this? |
| 16:13.09 | Janos | one last question about this, if i set bindaddr in sip.conf will this mean only that asterisk will listen on that ip only, or will asterisk also use that ip as source for sip packets ? |
| 16:13.29 | Janos | on a multi-interface system of course |
| 16:13.34 | Katty | russellb: how're you dear |
| 16:13.59 | BlackBishop | reads AGI and AEL |
| 16:14.56 | leifmadsen | BlackBishop: http://pastebin.com/taQh6A3C |
| 16:14.59 | leifmadsen | that's how I'd do it |
| 16:15.09 | leifmadsen | based on your single line pseudo code... |
| 16:15.27 | leifmadsen | russellb: that's how I'd do it ^^^ |
| 16:16.15 | WIMPy | Janos: Your routing should take care of that. |
| 16:16.38 | Janos | WIMPy: kk, makes sense |
| 16:16.45 | leifmadsen | BlackBishop: it depends exactly what you're trying to do as well, because it's possible the Set() could just contain an IF() function in it to set a variable |
| 16:16.48 | Janos | WIMPy: just wanted to confirm |
| 16:18.13 | BlackBishop | ael looks more like how I'm used to program stuff |
| 16:18.25 | leifmadsen | yep, which just converts back into dialplan |
| 16:18.55 | Janos | the weird thing here is ping packets and every other traffic did get the right source, going to have to look this up, thanks everyone |
| 16:19.46 | WIMPy | Janos: It is quite likely that the connection tracking shot in your foot. |
| 16:20.39 | WIMPy | You might want to make sure your private address space can never take the default route, even if not available (yet). |
| 16:21.56 | WIMPy | A set of reject roules for all private IPs should be standard on anything that routes. |
| 16:24.39 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:24.44 | Janos | WIMPy: i don't see the connection, the problem was asterisk actually using my public address to communicate with my internal networks, even if bind address was set to my private local ip and the route to reach the remote private network explicitly contained my private local ip as source |
| 16:24.47 | BlackBishop | leifmadsen: yeah, looks ok .. I'll use that .. |
| 16:26.33 | *** join/#asterisk nsgn (~nsgn@cpe-66-68-115-61.austin.res.rr.com) |
| 16:26.56 | WIMPy | Janos: Real local or VPN? |
| 16:27.02 | nsgn | hello. i've got a customer reporting that they loose calls periodically, almost always after pressing a key/making a DTMF tone. what are possible reasons for this? i'm having a hard time seeing why this is happening |
| 16:27.17 | WIMPy | I suspect the private address wasn't available when Asterisk started. |
| 16:27.25 | nsgn | the system is just a polycom phone going into an x86 asterisk box that is trunked to VoIP.MS via SIP |
| 16:28.03 | WIMPy | That then gives you an contrack entry and continues to mess it up even when it should work. |
| 16:28.46 | Janos | WIMPy: there is a vpn between my local private and my remote private net, but the bindaddr and the "src" in my route are both set to my real local private ip set on my real ethernet interface |
| 16:28.51 | WIMPy | conntrack can be really evil when it comes to UDP. |
| 16:29.27 | WIMPy | So the VPN might not have been up, yet. |
| 16:29.51 | WIMPy | meeting... |
| 16:31.15 | Dovid | nsgn: Could be your carrier is hanging up the call. what form of DTMF are you using ? rfc2833 ? info ? |
| 16:31.28 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 16:31.35 | nsgn | Dovid, i'm set to auto on the provider's side, same as i am at other locations with no problem |
| 16:32.15 | Dovid | nsgn: I have seen carriers (aka Voxbone) dump a call when they get DTMF via info |
| 16:32.47 | Dovid | yuo need to debug and see who hangs up |
| 16:33.33 | nsgn | whats funny is that i cant re-create it when i go onsite |
| 16:33.41 | nsgn | and yet the customer just emails me right now and says "Just sometimes, (if I called 10 times, it would happen 5 or 6 times)" |
| 16:33.46 | nsgn | oook |
| 16:34.02 | nsgn | why on earth would it do this to her only on 50 to 60% of calls |
| 16:34.13 | nsgn | seems if i had DTMF configured wrong i'd get dumped every time |
| 16:34.26 | Janos | WIMPy: it's possible but i'm sure i restarted asterisk many times after the server booted and the vpn was up by then, but even if the vpn is up, i explicitly specified the local private src address in the route, i don't see why asterisk would choose to use another ip, other traffic was working fine, but yeah going to check the routing thoroughly cuz it does not make sense for a program to... |
| 16:34.28 | Janos | ...set the source address of a packet, like you said that's the routing work |
| 16:34.42 | nsgn | i'd have traced it but when i went onsite last week i made more than 10 calls with no problems, thus having nothing to trace |
| 16:34.51 | nsgn | now she says its "almost unusable" |
| 16:38.04 | coppice | DTMF over SIP INFO messages is not standardised, so dropping calls when such a message is seen is not an unreasonable move |
| 16:38.30 | Katty | hi coppice |
| 16:38.37 | coppice | hi |
| 16:39.25 | citywok | nsgn sounds like you need to crank up logging and have them tell you what time it happened, then look at the logs. |
| 16:45.22 | *** join/#asterisk Godfather_ (~godfather@42.Red-88-9-178.dynamicIP.rima-tde.net) |
| 16:46.21 | barcef | Why do I need to save my extension list for my dadhi port to answer? |
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| 16:47.58 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 17:10.13 | neurosys | has anyone here played with the SPA3102 and overhead paging equipment? |
| 17:18.47 | hardwire | meh |
| 17:26.19 | neurosys | got it ;) |
| 17:26.20 | *** part/#asterisk neurosys (~neurosys@69.199.130.146) |
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| 18:08.59 | Tim_Toady | !squirrelcam |
| 18:09.26 | leifmadsen | ~squirrelcam |
| 18:09.32 | leifmadsen | ~katty |
| 18:09.32 | infobot | katty is, like, the only girl in the channel, so be nice to her |
| 18:09.42 | Tim_Toady | fail :D |
| 18:09.44 | leifmadsen | indeed |
| 18:09.57 | manji | and inaccurate |
| 18:10.41 | tzanger | nothing wrong with a little inaccuracy |
| 18:11.29 | manji | haha |
| 18:13.54 | Katty | leifmadsen: i like the backsplash! |
| 18:14.02 | Katty | infobot: crittercam |
| 18:14.02 | infobot | i guess crittercam is Katty's live broadcast of The Nut House at http://www.ustream.tv/channel-popup/squirrel-critter-cam |
| 18:14.57 | tzanger | mmmmmmm |
| 18:15.03 | tzanger | <-- AAlllllll coffeed up now |
| 18:15.07 | tzanger | but still feeling like a warmed over bag of shit |
| 18:15.11 | Tim_Toady | i failed miserably lol |
| 18:16.46 | *** join/#asterisk _zoom_ (~Administr@196.29.186.102) |
| 18:20.12 | manji | does anyone remember this song ? |
| 18:20.13 | manji | http://www.youtube.com/watch?v=4TtLI6HeDRw |
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| 18:23.10 | tzanger | damn, I need to find someone who speaks latin fluently |
| 18:23.33 | Freeaqingme | latin is a dead language |
| 18:23.45 | Freeaqingme | you may be able to find someone who's able to read/write it "fluently" though ;) |
| 18:24.00 | Tim_Toady | google translate supports latin |
| 18:24.07 | Freeaqingme | in beta, right? |
| 18:24.13 | Tim_Toady | dunno |
| 18:24.23 | Freeaqingme | but some terms simply cannot be translated because when latin wasn't dead yet, it didn't exist |
| 18:24.37 | manji | tzanger, out of curiosity |
| 18:24.39 | manji | why ?? |
| 18:24.42 | tzanger | Tim_Toady: maybe... I should check |
| 18:24.52 | manji | unless julius caesar is back |
| 18:24.58 | manji | and is in need of a PBX |
| 18:25.07 | tzanger | manji: in another channel someone was using the latin phrases that have made their way into English completely incorrectly |
| 18:25.26 | tzanger | and I wrote back "LATIN <expletive>, DO YOU SPEAK IT?" |
| 18:25.39 | tzanger | it was funnier at the time, but I need to get that translated properly so it can go into the bot |
| 18:25.43 | manji | lol |
| 18:26.43 | jaytee | I like "Science......it works, bitches!" |
| 18:32.41 | *** part/#asterisk _zoom_ (~Administr@196.29.186.102) |
| 18:39.48 | *** join/#asterisk moy_ (~moy@12.228.46.43) |
| 18:48.52 | drmessano-lt | YOU KNOW WHAT'S REALLY COOL? ICE |
| 18:51.36 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:51.36 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:52.18 | p3nguin | I must be one really cool guy, 'cause we have an enormous amount of ice here right now. |
| 18:52.22 | *** join/#asterisk jkroon (~jkroon@dsl-241-228-210.telkomadsl.co.za) |
| 18:53.33 | ClintGoudie-Nice | I've got a whole cup full |
| 18:53.51 | p3nguin | sorbet? |
| 18:54.28 | ClintGoudie-Nice | Diet coke with lime |
| 18:54.32 | ClintGoudie-Nice | over ice |
| 18:54.45 | p3nguin | Ugh. Diet Coke is evil. |
| 18:54.59 | ClintGoudie-Nice | I thought so too until I tried the "with lime" |
| 18:55.02 | ClintGoudie-Nice | it's quite tolerable |
| 18:55.10 | afink | even better with rum |
| 18:57.31 | manji | p3nguin, where to |
| 18:58.11 | p3nguin | manji: Hmm? |
| 18:58.27 | manji | [20:52] <p3nguin> I must be one really cool guy, 'cause we have an enormous amount of ice here right now. |
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| 19:01.04 | p3nguin | I know what I said, but I didn't understand your statement. |
| 19:02.29 | p3nguin | If you were trying to ask where the ice is... I'm in the midwestern USA. |
| 19:03.28 | jaytee | me too! I'm in Indiantucky |
| 19:04.17 | p3nguin | We're currently several hours into freezing rain. St. Louis has snow going on, but that hasn't moved in on us just yet. |
| 19:04.26 | jaytee | I can actually see the glaciers on the horizon, movin faster than a Windows OCS 2007 server :-) |
| 19:05.06 | jaytee | my car looked like a glazed donut this morning. took over an hour to get enough ice off it to drive. |
| 19:05.17 | p3nguin | *nod* |
| 19:05.39 | p3nguin | I don't expect to get out much today. |
| 19:05.41 | jaytee | we need solar mirrors in orbit to add more sunshine during the winter months. |
| 19:06.16 | p3nguin | STL has a blizzard warning! 10-24 inches of snowfall possible. Wow! |
| 19:06.40 | jaytee | wow is right, STL usually doesn't get much snow let alone 10-24 inches of it. |
| 19:07.10 | p3nguin | I wonder if they are overestimating this storm. I haven't seen a single flake of snow in a week. |
| 19:07.58 | manji | p3nguin, yes I meant where the ice was |
| 19:08.26 | p3nguin | It does look like they already have a couple inches on the ground from the looks of this remote newscast. |
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| 19:19.42 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 19:20.50 | MrTelephone | Fax for Asterisk shows a lot of examples for storing and sending faxes from the asterisk box. I'm guessing it also supports faxing from ATA clients? I am looking for a T38 gateway solution and don't mind paying digium if that is what Fax for Asterisk does. |
| 19:22.27 | tzanger | chan_fax is probably what you're after then |
| 19:22.44 | tzanger | they have a free (1 fax channel only) version for evaluation |
| 19:22.47 | Qwell | MrTelephone: ATAs are just SIP devices |
| 19:22.52 | MrTelephone | It's either that or reinvite fax clients to the cisco as5300 |
| 19:22.57 | Qwell | there's no reason it would be any different |
| 19:22.57 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 19:23.49 | manji | MrTelephone, asterisk+hylafax+t38modem might be what you are looking for |
| 19:24.01 | MrTelephone | Because there would be no reason to do t38 from the local machine that the PRI is connected to? |
| 19:24.19 | ruben23 | hi guys, i have an asterisk server behind NAt- i portfroward , mgcp, sip, rtp, iax2 ports, there are two users - local callers and remote callers- local callers can hear all the cleint but remote users they can hear anything from teh cleint they call any suggestions..? |
| 19:25.03 | MrTelephone | How about Arris 602G - T.38 - Asterisk - T.38 - Cisco AS5300? |
| 19:26.55 | ruben23 | i mean remote callers they cant hear anything from the client |
| 19:27.21 | MrTelephone | Qwell, if you have 2 sip endpoints sharing the same IP:PORT and the endpoint supports call screening *67. Asterisk picks the first registered peer in the list that matches the IP and PORT. Do you think it would be hard to design it to test the digest on multiple peers with the same IP:PORT? |
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| 20:06.11 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 20:06.31 | AliRezaTaleghani | hi |
| 20:07.20 | AliRezaTaleghani | i have a problem with Queue... |
| 20:07.20 | beek | hello |
| 20:08.28 | AliRezaTaleghani | I need to do some changes, on the Callers, whom are Queued... |
| 20:08.49 | AliRezaTaleghani | i don't know, if there is any way or not |
| 20:09.04 | AliRezaTaleghani | but, for an example.... |
| 20:09.17 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 20:10.07 | AliRezaTaleghani | i need to Pick a caller from Position 8 on Queue-A and place her on Position 4 on Queue-B |
| 20:10.41 | AliRezaTaleghani | :-./ is it possible? |
| 20:12.13 | Katty | heelllllllllooooooooooooooo nurse |
| 20:13.17 | tzanger | Katty: isn't that my line? |
| 20:13.26 | Katty | no |
| 20:13.32 | beek | AliRezaTaleghani: I don't think so. |
| 20:13.34 | tzanger | oh |
| 20:13.35 | Katty | i claimed it. |
| 20:13.35 | tzanger | that reminds me |
| 20:13.46 | Katty | with this gun from the national rifle association |
| 20:13.49 | Katty | and a flag. |
| 20:14.03 | tzanger | if any of you are looking for a **GOOD** SIP client for iphone, grab the acrobits app "GroundWire" it's $10 and antoher $10 if you want the g729a codec but it is AWESOME |
| 20:14.15 | AliRezaTaleghani | beek: :( but i need to do... |
| 20:14.21 | tzanger | I have tried every SIP client for the iphone, both in the app store and in Cydia, and this is worth it |
| 20:14.23 | Katty | why SIP when you can txt ;) |
| 20:14.27 | tzanger | the other ones are shite |
| 20:14.34 | jdoe | tzanger: bria works well for me. |
| 20:14.35 | AliRezaTaleghani | are u interested on AMI commands? |
| 20:14.39 | simplydrew | tzanger: I'll have to take a look. I've been looking at one for a while |
| 20:14.43 | Katty | do you have any idea how many txts i send a month |
| 20:15.43 | tzanger | jdoe: it's not a matter of "works well" it's a matter of OMGWTFBBQ THIS APP IS AWESOME... call recordings, multiple accounts, gphone integration, push notification (that's a big one for me) different codec priorities for cell/wifi, remote control client... AWRSOME |
| 20:15.51 | tzanger | Katty: nope, how many? |
| 20:16.07 | Katty | tzanger: guess. |
| 20:16.15 | Katty | tzanger: the average teenager sends 3000 a month. just FYI |
| 20:16.15 | tzanger | I texted my 15yo the other day and called him a n00b. he was hurt his lexicon was actually mine about the time he was born |
| 20:16.23 | tzanger | Katty: you're not a teenage |
| 20:16.24 | tzanger | r |
| 20:16.27 | Katty | nope. |
| 20:16.37 | tzanger | hmm |
| 20:16.46 | tzanger | I will hazard a guess of 3500. |
| 20:16.58 | simplydrew | tzanger: so if someone calls my DID will it ring my desk phone and the app on my iPhone as well? |
| 20:16.59 | Katty | i will go check, to get an acurate number. |
| 20:17.02 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 20:17.09 | simplydrew | looks good, but doesn't go into depth about incoming calls |
| 20:17.18 | tzanger | simplydrew: depends on how your asterisk server is set up, but that's what it does on mine, yes |
| 20:17.27 | simplydrew | tzanger: ah alright, just wondered |
| 20:17.32 | tzanger | simplydrew: it's a SIP client, what is there to know about incoming calls? :-) |
| 20:17.58 | tzanger | all my problems with SIP clients have to do with codecs, multiple accounts, audio quality, power consumption, etc. |
| 20:18.10 | Katty | 2058 |
| 20:18.20 | simplydrew | tzanger: well, incoming calls in the sense of it's already ringing my desk phone, so wasn't sure how it was going to do both |
| 20:18.25 | Katty | 32 minutes of phone time. |
| 20:18.33 | Katty | and 47mb of data. |
| 20:18.57 | simplydrew | I still manage to use a lot of minutes on my iPhone |
| 20:19.02 | simplydrew | and data |
| 20:19.05 | Katty | still 12 days to go, too |
| 20:19.10 | simplydrew | but data over everything else |
| 20:19.10 | tzanger | simplydrew: Dial(SIP/deskphone&SIP/iphone&SIP/12345678@provider) ... dialing is trivial |
| 20:19.25 | tzanger | Katty: hmm, I overestimated you. |
| 20:19.44 | Katty | 32 minutes of talk time a month is...awful |
| 20:19.44 | tzanger | my typical use is about 120 minutes of voice, 450MB of data and 100-200 SMS |
| 20:20.16 | Katty | i certainly have you beat on the sms. |
| 20:20.21 | tzanger | uh, yeah |
| 20:20.36 | Katty | :P |
| 20:21.25 | IsUp | 100-200 SMS monthly? |
| 20:21.57 | Katty | tzanger sends about 7 txts a day. |
| 20:22.06 | Katty | i send about 70 a day |
| 20:22.24 | IsUp | -> 3000/monthly here |
| 20:22.25 | tzanger | heh |
| 20:22.35 | tzanger | my fingers would be wore to the nubs |
| 20:22.53 | Katty | nah |
| 20:22.58 | Katty | mine are fine. |
| 20:23.19 | IsUp | i got blackberry 9700 last month, Vodafone offers 5000 SMS to Vodafone subscribers, and 500 SMS to other providers. for $7 |
| 20:23.58 | tzanger | I'm in Canada. we have costly cell plans. |
| 20:24.06 | *** join/#asterisk reber (~reber@ns1.sylogix.net) |
| 20:24.12 | Katty | mine's about 70 a month |
| 20:24.14 | Katty | unlimited data tho |
| 20:24.21 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 20:24.21 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:24.42 | tzanger | 6G of data is $30. 250min voice is $25. voicemail+unlimited sms+caller id is $10 I think, plus the system access fee bullshit and so on |
| 20:24.55 | IsUp | i have problems with data, Vodafone is too slow.. my plan includes 4GB data but i am nearly using 200MB monthly |
| 20:25.14 | tzanger | yeah I use about half a gig at peak unless I am relying on it for connectivity |
| 20:25.22 | *** part/#asterisk fixxar (~tmurch@pool-96-242-157-115.nwrknj.fios.verizon.net) |
| 20:25.30 | tzanger | I am happy with Rogers HSPA network though, it's always been pretty responsive |
| 20:25.52 | IsUp | also Vodafone doesnt include "apn" for this plan. if you are going to connect your BB to laptop or pc, they are charging extra fee |
| 20:26.39 | IsUp | we have 3G in Turkey :p its new actually |
| 20:26.45 | jdoe | tzanger: ... push notification of... what? |
| 20:27.06 | tzanger | jdoe: if the app is not running it can receive a push notification to start up to accept the call |
| 20:27.19 | tzanger | oh it also has encryption and TCP support if you so desire |
| 20:27.23 | jdoe | yeah, I really want to give some random iphone dev my sip creds. |
| 20:27.24 | jdoe | pass :P |
| 20:27.36 | tzanger | and you can play with the settings if you're using streaming voice (not UDP) similar to IMAP IDLE for faster wakeup |
| 20:28.07 | tzanger | jdoe: that's why you give them the creds to the phone and you code up some standard protective logic in the dialplan for that account |
| 20:28.18 | tzanger | it's really not a big deal, especially for the flexibility it offers |
| 20:29.04 | jdoe | to each their own, I guess. |
| 20:29.07 | tzanger | yep |
| 20:29.13 | jdoe | the... core phone stuff you mentioned there? bria does all that. |
| 20:29.29 | tzanger | you don't HAVE to have push support, as i said there are other ways to go about it but it is a pretty nice feature I think |
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| 20:49.29 | drmessano-lt | loves PASSTHRU() in 1.8 |
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| 20:56.55 | j4m3s | I have asterisk with a2b and i'm trying to configure wholesale call routing, but i'm getting a false answer. |
| 20:57.55 | j4m3s | rather, asterisk answers with a 200 OK to a invite before it initiates an invite on the outbound gw to the caller's DNID |
| 20:59.26 | j4m3s | asterisk 1.8 |
| 21:10.52 | leifmadsen | j4m3s: right, because you are probably doing an Answer() or audio playback |
| 21:11.23 | j4m3s | leifmadsen: that's a negative, but it appears the H option will create that behavior |
| 21:11.24 | leifmadsen | Asterisk is a B2BUA so that is typically the behaviour. Make sure you don't answer or anything before the other side answers. You probably want Progress() and not Answer(). |
| 21:11.44 | j4m3s | H option for dial |
| 21:11.46 | leifmadsen | anything which causes Asterisk to answer or listen for audio/dtmf will cause it to send a 200 OK |
| 21:12.01 | leifmadsen | I don't know what H option for dial is because I Haven't memorized all 50 options :) |
| 21:15.12 | j4m3s | leifmadsen: why the hell not leif! |
| 21:15.21 | j4m3s | it's on the quiz |
| 21:15.38 | leifmadsen | j4m3s: I only wrote the quiz, doesn't mean I know the answers! |
| 21:15.41 | leifmadsen | that's what the answer sheet is for! |
| 21:15.47 | j4m3s | cheater |
| 21:15.55 | leifmadsen | no, that's what makes me a teacher! |
| 21:16.01 | j4m3s | all you canadians are the same |
| 21:20.30 | n3hxs-wk | What? Red Coat, funny hat, Big horse. |
| 21:22.50 | keith4 | what happens if I have "exten => 600," and "exten => _6XX," in the same context? |
| 21:23.21 | leifmadsen | 600 has preference over _6XX |
| 21:23.30 | leifmadsen | if you dial 600, exten => 600 will match first |
| 21:23.36 | tzanger | keith4: leifmadsen will have the official answer, but IIRC it is first match wins. they may have fixed that for most specific match wins in later versions though |
| 21:23.37 | leifmadsen | 601 will match _6XX |
| 21:23.42 | tzanger | ah, there you have it |
| 21:24.14 | leifmadsen | 'dialplan show <context>' should show you the preferred matching order too I think |
| 21:24.33 | leifmadsen | or you can do something like: 'dialplan show 600@<context>' and that'll show you what matches |
| 21:24.48 | keith4 | right, i forgot the "and I dial 600" part of the question ;-) |
| 21:25.11 | leifmadsen | I presumed that's what you meant to add |
| 21:25.19 | keith4 | yes. and this is.... 1.4 still |
| 21:25.40 | keith4 | doesn't matter which order the extensions are, in the dialplan, right? |
| 21:26.36 | leifmadsen | nope |
| 21:26.38 | leifmadsen | not at all |
| 21:26.41 | keith4 | okay. thanks |
| 21:26.48 | leifmadsen | the include => order does if you are including contexts |
| 21:27.09 | keith4 | tzanger: any idea what qualifies as a "later version"? |
| 21:28.01 | tyman | configuring * to use gvoice as per http://bit.ly/9Prprf. I'm running into the following error now and i'm not finding hits on google on this: [Feb 1 13:27:44] ERROR[21518]: res_jabber.c:3113 aji_initialize: JABBER ERROR: No DNS guest for client to /asterisk |
| 21:29.57 | leifmadsen | what ftp server do people prefer for polycom firmware serving? vsftpd or proftpd? |
| 21:30.30 | paulc | leifmadsen: I've used vsftpd and it's been great, but mostly because it's what came with the distro rather than a conscious choice |
| 21:30.32 | j4m3s | leifmadsen: asterisk no longer answers the call, now when i get 183 i don't hear the ringing |
| 21:31.05 | leifmadsen | paulc: ya I'm just looking for something easy to setup. I haven't actually setup one of these for over a year and I forget what I used last time :) |
| 21:31.49 | leifmadsen | j4m3s: not sure |
| 21:32.16 | paulc | leifmadsen: got a minute for something unrelated? |
| 21:32.28 | leifmadsen | not really, but you can still try asking :) |
| 21:34.57 | jaytee | quittin time, be back later |
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| 22:05.46 | citywok | i've got a system that's having issues with call quality, i made a wireshark capture of it, and it has intermittent HORRIBLE screeching sounds. |
| 22:06.02 | citywok | the user on the system doesn't hear it, only the person calling in, but i hear it in the pcap |
| 22:22.20 | tyman | has anyone heard of an issue with Cepstral text-to-speech with large (several min) delays on * restarts when using aliased interfaces |
| 22:23.10 | tyman | server is completely offline waiting for cepstral modules to load for ~5 mins since adding aliased interface |
| 22:42.40 | citywok | craaaaaaaaaaaazy, the screeching only happens when i speak something, i hear it come back to me right away. |
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