| 00:00.07 | florz | no, that depends on the number of possible recipients that share the zip code |
| 00:00.42 | *** join/#asterisk jpvoip (~bd55b6c4@gateway/web/freenode/x-mqwfzwytujursfwp) |
| 00:00.55 | p3nguin_ | You didn't say registration was "not just about ip addresses." You said "registration is not about ip addresses." |
| 00:01.05 | p3nguin_ | not about |
| 00:01.20 | p3nguin_ | excluding ip addresses from what it is about. |
| 00:01.33 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 00:02.09 | florz | which is exactly true |
| 00:02.21 | p3nguin_ | And the circle continues. |
| 00:02.27 | florz | ip addresses can appear in the registered URI, but that's not necessarily the case |
| 00:02.54 | p3nguin_ | If you have excluded it, then you've declared that it's not even possible. |
| 00:03.01 | p3nguin_ | And you did exclude it. |
| 00:03.06 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:03.17 | p3nguin_ | But now you're trying to say it is possible. |
| 00:04.11 | florz | so, apple juice is about electromagnetic fields, for they do occur in apple juice, right? |
| 00:04.51 | florz | it's simply the wrong layer of abstraction |
| 00:04.54 | p3nguin_ | If you've stated that apple juice is not about electromagnetic fields, then they must not occur within applie juice. |
| 00:07.04 | florz | well, maybe it's just my understanding of the English language that's lacking there ... but that really sounds like a strange way to view things ;-) |
| 00:09.05 | p3nguin_ | So registration is about IP addresses, but not ONLY about IP addresses. Does that seem like a fair statement? |
| 00:09.25 | florz | not even necessarily |
| 00:09.48 | Kobaz | p3nguin_'s statement is correct |
| 00:09.50 | florz | registration can still delegate ip address resolution to the DNS |
| 00:10.02 | *** join/#asterisk Z_God (~julius@wlan229109.mobiel.utwente.nl) |
| 00:10.17 | Z_God | can anyone explain me how to use the resample codec in asterisk? |
| 00:10.48 | p3nguin_ | eh, "the resample codec" is what, now? |
| 00:10.58 | Kobaz | 'explain to me' would be the proper phrasing |
| 00:11.25 | Z_God | sorry, I mean codec_resample |
| 00:11.27 | p3nguin_ | codec_resample.so? |
| 00:11.32 | Z_God | yeah |
| 00:11.55 | Z_God | I would like to resample so that I can interoperate with phones which only support wideband |
| 00:11.58 | p3nguin_ | It isn't something I have, so I didn't know that was really a codec name. |
| 00:12.42 | Z_God | yeah, it must seem confusing indeed |
| 00:12.53 | Z_God | I'd expect it to be used in combination with speex for example |
| 00:13.03 | Z_God | but I can't seem to find any docs on it |
| 00:13.14 | Kobaz | there's always the source code |
| 00:13.25 | florz | (and BTW, in the original context, Manxpower's statement was essentially that registration was pointless because the IP address was constant - which really only would make sense if registration was about ip addresses only, and in that context, I would argue that it was pretty clear what I was trying to say) |
| 00:13.27 | file | there aren't any docs because you don't use it, the translation core internally will use it as necessary |
| 00:13.50 | Z_God | file: that seems good, but it doesn't work yet |
| 00:13.59 | Z_God | do I need to enable it somewhere? |
| 00:14.01 | Z_God | or allow it? |
| 00:14.17 | Z_God | frame.c:1410 speex_samples: Not enough bits remaining after wide band for speex samples. |
| 00:14.25 | Z_God | this is the error I get now & stuttering audio |
| 00:15.40 | file | codec_speex doesn't support wideband afaik |
| 00:15.59 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 00:16.19 | Z_God | file: ok then it's clear, thanks :) |
| 00:19.24 | Katty | howdy |
| 00:19.27 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 00:20.42 | coppice | Z_God: what phone only supports wideband? |
| 00:20.58 | Z_God | coppice: http://www.psi-im.org/ |
| 00:21.16 | Katty | i have snickerdoodle cookies in the oven. |
| 00:21.17 | Z_God | I'm already hacking it to let it support narrowband too, so I'll just continue down that path for now |
| 00:22.46 | TJNII | Ooh. Snickerdoodles. |
| 00:25.50 | *** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh) |
| 00:28.22 | p3nguin_ | Hmm, I use Psi, but I didn't know it supported voice. |
| 00:28.56 | Z_God | p3nguin_: since version 0.13 |
| 00:29.06 | Z_God | I've got it working with asterisk here |
| 00:29.11 | Z_God | but with hacks on both sides ;) |
| 00:29.34 | coppice | Z_God: the web site isn't exactly advertising that ability |
| 00:30.20 | Z_God | coppice: true :) |
| 00:30.41 | Z_God | seems it's only in the changelog |
| 00:31.12 | Z_God | the problem on the asterisk side is that it doesn't answer a stun request properly btw |
| 00:31.50 | Z_God | I hacked psi now to ignore that, but I guess this should be fixed in asterisk |
| 00:32.17 | Z_God | I have also had mail contact with the jingle maintainer, but it seems he hasn't been able to find it either yet |
| 00:33.23 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 00:34.43 | roe | I may be missing them, but I am surprised there aren't thunderbird plugins to enable C-2-D through the managers interface |
| 00:35.06 | roe | I remember there used to be a snapanumber plugin, but now that site just redirects to the forums |
| 00:41.52 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-161-207.dsl.stlsmo.sbcglobal.net) |
| 00:44.00 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-162-219.nycmny.east.verizon.net) |
| 00:44.13 | KingDavidNYC | hello everybody |
| 00:47.17 | Katty | ATTENTION |
| 00:47.27 | Katty | I HAVE WARM SNICKERDOODLE COOKIES STRAIGHT FROM THE OVEN |
| 00:47.37 | KingDavidNYC | Attention Madam |
| 00:47.45 | Katty | that is all. |
| 00:48.19 | ChannelZ | I made brownies |
| 00:48.31 | Katty | yum. |
| 00:48.35 | Katty | did you follow a recipe? |
| 00:48.37 | ChannelZ | Mint. |
| 00:48.41 | KingDavidNYC | everyone please gather in the kitchen to pickup your cookies |
| 00:48.41 | ChannelZ | Yeah the one on the box |
| 00:48.45 | Katty | lol |
| 00:49.12 | Katty | there's a local chain pizza place here called Cici's |
| 00:49.19 | Katty | their pizza sucks majorly, but their brownies are wonderful. |
| 00:49.49 | ChannelZ | I've seen commercials but never been there |
| 00:50.04 | Katty | it's a place where you'll find screaming, bratty children |
| 00:50.10 | Katty | demanding more quarters for the arcade machines |
| 00:50.37 | Katty | and cheaply made pizza with minimal toppings on their buffet |
| 00:52.25 | ChannelZ | Sounds like Chuck-E-Cheese |
| 00:52.58 | Kobaz | ~peer |
| 00:52.58 | infobot | rumour has it, peer is the most elusive script kiddie this side of Jupiter |
| 00:53.04 | Kobaz | ~friend |
| 00:53.05 | infobot | man, /usr/doc/<package>/*, what else, or the apache error log |
| 00:53.23 | Katty | ChannelZ: never been there |
| 00:53.30 | *** join/#asterisk geneticx (~geneticx@c-75-74-66-161.hsd1.fl.comcast.net) |
| 00:53.35 | ChannelZ | They have robots! |
| 00:54.48 | Katty | is their pizza awful? |
| 00:54.54 | ChannelZ | yes |
| 00:54.55 | Katty | 90% bread. |
| 00:55.00 | Katty | ugah. |
| 00:55.16 | ChannelZ | ironically I just had pizza |
| 00:55.33 | *** join/#asterisk lost_soul (~noymfb@cpe-67-241-68-202.twcny.res.rr.com) |
| 00:55.36 | Katty | (= |
| 00:55.54 | Katty | Mmmmm pizza |
| 00:56.14 | Katty | Dear Universe, thank you for pizza |
| 00:56.24 | Katty | so. |
| 00:56.29 | Katty | any xbox 360 players in here |
| 00:57.22 | ChannelZ | is a PS3 man |
| 00:58.45 | Katty | well i don't have a ps3. |
| 00:58.55 | Katty | do ps3s have internet related things. |
| 00:59.02 | Katty | like xbox 'live' |
| 00:59.26 | ChannelZ | Yes-ish, I'm not sure what all Live does |
| 00:59.45 | ChannelZ | But some games are multiplayer over the net, you can have 'friends' and chat with people, etc. |
| 01:00.05 | ChannelZ | There is a 'Playstation Home' which is like a virtual world you can run around in and talk to people |
| 01:00.09 | ChannelZ | And pay real money for virtual crap |
| 01:02.40 | Katty | hmm. |
| 01:02.51 | Katty | sounds like xbox live, but i'm not sure if they have any sort of 'xbox home' thing |
| 01:03.48 | Katty | it does have a media center externder |
| 01:03.50 | Katty | extender. |
| 01:04.08 | *** part/#asterisk kotp (~vgoff@96.2.187.66) |
| 01:07.29 | ChannelZ | yeah the PS3 can play a bunch of media files, I also use a little app on my computer that makes stuff on my computer show up as a network media server |
| 01:08.00 | ChannelZ | So it'll for instance play h264 .mkv files over the network |
| 01:17.30 | Katty | how about an nes/snes emulator? |
| 01:20.50 | ChannelZ | Dunno. There might be one you can buy |
| 01:20.52 | *** join/#asterisk knctrnl (~aembrey@user-69-1-12-224.knology.net) |
| 01:21.04 | Katty | well. |
| 01:21.09 | Katty | i have an emulator for xp |
| 01:21.13 | ChannelZ | There's an playstation store that sells all kinds of add-ons and full games, and even movies and tv shows |
| 01:21.19 | Katty | and there's nescafe which is an emulator for the windows media center. |
| 01:21.22 | Katty | so....technically |
| 01:21.40 | Katty | i can play nes games on the xbox using the windows media center extender thingy |
| 01:21.59 | Katty | but i think i'd rather have a nice wireless game thingy |
| 01:22.01 | ChannelZ | that's cool |
| 01:23.01 | Katty | yeah i just need to find one i guess |
| 01:23.45 | Katty | newegg probably has one |
| 01:24.06 | Katty | and i already have a windows media center workstation connected to the tv |
| 01:26.47 | Katty | checks newegg |
| 01:29.52 | ChannelZ | I think I'm confused at this point about what you're looking for :) |
| 01:30.21 | jpvoip | Hello guys.. im searching for documents that talks about Asterisk + XMPP, XMPP vs SIMPLE, and other Unified Communications topics relationed to Asterisk.... anyone has any suggestion? |
| 01:34.10 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 01:34.58 | Katty | http://www.newegg.com/Product/Product.aspx?Item=N82E16826127209 |
| 01:36.34 | ChannelZ | oh.. a controller for the PC, I get it now |
| 01:36.46 | Katty | yesh |
| 01:36.55 | Katty | the highest rated seems to be a usb xbox controller |
| 01:36.58 | Katty | but i want wireless |
| 01:37.53 | Katty | so that logitech is the next best |
| 01:38.48 | ChannelZ | looks like the ps3 controller :) |
| 01:38.58 | Katty | does it? |
| 01:39.17 | Katty | lol |
| 01:39.22 | Katty | i googled NEScafe for the emulator |
| 01:39.23 | ChannelZ | Yah. Though they all kinda look alike |
| 01:39.27 | Katty | and it gave me nescafe |
| 01:39.35 | Katty | isntant coffee |
| 01:40.09 | ChannelZ | heh |
| 01:40.18 | ChannelZ | Google no case sensitive |
| 01:40.29 | Katty | it's okay |
| 01:40.31 | Katty | i just found it funny |
| 01:48.21 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
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| 01:55.01 | *** join/#asterisk [8none1] (~8none1]@ps14528.dreamhost.com) |
| 02:13.00 | LemensTS | transfers and parked calls suck |
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| 04:17.28 | pawz | Hey I'm getting 400 Bad Request back from my provider |
| 04:17.37 | pawz | can anyone tell why, from this trace: http://pastebin.com/U47qmkHx |
| 04:39.30 | *** join/#asterisk [8none1] (~8none1]@ps14528.dreamhost.com) |
| 04:43.00 | ChannelZ | What are all these periods at the end of the lines? |
| 04:47.27 | TJNII | Grammar is important and should not be overlooked, even when debugging. |
| 04:49.10 | pawz | periods ? |
| 04:50.21 | pawz | hrmm they are in the trace |
| 04:50.29 | pawz | i don't know why |
| 04:52.29 | pawz | basically it seems my asterisk box is sending an OPTIONS packet.. |
| 04:52.37 | pawz | and the server responds with 400 Bad Request |
| 04:53.32 | ChannelZ | well I'm not enough of an expert with SIP to know why, but it's odd to be putting your LAN address in all the headers (I think) |
| 05:02.49 | idespinner | pawz, do you have qualify=yes in your peer definition or sip.conf? |
| 05:02.56 | pawz | yes |
| 05:03.26 | idespinner | i think the qualify statement causes asterisk to send out SIP Options packets... |
| 05:03.30 | idespinner | i could be wrong |
| 05:04.15 | pawz | oh my god |
| 05:04.18 | idespinner | looks like im right |
| 05:04.24 | idespinner | http://www.voip-info.org/wiki/view/Asterisk+sip+qualify |
| 05:04.34 | idespinner | "By sending the OPTIONS request, the UDP port binding in the NAT (on the outside address of the NAT/firewall device) is maintained by sending traffic through it" |
| 05:04.44 | pawz | i deleted it all and added it again and this time added a register string |
| 05:04.44 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 05:04.50 | pawz | and it seems to have worked |
| 05:05.03 | pawz | i don't know if it will KEEP working, but it's stopped getting 400 Bad Request |
| 05:05.33 | pawz | I don't know what I was doing to cause that, but I deleted and re-added the trunk at least 20 times with the same settings |
| 05:06.19 | pawz | yup, she's good as gold now |
| 05:06.37 | pawz | that's so frustrating when you do the same thing over and over and sometimes it works and sometimes it doesn't |
| 05:06.52 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 05:07.10 | p3nguin_ | Obviously it wasn't "the same thing" if you end up with different results. |
| 05:07.32 | pawz | yeah. i guess so. but whatever the difference was, i wasn't aware of it |
| 05:07.32 | ChannelZ | if you weren't registering with them it was probably because you weren't auth'd on their side |
| 05:09.41 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 05:16.58 | idespinner | for the future you can see what the 'current' settings are with sip show registry and sip show peer [peername] |
| 05:17.40 | pawz | yeah i know those two |
| 05:26.20 | hipitihop | my /var/log/asterisk/cdr-csv/Master.csv shows GMT times wha tis the correct way to set * timezone |
| 05:27.59 | hipitihop | or is there another facility I should use to check call logs ? I'm assuming there is no http interface or reporting facility |
| 05:30.27 | utahsaint | hipitihop: do you have your time on the asterisk box set correctly? |
| 05:31.32 | utahsaint | also it might be a good idea to get the box to sync up with an ntp server |
| 05:31.35 | hipitihop | utahsaint, I believe so... it is also a mythtv box and all seems fine... evem if in cli I do 'sip show registry' then the last registraiton time reported is using my tz |
| 05:31.53 | *** join/#asterisk smooth_penguin (~smoove@59.95.8.102) |
| 05:32.15 | hipitihop | so it seems the csv log is at fault as opposed to my box or other parts of * |
| 05:32.35 | utahsaint | so if you type 'date' from the cli do you get the correct date/time? |
| 05:33.43 | ManxPower-work | you want your logs in GMT |
| 05:34.41 | hipitihop | date is not recongnised. |
| 05:35.28 | utahsaint | linux CLI not Asterisk CLI |
| 05:35.53 | hipitihop | ManxPower-work, probably right in terms of aactual storage but would be nice to easily see report cast in my timezezone, hence looking at what hhtp if might offer |
| 05:36.22 | ManxPower-work | hipitihop, It is the reporting tool's job to convert into local timezone. |
| 05:36.50 | hipitihop | ManxPower-work, indeed that makes sense... does such a tool exist ? |
| 05:37.13 | ManxPower-work | I'm pretty sure there are thousands of tools you can use to generate reports from CSV files. |
| 05:37.21 | ManxPower-work | I don't bill for calls so it's not an issue. |
| 05:37.50 | *** join/#asterisk Tulga (~chatzilla@203.91.113.10) |
| 05:37.53 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 05:37.59 | hipitihop | utahsaint, yes date/time reported in linux cli is correct details but actually says EST, whereas it should be whatever the equivalent of GTM+10:00 |
| 05:38.11 | Tulga | how to configure asterisk as startup on ubuntu 9.04? |
| 05:38.34 | ManxPower-work | Tulga, "make config" |
| 05:38.37 | utahsaint | who hipitihop |
| 05:38.44 | utahsaint | er.. |
| 05:38.49 | utahsaint | hipitihop: EST would be GMT -5:00, |
| 05:38.51 | ManxPower-work | then whatever you do on your distro 8-) |
| 05:39.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:39.30 | hipitihop | Tulga, typical is an entry is in /etc/init.d/ however ubuntu now uses upstart so I would look that up. |
| 05:41.26 | hipitihop | Tulga, my build from source automatically added a /etc/init.d/asterisk |
| 05:41.43 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 05:41.45 | Tulga | I installed it from source. not apt-get |
| 05:41.52 | Tulga | but I think I resolved problem |
| 05:41.57 | Tulga | now restarting server |
| 05:42.56 | hipitihop | utahsaint, I see ... looks like it may be not quite right then .. looks like I may have never set the timezone but set the time to my local time |
| 05:48.40 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 05:52.35 | hipitihop | utahsaint, hmm, just checked my distro via 'dpkg-reconfigure tzdata' and all is as expected, yet the local time is reported as EST so looks like all well appart from confusing me with EST possibly it means Pacific which is correct |
| 05:54.08 | Tulga | hipitihop: how to enter CLI? |
| 05:54.22 | Tulga | I tried asterisk -vvvc, but another asterisk already running background |
| 05:55.16 | hipitihop | Tulga you need to -r to recconect to the running instance |
| 05:56.29 | Tulga | asterisk.ctl not exists on /var/run/asterisk |
| 05:58.33 | hipitihop | tulga not sure about that, but general guide here http://www.voip-info.org/wiki/view/Asterisk+Starting+and+Stopping |
| 06:00.32 | hipitihop | is the http server avaible in 1.6.x ? or only in the 1.4 series ? |
| 06:01.34 | kaldemar | hipitihop: it is in 1.6.x too. |
| 06:02.45 | *** join/#asterisk Arsenick (~y@modemcable022.82-21-96.mc.videotron.ca) |
| 06:03.18 | hipitihop | kaldemar, is there a handy guide as to what it provides out of the box ? |
| 06:07.50 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:07.59 | Tulga | tnx |
| 06:08.39 | kaldemar | hipitihop: doc/tex/asterisk/node205.html |
| 06:09.40 | kaldemar | don't know about the handy part. |
| 06:10.57 | hipitihop | I've seen some references similar but so far failed to find the location of doc/ ... I guess I have to review where asterisk keeps stuff. |
| 06:14.39 | kaldemar | it's a directory in the source package. |
| 06:19.24 | hipitihop | indeed, thanks again. |
| 06:32.32 | KingDavidNYC | quit |
| 06:36.47 | hipitihop | with 'core set debug 3' I should be seeing calls coming and going ? |
| 06:39.09 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 06:42.24 | *** join/#asterisk k5tux (~RussW_K5T@tempest.bluecows.com) |
| 06:44.13 | k5tux | Hi. Anyone here familiar with setting up SIPstation SIP trunks using Asterisk? |
| 06:45.17 | jmcdowell | not i |
| 06:46.00 | kaldemar | hipitihop: core set verbose |
| 06:48.59 | k5tux | Or what chan_sip.c "Forbidden" errors are? |
| 07:00.57 | *** join/#asterisk godsyn (~godsyn@c-67-167-168-141.hsd1.tn.comcast.net) |
| 07:03.53 | godsyn | Throw me a bone, and I'll go google-ing. I want to play audio, IE: mp3, to both parties on a call. IE: user presses #, and a predefined message plays.. after the call is connected (not an IVR). All I can think of is creating a convoluted mess with MEETME.. please tell me there is a better way. |
| 07:12.14 | *** join/#asterisk Chris-NB (~chris@85-126-61-10.work.xdsl-line.inode.at) |
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| 07:42.29 | godsyn | Throw me a bone, and I'll go google-ing. I want to play audio, IE: mp3, to both parties on a call. IE: user presses #, and a predefined message plays.. after the call is connected (not an IVR). All I can think of is creating a convoluted mess with MEETME.. please tell me there is a better way. I'm wanting to insert audio into an active call. |
| 07:54.33 | creativx | godsyn: could you repeat that |
| 07:58.47 | godsyn | creativx: Eh? Im sorry if i'm unclear. I've been teaching myself and don't know the proper terms. I'm looking for a means to inject audio (recordings) into an active call. |
| 08:01.21 | godsyn | If it were an IVR, it would be as simple as "play" or "background", but I can't think of a means to do so to a bridged call. |
| 08:01.57 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 08:06.05 | godsyn | ie: s,1,wait(23) s,n,play(filename) s,n,wait(34) s,n,play(filename2) |
| 08:07.31 | godsyn | I suppose I could make it an extension, and 3way call them, but I've yet to find out how to create a 3 way call. |
| 08:17.16 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:17.55 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 08:18.00 | DelphiWorld | hi all |
| 08:18.01 | DelphiWorld | any gizmo5 user here? |
| 08:26.25 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:37.45 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 08:44.18 | angryuser | hello, i have incoming call >> asterisk1 Forward >> Asterisk 2. The calls come in g729. Do i need the licenses for the asterisk 1 ? |
| 08:44.45 | angryuser | (i think not but dont remember exactly) |
| 08:48.16 | *** join/#asterisk Rajmohan (~raj@122.164.190.219) |
| 08:50.24 | Rajmohan | hi, do any one know cheap voip provider for us number that i can configure to linksys pap2, and make calls to usa and canada, thanks in advance |
| 08:54.03 | godsyn | quit |
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| 09:22.50 | Rajmohan | hi, do anyone know us viop number provider for linksys pap2, both incoming and outgoing |
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| 10:02.36 | hipitihop | Rajmohan, still there ? |
| 10:02.59 | hipitihop | Rajmohan, never mind, srry just reread your question, pls ignore |
| 10:04.40 | hipitihop | anyone here using the http server ? I'm trying to setup for first time following the docs but seems something still not right, no access on port 8088 |
| 10:05.45 | *** join/#asterisk smooth_penguin (~smoove@59.95.7.205) |
| 10:11.28 | *** join/#asterisk gelpg (~chatzilla@qf.invitel.hu) |
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| 10:14.30 | gelpg | hi, is there any option to pick up a certain exten when it's ringing? If I use the *8, it picks up the last call |
| 10:15.34 | hipitihop | is it possible for an outside caller to dial an extension directly without the need for an interactive menu ? if so can someone point me ot a sample dialplan |
| 10:17.39 | *** join/#asterisk soman (~somnath@212.226.87.141) |
| 10:20.45 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 10:20.59 | kruemeltee | hello :-) |
| 10:21.58 | Rajmohan | hipitihop: iam here pls |
| 10:22.27 | Rajmohan | can you give me some voip service providers for US numbers. |
| 10:22.34 | Rajmohan | that i can configure to linksys pap2 |
| 10:22.59 | *** part/#asterisk manveru (~manveru@b08s28ur.corenetworks.net) |
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| 10:23.20 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.176.205.72) |
| 10:33.54 | angryuser | ~providers |
| 10:33.55 | infobot | somebody said providers was http://www.voipreview.org/service.all2.aspx?Country=1&Area_Code=0&CallingArea=0&provider=0&serviceType=1&Adv=1&Features=43 |
| 10:48.57 | *** join/#asterisk dinesh___ (~dinesh@77-58-221-165.dclient.hispeed.ch) |
| 10:50.28 | dinesh___ | hi folks. is there a good tutorial that explains how to set up an SIP server on asterisk? Right now I configured it to be a SIP client for 1 incoming number, and SIP client for several outgoing ones |
| 10:50.54 | *** join/#asterisk danj1980 (~dan@91.109.112.235) |
| 10:51.06 | danj1980 | Hi all. |
| 10:51.23 | danj1980 | Has anyone had a problem with Polycom phones rebooting during a call? |
| 10:55.41 | tzafrir | hipitihop, is the httpd running? http show status |
| 10:55.51 | tzafrir | hipitihop, and: hi :-) |
| 11:25.44 | Dovid | Rajmohan: You can try jivetel.com and newtelsystems.com both friends of mine |
| 11:27.30 | *** join/#asterisk soman (~somnath@212.226.87.141) |
| 11:40.15 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 11:41.59 | dinesh___ | how do I forward a call from an incoming SIP number to a client registered to the local asterisk server ? |
| 11:42.22 | Dovid | learn how to work asterisk ;) |
| 11:42.24 | Dovid | !book |
| 11:42.26 | Dovid | ~book |
| 11:42.27 | 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 |
| 11:43.57 | dinesh___ | heh I was more looking for a command to use in the extensions ;) |
| 11:44.12 | dinesh___ | I'll download it and do some search thanks |
| 11:49.47 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 11:52.02 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 11:54.07 | redax | hm. I trying to call a fax using SPA-8000, over a SIP trunk, and when the remote party start the fax handshaking voice, the call hangs up immediatly. |
| 11:54.18 | redax | what should be the problem here? |
| 11:55.26 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 11:55.56 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 11:56.47 | jaytee | do you have t38udptl set in your sip.conf? |
| 11:57.56 | redax | jaytee: actuall t38pt_udptl = no, but tried with yes lately |
| 11:57.58 | redax | same effect |
| 11:58.19 | redax | t38pt_rtp = no, t38pt_tcp = no |
| 11:59.36 | *** join/#asterisk aandrade (~aandrade@200.146.4.245.dynamic.dialup.gvt.net.br) |
| 11:59.49 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 12:01.01 | jaytee | what about the line settings on the port on the SPA8000 you're using? |
| 12:07.30 | dinesh___ | hm how to use Transfer() ? I'm trying: xten => 1000,1,Transfer(SIP/@home) as first rule for incoming calls, but it says it requires an extension, I don't know what to put there |
| 12:07.35 | dinesh___ | there's only a single user on "home" |
| 12:08.26 | florz | you essentially should put there the localpart of the SIP URI you are trying to call |
| 12:10.15 | dinesh___ | so I need to define a callerid for my user, and put that callerid as lcoalpart ? |
| 12:11.57 | redax | jaytee: :/ t38 disabled. |
| 12:12.06 | redax | I'll try t38 enabled |
| 12:12.13 | jaytee | it should be enabled on the SPA8000 |
| 12:13.16 | florz | dinesh___: I don't understand what you mean by that - probably you are mixing up concepts ... |
| 12:13.18 | redax | which Fax Passthrough method should one use? ReINVITE or NSE? |
| 12:13.22 | jaytee | I've been using t38 passthrough on asterisk 1.4.x for a couple years now with fax machines on SPA2102s and SPA8000s. |
| 12:13.46 | jaytee | I left that at the default which I believe IIRC is NSE |
| 12:14.21 | *** join/#asterisk vader-- (~me@c-68-36-9-8.hsd1.nj.comcast.net) |
| 12:17.07 | redax | thanks, jaytee now it works. |
| 12:17.22 | jaytee | your welcome, glad to be of assistance |
| 12:18.10 | redax | my collegue claims that SPA8K worked for him withouth t38 enabled a few months earlier :D |
| 12:18.39 | redax | but if the ATA supports t38 what normal reason should be to turn off |
| 12:19.06 | hipitihop | tzafrir, sorry back now ... |
| 12:19.47 | hipitihop | tzafrir, is 'http show status' is that a * cli command ? |
| 12:19.50 | redax | the ReINVITE vs NSE question is easy, if canreinvite=yes for the xtension, you can use both, if canreinvite=no, then only NSE. |
| 12:21.12 | jaytee | ah, didn't know that |
| 12:21.31 | hipitihop | tzafrir, ok tried and get response "Server Enabled and Bound to 127.0.0.1:8088" |
| 12:21.48 | jaytee | time for work, gotta run. |
| 12:22.16 | redax | jaytee: tried, both working here... but canreinvite=yes for the given extension. |
| 12:22.47 | dinesh___ | probably florz, basically I'm trying to transfer each incoming call on a SIP number i'm registered to, to locally registered SIP user (x-lite connected to my asterisk server), without Answering the call first (otherwise I'd just do an Answer() + Dial()) |
| 12:23.28 | hipitihop | is there a chance that I have a conflict with two http servers running ? I also have mythtv running on this box and it has a http serve on the standard port 80 |
| 12:24.03 | dinesh___ | the thing is that it might happen that when an incoming calls comes, the x-lite client is not registered, in that case the caller should hear a proper error message (bla bla is temporary unavailable) |
| 12:24.40 | tzafrir | hipitihop, so it's listening on the loopback interface only (you can't connect to it from any other machine) |
| 12:27.46 | florz | dinesh___: oh, in that case you should simply drop the @ |
| 12:27.50 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 12:28.29 | *** join/#asterisk ralonso (~ralonso@140.Red-88-2-26.staticIP.rima-tde.net) |
| 12:32.01 | ralonso | hi, is posible to attach one file (no vm file) when voicemail send the email |
| 12:32.04 | hipitihop | tzafrir, makes sense however I can't seem to bring it up when I run a browser via a local network ssh session either .. I must be doing something stupid as the otheer hhtp server on the box which does allow other machines and is the default url also does not fire when I do http://localhost |
| 12:33.19 | hipitihop | tzafir, I would have thought there is no conflict between the two since one is on standard port 80 while * is on 8088 |
| 12:34.17 | tzafrir | hipitihop, what address do you look at, exactly? What is the exact error you get? |
| 12:35.41 | hipitihop | tzafrir, if I try http://localhost:8088/httpstatus I get "can't establish a connection to the server at localhost:8088" |
| 12:36.16 | *** join/#asterisk harryv (~harry@67.207.147.205) |
| 12:36.25 | harryv | I'm definitely stupid, but how do I originate a call from the cli w/ 1.6? |
| 12:36.36 | harryv | … I'm using skype for asterisk, so it should go over that. |
| 12:36.46 | tzafrir | originate . just like in 1.4 |
| 12:36.56 | harryv | gives me No such command |
| 12:37.36 | harryv | oh wait. 2s |
| 12:38.48 | hipitihop | tzafrir, the cli show status shows /httpstatus /phoneprov /manager /rawman as enabled uri's |
| 12:39.38 | harryv | still getting No such command. |
| 12:40.40 | hipitihop | tzafrir, I'm going to change the bindaddr in http.conf to actual internal server ip e.g. 192.168.0.105 and restart asterisk |
| 12:44.13 | hipitihop | tzafrir, ok, changed bindaddr to 0.0.0.0 restarted and now I can get in... sorry for the noise |
| 12:46.07 | ralonso | anyone know if is posible attach a file (no vm mensage) when voicemail send the notification email¿? |
| 12:46.45 | *** join/#asterisk HenrikJott (~info@d83-183-134-141.cust.tele2.se) |
| 12:48.24 | HenrikJott | hi! i´m createing call-files for asterisk and i set the WaitTime to 20s, but i seems asterisk only calls for about 15 secs, is WaitTime calculated from when the file was created end does in include the time asterisk needs to connect the call and get ring tones? |
| 12:51.25 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
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| 13:02.28 | *** join/#asterisk voipmonk (~shido6@66.49.232.103) |
| 13:03.06 | *** part/#asterisk harryv (~harry@67.207.147.205) |
| 13:14.46 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 13:16.42 | *** join/#asterisk superc (~superc@88.128.33.206) |
| 13:17.48 | superc | is it common that a phpagi script is terminated when there is a returncode -1 from an EXEC command? |
| 13:18.17 | benngard | did setup a sip trunk to tele2 in sweden, outgoing calls works as a clock but i can get incomming connections from different ip, for both sip-corporate1.tele2.se and sip-corporate2.tele2.se, "host=sip-corporate1.tele2.se" works for outgoing and i i am lucky and the incoming call comes from that box, it works, do i need to create 2 peers? or do u know a better way to do it? |
| 13:18.20 | *** join/#asterisk Pegasus_RPG (~chatzilla@p4FF903B3.dip.t-dialin.net) |
| 13:19.43 | superc | where is the problem? that are in fact 2 different peers? |
| 13:19.52 | *** join/#asterisk [psy] (~psy0rz@lounge.datux.nl) |
| 13:20.22 | benngard | no, it is 1 peer but that peer are having multiple servers on different ip |
| 13:20.24 | Pegasus_RPG | hello. Ever since updating asterisk to 1.6.2.0 (the latest version offered in Debian Testing) I can't receive inbound calls from Broadvoice anymore. The Broadvoice server tells the caller that I'm busy |
| 13:20.28 | [psy] | is it possible to use asterisk-functions directly from the cli, for testing/debugging purposes? |
| 13:21.42 | Pegasus_RPG | and I periodically see > doing dnsmgr_lookup for 'sip.broadvoice.com' about once a minute, though I can place calls just fine and if I turn off Asterisk and tell the SIP phone to register with BV directly, inbound calls work fine. Any idea what the problem might be? |
| 13:28.14 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 13:28.54 | *** join/#asterisk thazza (~thazza@124-254-81-140-static-dsl.ispone.net.au) |
| 13:29.04 | thazza | Hey All. |
| 13:29.07 | adnc | hello, I'm using several sip provider. one of them sends a simple number as caller-id, so people can not call me back. is there a way settings this callerid via asterisk? |
| 13:30.08 | superc | if your carrier is sending a callerid for you you'll most likely not be able to set another one |
| 13:30.33 | superc | there are some carriers without proper callerid handling |
| 13:30.47 | *** join/#asterisk StuZZZs (~stuart@rabbit.dbplc.com) |
| 13:30.53 | thazza | I am looking for an up to date version of the realtime iax and sip table structure.. does anyone know where i can find this? |
| 13:30.58 | adnc | superc, is there a documentation how asterisk can be set up. at least i could try this |
| 13:31.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:31.50 | superc | yes, google for asterisk callerid |
| 13:33.11 | adnc | http://www.voip-info.org/wiki/view/CallerID |
| 13:33.21 | adnc | i had this page, but there is no indication how to do this |
| 13:35.19 | superc | http://www.voip-info.org/wiki/view/Setting+Callerid |
| 13:36.35 | Thazza-Laptop | Anyone know much about asterisk and realtime? |
| 13:36.42 | *** join/#asterisk Thazza (~thazza@124-254-81-140-static-dsl.ispone.net.au) |
| 13:38.29 | tzafrir | Pegasus_RPG, what exact version of Asterisk do you have installed right now? |
| 13:38.40 | tzafrir | Are you connected to Asterisk via SIP? |
| 13:38.54 | Pegasus_RPG | 1.6.2.0-1 |
| 13:38.56 | Pegasus_RPG | I am |
| 13:39.04 | Pegasus_RPG | And this used to work (tm) :) |
| 13:39.12 | [TK]D-Fender | adnc: "core show function CALLERID" |
| 13:39.52 | adnc | [TK]D-Fender, ahh, thank you |
| 13:40.15 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 13:40.28 | Pegasus_RPG | tzafrir: I only see == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 on the * console with -vvvv when there's an incoming call |
| 13:42.39 | soman | Hi, I am using asterisk 1.6.1 with realtime configuration. It use to work quite good. But suddenly the server gets hung when there are calls from any user. and the system needs to be rebooted. I am using mysql for realtime data and using asterisk-addons 1.6.1.0. Can anyone help me regarding why the server is getting hanged? |
| 13:43.38 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 13:44.51 | *** join/#asterisk dunkoh (~dunkoh@rrcs-74-219-209-194.central.biz.rr.com) |
| 13:45.07 | [TK]D-Fender | Pegasus_RPG: "sip set debug on" <------------ |
| 13:47.11 | Pegasus_RPG | when it's just sitting idle, I see this every 30s http://pastebin.ca/1817098 |
| 13:48.05 | ManxPower-work | Contact: <sip:111@10.0.200.15> |
| 13:48.11 | ManxPower-work | Looks like you did not set up your NAT correctly. |
| 13:49.12 | Pegasus_RPG | daah |
| 13:49.13 | soman | I can see an error in asterisk log "res_config_mysql.c: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect" after which the server has been rebooted automatically... can any one help what could be the problem |
| 13:49.27 | ManxPower-work | soman, not many people here use Realtime |
| 13:49.55 | Thazza-Laptop | has notice this recently as well ManxPower-work |
| 13:50.45 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 13:52.07 | [TK]D-Fender | ~sipnat |
| 13:52.07 | infobot | [~sipnat] Quick guide on configuring * + 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 |
| 13:52.09 | [TK]D-Fender | ^^^^^^^ |
| 13:57.13 | Pegasus_RPG | thanks...I added externhost and the contact field looks good (my public address) but it's still not receiving calls |
| 13:57.56 | Pegasus_RPG | and it's still doing the retransmitting thing |
| 13:58.09 | Pegasus_RPG | reads the first link |
| 13:58.33 | *** join/#asterisk Skeeter- (~Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 13:59.46 | ManxPower-work | you need externip or externhost (you can't use an ip in externhost), localnet |
| 14:00.01 | ManxPower-work | As well as the canreinvite option listed in the Asterisk NAT info pages |
| 14:00.46 | Pegasus_RPG | yeah, I have externhost, localnet, and canreinvite=no under the [sip.broadvoice.com] section |
| 14:00.56 | ManxPower-work | This is not rocket scuence, but uou do have to follow the directions |
| 14:01.02 | ManxPower-work | Pegasus_RPG, then you put it in the wrong place/ |
| 14:01.18 | Pegasus_RPG | sorry, asterisk config is quite intimidating |
| 14:01.25 | ManxPower-work | externhost and localnet need to be in [general] JUST LIKE THE EXAMPLES SHOW YOU |
| 14:03.07 | ManxPower-work | Also you really should not name your sip peers after their host names -- it will confuse you. |
| 14:03.14 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:03.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:04.13 | *** join/#asterisk doubletoker (~warlock@adsl-17-233-129.jax.bellsouth.net) |
| 14:04.46 | doubletoker | I was working with someone last night, and his machine |
| 14:05.01 | doubletoker | doesn't have the dialplan reload command |
| 14:05.25 | ManxPower-work | doubletoker, what version of Asterisk |
| 14:05.27 | doubletoker | so we uninstalled asterisk, and reinstalled |
| 14:05.31 | doubletoker | 1.6.2 |
| 14:05.41 | doubletoker | and we had dialplan reload again |
| 14:05.49 | ManxPower-work | that usually means you screwed up the first line of extensions.conf |
| 14:05.49 | doubletoker | then we stopped the service |
| 14:06.09 | doubletoker | really? |
| 14:06.26 | doubletoker | our first line is [general] |
| 14:06.29 | ManxPower-work | yup. Could be a different basic messed up config. |
| 14:06.54 | ManxPower-work | doubletoker, but if your first like was, for example " [general]" that would fail, since the first char is a space |
| 14:07.09 | doubletoker | true |
| 14:07.31 | doubletoker | would it matter if the first line was "[general] " |
| 14:07.39 | ManxPower-work | Asterisk requires [general] to be the first "context", then [globals] then everything else. |
| 14:07.47 | ManxPower-work | doubletoker, it might. Asterisk cares about spaces |
| 14:08.15 | ManxPower-work | All the editors I use remove trailing spaces from lines when you save the file. |
| 14:08.56 | doubletoker | yea, he was inside a wm and using gedit I believe |
| 14:09.16 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:09.39 | *** join/#asterisk smooth_penguin (~smoove@59.95.7.205) |
| 14:11.01 | doubletoker | he's asleep now figured I would try to find out, what it was, but like if we reinstalled, it would work, till reboot, so I reinstalled and just stop it and restart and it's missing, but thanks for you help, we'll try that |
| 14:11.14 | doubletoker | to see if it fixes it, I hope so |
| 14:11.32 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 14:11.33 | Kobaz | so many bugs |
| 14:11.45 | Kobaz | it's gonna take a week just to submit them all |
| 14:11.57 | ManxPower-work | Kobaz, and the rest of your life to defend the bugs. |
| 14:12.27 | Kobaz | yeap |
| 14:13.10 | Kobaz | i was at the office till 12 last night, trying to figure out the various differences between 1.6.0.19 and 1.6.0.25 |
| 14:13.29 | Kobaz | sometimes i'm depending on bugs that were fixed, and other times there's just new bugs |
| 14:14.30 | Kobaz | the key is though... applications should be able to work the same from minor version to minor version |
| 14:14.33 | Kobaz | but that's never the case |
| 14:16.53 | *** part/#asterisk Thazza-Laptop (~thazza@124-254-81-140-static-dsl.ispone.net.au) |
| 14:21.20 | ManxPower-work | Eventually you'll learn to stop upgrading. 8-| |
| 14:21.46 | Kobaz | well |
| 14:21.48 | Katty | infobot: hi |
| 14:21.49 | infobot | hello, katty |
| 14:21.50 | Katty | infobot: thebook |
| 14:21.51 | infobot | from memory, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 14:21.55 | Katty | infobot: thankyoudear. |
| 14:21.55 | infobot | pas de quoi, Katty |
| 14:22.00 | Kobaz | the problem is... various bugs that are killing me, tend to get fixed in new versions |
| 14:22.16 | Kobaz | so i try a new version, run my test suites, do lots of experiments, and uncover 283974892374982734 new bugs |
| 14:24.16 | beek | hugs Katty |
| 14:24.59 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 14:25.00 | Pegasus_RPG | Sheez, I have all the items specified in the examples and have fixed my outside IP, but it still keeps trying to reregister: http://pastebin.ca/1817149 |
| 14:25.14 | coppice | relax. once the bugs have killed you, you'll have no more worries |
| 14:25.22 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 14:27.49 | Katty | hugs benngard |
| 14:27.50 | Katty | oh |
| 14:27.53 | Katty | hugs beek, too |
| 14:28.55 | ManxPower-work | Yay! Olympics are over! Television can get back to normal |
| 14:29.04 | tuxx- | whiej |
| 14:30.07 | leifmadsen | I'm kinda sad the Olympics are over. I was having a great time watching them. |
| 14:30.54 | coppice | I guess the snow was waiting for the olympics to end |
| 14:31.33 | ManxPower-work | I didn't watch even a single min of them. |
| 14:31.51 | leifmadsen | ManxPower-work: good for you, we're all impressed |
| 14:32.01 | ManxPower-work | hugs his TiVo |
| 14:33.23 | *** join/#asterisk sulex (~sulex@88-149-154-95.static.ngi.it) |
| 14:34.22 | Pegasus_RPG | gives up for now and shuts down Asterisk |
| 14:35.28 | beek | ManxPower-work: TiVo rocks! |
| 14:35.46 | *** join/#asterisk doolittlework (~d@196.211.34.2) |
| 14:36.12 | doolittlework | join #mysql |
| 14:36.38 | Pegasus_RPG | I don't get it...my SIP phone's contact field has the private IP address |
| 14:36.44 | Pegasus_RPG | is a port number required? |
| 14:36.56 | ManxPower-work | Pegasus_RPG, put a copy of your sip.conf on pastebin.ca masking ONLY passwords |
| 14:39.51 | Pegasus_RPG | http://pastebin.ca/1817172 |
| 14:40.09 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-17-NewEngland.hfc.comcastbusiness.net) |
| 14:40.39 | [TK]D-Fender | Pegasus_RPG: .... you hve to have EVERYTHING under [general] BEFORE your REGISTER directives <----- |
| 14:40.47 | [TK]D-Fender | Pegasus_RPG: Everything below is discarded <- |
| 14:40.59 | Pegasus_RPG | [TK]D-Fender: ok. I did try it that way before with no improvement |
| 14:41.43 | [TK]D-Fender | Pegasus_RPG: What do you have forwarded to your server precisely? |
| 14:41.55 | Pegasus_RPG | firewall-wise? |
| 14:42.31 | ManxPower-work | Pegasus_RPG, http://pastebin.ca/1817173 |
| 14:42.36 | Pegasus_RPG | 69 UDP, 5060-5063 UDP, 10000-20000 UDP |
| 14:42.39 | ManxPower-work | stop addindg extra crap to your config |
| 14:42.47 | ManxPower-work | Pegasus_RPG, what is 69/UDP for? |
| 14:43.02 | Pegasus_RPG | I dunno I saw it in a forum post |
| 14:43.10 | ManxPower-work | Now we know why it's not working for you. |
| 14:43.18 | ManxPower-work | UNDERSTAND the options you add. |
| 14:43.32 | ManxPower-work | see if the modified sip.conf I just gave you solves the issue. |
| 14:43.33 | Pegasus_RPG | the stuff under register is because this server is also an OpenVPN for remote clients to connect |
| 14:43.36 | Pegasus_RPG | and use SIP |
| 14:43.45 | ManxPower-work | "under register" means nothing. |
| 14:43.46 | Pegasus_RPG | and it works (or did the last time someone tried it) |
| 14:44.52 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 14:45.00 | Katty | hrmmm |
| 14:45.02 | Katty | okay folks |
| 14:45.04 | Pegasus_RPG | k, testing |
| 14:45.07 | Katty | do i want this: http://www.newegg.com/Product/Product.aspx?Item=N82E16830120352 |
| 14:45.15 | Katty | or do i want this: http://www.newegg.com/Product/Product.aspx?Item=N82E16830120262 |
| 14:45.40 | Katty | ^- note: for mother. |
| 14:46.45 | doolittlework | Katty she wants this http://www.hawaiipackage.com/ |
| 14:47.25 | Katty | she wants a camera. |
| 14:47.53 | doubletoker | which tts is better with asterisk? |
| 14:48.01 | [TK]D-Fender | Katty: What is she going to do with it? |
| 14:48.17 | [TK]D-Fender | doubletoker: Cepstral |
| 14:48.24 | doolittlework | i have one of these Katty: must say i love it http://www.dpreview.com/reviews/canoneos1dsmkii/ |
| 14:48.40 | Katty | [TK]D-Fender: you know...go to the zoo |
| 14:48.44 | Katty | [TK]D-Fender: family occasions |
| 14:48.48 | doolittlework | abit bulky but gets the job done |
| 14:48.59 | Katty | [TK]D-Fender: possibly photograph the bunnies in the back yard, and deer on the side of the road. |
| 14:49.09 | *** join/#asterisk moy (~chatzilla@74.12.129.100) |
| 14:49.13 | [TK]D-Fender | Katty: forget both and get her a 10x+ P&S model |
| 14:49.14 | Katty | [TK]D-Fender: usual mom things. |
| 14:49.21 | Katty | [TK]D-Fender: link? |
| 14:50.07 | doolittlework | Katty: for the deer i recomment this for a perfect shot http://www.chuckhawks.com/6x6.htm |
| 14:50.17 | Pegasus_RPG | ManxPower-work: that file doesn't work either. |
| 14:50.35 | ManxPower-work | Pegasus_RPG, pastebin a new sip debug |
| 14:50.43 | Pegasus_RPG | yep in progress |
| 14:50.50 | Katty | doolittlework: you make me sad. |
| 14:51.06 | doolittlework | lol |
| 14:51.38 | doolittlework | just kidding canon powershot good all round cammera |
| 14:51.40 | Katty | [TK]D-Fender: "powershot"? |
| 14:51.49 | Pegasus_RPG | ManxPower-work: http://pastebin.ca/1817188 |
| 14:51.55 | [TK]D-Fender | Katty: http://www.newegg.com/Product/Product.aspx?Item=N82E16830120380 |
| 14:52.04 | [TK]D-Fender | Katty: Forget useless modelnames like that.. |
| 14:52.17 | Pegasus_RPG | uri="sip:broadvoice" is that a problem? |
| 14:52.32 | [TK]D-Fender | Katty: Virtually all near-spec models from any reputable manufacturer are about 1% different from each other |
| 14:52.35 | ManxPower-work | Pegasus_RPG, set srvlookup=no in [general] BEFORE the register. |
| 14:53.25 | ManxPower-work | Pegasus_RPG, try upgrading too. |
| 14:53.35 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:53.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:53.41 | [TK]D-Fender | Katty: http://www.newegg.com/Product/Product.aspx?Item=N82E16830120340 |
| 14:54.15 | ManxPower-work | Pegasus_RPG, at this point you may have a firewall issue. |
| 14:54.44 | Pegasus_RPG | ugh, then its my ISP since I was using the same firewall before I moved |
| 14:54.47 | *** join/#asterisk high-freq (~hfreq@99.188.122.87) |
| 14:55.14 | [TK]D-Fender | Pegasus_RPG: pastebi yoru new configs |
| 14:55.17 | Pegasus_RPG | but the question remains: how can my SIP phone work fine? |
| 14:55.54 | ManxPower-work | Pegasus_RPG, Registration only notifies the remote server what your IP address is. It has nothing with being able to PLACE calls via your service provider |
| 14:55.55 | [TK]D-Fender | Pegasus_RPG: From where? |
| 14:56.18 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 14:57.06 | Pegasus_RPG | from the same subnet |
| 14:57.25 | Pegasus_RPG | And I can _place_ calls just fine. I can't _receive_ them into * |
| 14:57.29 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 14:57.48 | Pegasus_RPG | (or message waiting indicator for that matter) |
| 14:58.37 | ManxPower-work | 1) Pegasus_RPG, what happens if you turn off your firewall 2) did you mention this before? |
| 14:58.46 | Pegasus_RPG | I did mention this |
| 14:59.14 | Pegasus_RPG | Turning off the firewall (the SPI section) didn't make a diff either but I'll test again |
| 14:59.41 | ManxPower-work | turning off just the "SPI section" is not "turning off the firewall" |
| 14:59.59 | ManxPower-work | which firewall do you have? |
| 15:00.19 | Katty | [TK]D-Fender: hmm. |
| 15:00.37 | Pegasus_RPG | WRT310N |
| 15:00.37 | Katty | [TK]D-Fender: does that sx20 allow for lense attachment by chance? |
| 15:00.39 | Pegasus_RPG | v1 |
| 15:00.49 | Pegasus_RPG | I just put the * into the DMX |
| 15:00.51 | Pegasus_RPG | Z |
| 15:00.53 | ManxPower-work | Pegasus_RPG, on the asterisk server type "service iptables stop" |
| 15:01.01 | [TK]D-Fender | Katty: generally no. |
| 15:01.04 | ManxPower-work | putting Asterisk in the DMZ does very little to help anything. |
| 15:01.17 | Katty | [TK]D-Fender: bummer. not that she really needs lenses |
| 15:01.19 | Katty | [TK]D-Fender: but i am spoiled. |
| 15:01.25 | Katty | [TK]D-Fender: also, i need to show you a new photo i took (= |
| 15:01.34 | [TK]D-Fender | Katty: But this is your mother you're talking about... Zoo pictures. from a non-techie who won't want to drab a dedicated camera bag or spend a fortune |
| 15:01.34 | Pegasus_RPG | ManxPower-work: "iptables: unrecognized service" |
| 15:01.48 | [TK]D-Fender | [10:01]<Katty>[TK]D-Fender: bummer. not that she really needs lenses <- O RLY? Why? |
| 15:02.08 | Katty | she won't use lenses |
| 15:02.09 | ManxPower-work | Pegasus_RPG, try "iptables -L -v" |
| 15:02.10 | [TK]D-Fender | Katty: Moms = P&S users |
| 15:02.28 | [TK]D-Fender | Katty: Sorry, read that backwards.. |
| 15:03.20 | Katty | [TK]D-Fender: http://farm4.static.flickr.com/3639/3427177095_e20e1d1cfe_b.jpg <- |
| 15:03.20 | Pegasus_RPG | ManxPower-work: all blank tables |
| 15:03.20 | [TK]D-Fender | Katty: So get her the big zoom model of some sort so she won't comlpain about not getting a good shot of that tiger 50 yards away |
| 15:03.20 | dmz | ManxPower-work check out www.fwbuilder.org, much easier than manual editing rules |
| 15:03.20 | Katty | [TK]D-Fender: http://farm4.static.flickr.com/3578/3427175487_3297f4f588_b.jpg |
| 15:03.26 | [TK]D-Fender | Katty: that off the 50mm macro? |
| 15:03.56 | Katty | [TK]D-Fender: no |
| 15:04.02 | Katty | [TK]D-Fender: that's just straight of the camera |
| 15:04.13 | Katty | [TK]D-Fender: well friends camera. |
| 15:04.19 | Katty | [TK]D-Fender: she came by and i could help but tinker with it |
| 15:04.39 | [TK]D-Fender | Katty: What distance/ |
| 15:04.44 | Katty | [TK]D-Fender: 5.9mm focal length |
| 15:04.45 | [TK]D-Fender | Katty: there's no EXIF on that |
| 15:04.47 | ManxPower-work | dmz, Never found a "gui" that made setting up a firewall "easier". Well, easier in "I have no ide what I'm doing, I hope the computer does" |
| 15:04.57 | Pegasus_RPG | haha |
| 15:05.02 | *** join/#asterisk emyrddin (~54fda312@gateway/web/freenode/x-vdnjqxjyewtnnpwh) |
| 15:05.06 | [TK]D-Fender | Katty: 5.9? no chance I'll buy that as the effective |
| 15:05.24 | Pegasus_RPG | I could try using dd-wrt |
| 15:05.39 | [TK]D-Fender | Katty: You'd have some nasty FOV distortion if you were that low scaled back |
| 15:05.47 | ManxPower-work | Pegasus_RPG, Um, of you can't make simples calls into and out of Asteisk then you do not have a firewall/NAT issue. |
| 15:05.56 | Katty | [TK]D-Fender: dunno. the camera was a finepix f480, if that helps |
| 15:06.20 | Katty | [TK]D-Fender: auto exposure |
| 15:06.45 | emyrddin | hi folks, anyone here with asterisk+grandstream 2010? |
| 15:06.48 | ManxPower-work | Heh! Today's Word of the Day is "Asterisk" |
| 15:07.26 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 15:07.34 | Katty | [TK]D-Fender: http://farm4.static.flickr.com/3655/3427172977_848864fdd8_b.jpg |
| 15:09.06 | Katty | [TK]D-Fender: i think the camera did a great job consider it's cheap |
| 15:09.10 | [TK]D-Fender | Katty: f=4.6mm - 18.4mm, Equivalent to 28-112mm on a 35mm camera |
| 15:09.13 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 15:09.37 | [TK]D-Fender | Katty: 28mm =~ 18mm on digital, which is fairly wide... |
| 15:09.48 | [TK]D-Fender | (APS-C)\ |
| 15:10.29 | [TK]D-Fender | Katty: Is your mom likely to be doing a decent amount of low-light shooting? |
| 15:10.45 | Katty | hmm. |
| 15:10.47 | Katty | possibly |
| 15:10.56 | [TK]D-Fender | Katty: think long and hard on that... |
| 15:11.02 | Katty | yeah |
| 15:11.12 | [TK]D-Fender | Katty: thats where DSLR starts paying off. |
| 15:11.22 | beek | Katty: Just get a Nikon D3 and be done with it. |
| 15:11.25 | emyrddin | could i compose numbers with my computer using the gxp phone? |
| 15:11.26 | dmz | ManxPower-work, heh |
| 15:11.51 | Katty | [TK]D-Fender: i don't want to get something too confusing for her tho |
| 15:12.20 | ManxPower-work | emyrddin, you are not making any sense. |
| 15:12.31 | Katty | beek: i want to get her a camera, not a keychain :P |
| 15:12.36 | *** join/#asterisk ellisdee (~ellisdee@cosmic.sized.penisinyourface.com) |
| 15:13.26 | Katty | meh, i'll just get her this sx20 |
| 15:13.39 | Katty | and a tripod |
| 15:13.40 | [TK]D-Fender | Katty: I've got a 70-210 F4 for that which I haven't really used.... I use an 18-250mm F3.5-6.3. lets jsut say you end up in the 5's by around 120mm easily... and to be sharp you'll up it a stop. Its a GREAT daytime walk-around lens, but for fast motion in less than broad daylight.... might be hit/miss even with IS |
| 15:13.41 | Katty | she shakes. |
| 15:14.06 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:14.19 | beek | Katty: What's the price range you're looking at? I got a kick-around point-and-shoot Panasonic DMC-ZS3 that works a treat. |
| 15:14.46 | Katty | 400 or 500 |
| 15:14.52 | *** join/#asterisk curious101 (~curious10@110.55.168.87) |
| 15:15.39 | [TK]D-Fender | Katty: Katty tripod is good. Here's another great & easy idea : Get a nut & bolt (with an eye) with the same thread count as your camera mount, and a strong piece of string. Measure of just enough string to go between her foot standing on the nut and the bolt in the camera. |
| 15:15.46 | *** part/#asterisk doolittlework (~d@196.211.34.2) |
| 15:16.34 | Pegasus_RPG | I don't have any more time right now. Thank you very much ManxPower-work and [TK]D-Fender for all your help though!! |
| 15:16.43 | [TK]D-Fender | Katty: when pulled tight it will seriously reduce shaking. |
| 15:16.48 | beek | Katty: The ZS3 is in that range, has good low-light capability, can do HD video. Sweet little camera. I bought it for my trip to Phoenix and Astricon last year (didn't want to carry the DSLRs). |
| 15:16.53 | [TK]D-Fender | Katty: and fits in your hand as you walk around |
| 15:17.38 | [TK]D-Fender | Katty: I'll also vouch for the Panasonic Lumix series.... good stuff |
| 15:18.33 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 15:20.11 | Katty | low opitcal zoom |
| 15:20.16 | Katty | compared to that sx20 |
| 15:20.44 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 15:20.56 | beek | Leica glass... great stuff. |
| 15:21.12 | Katty | this is my mother we're tlaking about |
| 15:21.21 | [TK]D-Fender | beek: Absudly expensive... and not really worth it :0 |
| 15:21.32 | [TK]D-Fender | (for the dedicated stuff that is) |
| 15:21.43 | [TK]D-Fender | beek: Panasonic's OEM is another matter :) |
| 15:21.57 | [TK]D-Fender | Katty: Panasonic has super zoooms to match |
| 15:22.01 | [TK]D-Fender | Katty: shop around |
| 15:22.26 | Katty | the image sensor is a bit bigger. |
| 15:23.11 | Katty | [TK]D-Fender: you have a tripod recommendation? |
| 15:23.31 | *** part/#asterisk Pegasus_RPG (~chatzilla@p4FF903B3.dip.t-dialin.net) |
| 15:23.45 | [TK]D-Fender | Katty: Generally any will do.... Manfrotto if you're feel fancy |
| 15:24.05 | Katty | i have a dolica one |
| 15:24.10 | Katty | but it can support 20lbs |
| 15:24.32 | Katty | well probably more like 15 |
| 15:24.52 | [TK]D-Fender | Katty: .... a P&S won't weith 5 |
| 15:24.57 | [TK]D-Fender | weigh* |
| 15:25.33 | Katty | well yeah |
| 15:25.41 | Katty | but i have lenses on mine |
| 15:25.48 | *** join/#asterisk WinZ (~winz@82.146.61.218) |
| 15:25.52 | Katty | which is why i opted for a heavier load |
| 15:27.30 | sun28 | moin \o/ |
| 15:29.32 | WinZ | guys, is it possible to have Phone and Fax on the same line (parallel) --> ATA with T.38 --> Asterisk 1.6.2 and _receive_ faxes coming to Asterisk via SIP? |
| 15:29.33 | [TK]D-Fender | Katty: wonder if yours weight like my 70-210 F4 :) All metal & glass... |
| 15:30.52 | *** join/#asterisk radcliff (~radcliff@h-63-22.A259.priv.bahnhof.se) |
| 15:33.11 | Katty | [TK]D-Fender: never weighed it, but it's quite an armful |
| 15:33.52 | [TK]D-Fender | Katty: I was looking at a 70-210 F2.8 thatou would be.... well... harsh :0 |
| 15:35.20 | radcliff | Hi all, I have a problem I have been struggling with for a while, incoming calls does not work well, it works maybe 20% of time, the rest of the time asterisk console just tells me: " chan_sip.c:19961 handle_request_invite: Failed to authenticate device XXXXXXX <sip:XXXXXX@Y.Y.Y.Y;user=phone>", where XXXXXX is the external callerid and Y.Y.Y.Y is the ip of my SIP-provider, I am using insecure=invite and asterisk 1.6.2.5... |
| 15:35.29 | Katty | [TK]D-Fender: ha! |
| 15:36.17 | Katty | well i got her a cheap tripod |
| 15:36.20 | Katty | it's got hollow legs |
| 15:36.28 | Katty | but hopefully she won't be standing in wind |
| 15:36.48 | Katty | it's a shame they don't have snakeskin camera bags on newegg :< |
| 15:37.48 | WinZ | radcliff, try insecure=port,invite |
| 15:40.03 | radcliff | I have tried that aswell, no difference Iäm afraid... |
| 15:41.13 | *** join/#asterisk ChkDigit (~mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:43.01 | *** join/#asterisk V4mpire (~gary@82.118.111.252) |
| 15:43.23 | smooth_penguin | hey Katty :> |
| 15:44.32 | Katty | hello smoooooooooooth operator |
| 15:44.43 | giesen | Anyone done streaming music on hold with asterisk? Every solution I've tried, I run up against the "buffer" problem |
| 15:44.47 | Katty | [TK]D-Fender: any other accessories you think my mom might like? |
| 15:45.06 | Katty | [TK]D-Fender: i found a camcorder baggy that will fix that sx20, but i don't think she'll like it. i'll go find her something Pretty(tm) at target. |
| 15:45.39 | [TK]D-Fender | Katty: Thats why I'd look at a 10-12X compact super-zoom.. so it can fit in her purse |
| 15:46.01 | *** part/#asterisk [psy] (~psy0rz@lounge.datux.nl) |
| 15:46.11 | [TK]D-Fender | Katty: so the bolt & nut stabilizer for the purse, tripod for when she cares,a nd a camera she can just carry around all the time |
| 15:47.28 | radcliff | Does anyone know if there can be a problem with PostgreSQL realtime static and using a comma in "port,invite" for insecure in sip.conf??? I am running out of ideas :) |
| 15:50.01 | [TK]D-Fender | radcliff: does it work when using flat files? |
| 15:51.07 | radcliff | [TK]D-Fender: Havn't tried flat files in a couple of years... I was hoping I could find an answer to save me the trouble :) |
| 15:51.31 | [TK]D-Fender | radWell you just made a guess which i would like to have thought you would have tested. |
| 15:51.44 | [TK]D-Fender | radtill then you haven't shown us anything |
| 15:53.17 | ManxPower-work | not all that many people use realtime here. |
| 15:54.43 | leifmadsen | radcliff: I have a feeling multiple options to the same column name needs to be separated with a semi-colon |
| 15:55.05 | radcliff | leifmadsen: hmm, I'll try that! thanks! |
| 15:55.09 | leifmadsen | like: |
| 15:55.10 | leifmadsen | allow |
| 15:55.13 | leifmadsen | ulaw;alaw;gsm |
| 15:57.58 | radcliff | hmm, didn't work, still the same error... :( |
| 15:58.45 | radcliff | I'll try flat files and get back to you, thanks a lot! |
| 15:59.38 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:00.39 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:00.59 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:01.29 | ariel_ | Morning |
| 16:01.52 | Katty | hey ariel |
| 16:02.06 | ariel_ | hugs Katty |
| 16:06.13 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 16:06.53 | *** join/#asterisk cguerrero (~cuauhtemo@200.79.231.94) |
| 16:07.22 | *** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh) |
| 16:07.58 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 16:08.19 | ManxPower-work | radcliff, what verison of Asterisk? |
| 16:10.10 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:12.38 | *** join/#asterisk Corydon76-lap (~Corydon76@nat/digium/x-qlfuqrygeqjcqayb) |
| 16:12.38 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
| 16:16.46 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 16:16.46 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:17.03 | *** join/#asterisk voipmonk (~shido6@66.49.232.103) |
| 16:25.29 | Katty | just spent another 200 bucks on ferret Toys and Accessories. |
| 16:25.43 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:27.56 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 16:28.01 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 16:28.33 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:28.44 | beek | Katty: Those little critters can become expensive, can't they? |
| 16:29.19 | Katty | beek: they certainly are the highest maintenance pet i've ever had. |
| 16:29.45 | *** join/#asterisk Z_God (~julius@wlan225206.mobiel.utwente.nl) |
| 16:29.52 | wcselby | o/ |
| 16:30.06 | Katty | hi wcselby :> |
| 16:30.13 | wcselby | howdy |
| 16:30.13 | wcselby | :) |
| 16:30.29 | beek | Katty: Those photos you posted are darned cute though... |
| 16:30.33 | wcselby | although with this headache, i feel more like - "howdy, damnit" |
| 16:30.54 | EmleyMoor | "howdy!" |
| 16:32.22 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 16:34.54 | Katty | beek: yesh. |
| 16:34.57 | Katty | beek: i luvs them to bits. |
| 16:38.20 | Naikrovek | moiseur turtle bits? |
| 16:38.33 | Naikrovek | gratz canada on hockey gold, btw |
| 16:39.16 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 16:39.59 | WinZ | guys, when I'm trying to send a fax, my SIP provider keeps sending me "T.38 UDP: UDPTLPacket Seq=00000 t30ind: v21-preamble", but my Asterisk keeps answering "RTP PT=ITU-T G.711 PCMU" -- where can I dig? |
| 16:40.16 | wcselby | do you have Fax for Asterisk installed? |
| 16:40.20 | WinZ | no |
| 16:40.29 | wcselby | then how do you plan on sending t38? |
| 16:40.29 | WinZ | asterisk 1.6.2.2 |
| 16:40.41 | wcselby | you need FFA to send t38, or at least some sort of fax app |
| 16:40.50 | wcselby | not sure, does spandsp send t38? |
| 16:41.38 | WinZ | I send a fax from Zoiper, T.38. Have t38pt_udptl=yes in sip.conf in general and peers sections |
| 16:41.44 | wcselby | i guess the appropriate question to ask is, what fax utility are you using to achieve t.38? either FFA, or SpanDSP, or something.... |
| 16:42.07 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:42.17 | WinZ | just the default build of Asterisk. SpanDSP I guess |
| 16:42.42 | WinZ | it said that Asterisk supports t.38 out of the box |
| 16:43.05 | wcselby | t.38 passthrough |
| 16:43.12 | wcselby | what's "it" |
| 16:43.13 | wcselby | ? |
| 16:43.19 | wcselby | "it" said.... |
| 16:43.47 | wcselby | okay....start over |
| 16:43.48 | beek | Katty: You want high maintenance? Get a parrot. Dewey takes more room in my home than I do. |
| 16:43.57 | wcselby | (sorry, have a headache, things are difficult to concentrate on) |
| 16:44.22 | wcselby | WinZ - what are you sip.conf settings for your zoiper client, and also your itsp, and maybe you're general settings |
| 16:44.33 | Naikrovek | "dewey" great name |
| 16:44.42 | giesen | Is there any way to have asterisk spawn a new process for moh for each caller |
| 16:44.46 | giesen | rather than share a process? |
| 16:45.03 | beek | Naikrovek: Ever watch the sci-fi movie "Silent Running?" |
| 16:45.15 | Naikrovek | beek: no |
| 16:45.25 | Naikrovek | i'm not really into pokemon |
| 16:45.28 | Naikrovek | :) |
| 16:45.32 | *** join/#asterisk rickross (~rickross@supporter/active/rickross) |
| 16:45.51 | beek | There were three robots named 'Hewey, Dewey and Lewey". The parrot walks just like they did, so he got "Dewey". |
| 16:45.53 | WinZ | wcselby, ok, I need some time to dig it myself. I thought there was a quick answer, maybe some option in sip.conf. I'll come back if no luck. Thank you |
| 16:46.10 | beek | Naikrovek: What does "Silent Running" have to do with pokemon? |
| 16:46.13 | Naikrovek | beek: you ever see those old disney cartoons with huey, dewey, and louie? |
| 16:46.22 | beek | Yep... |
| 16:46.23 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 16:46.24 | giesen | DuckTales |
| 16:46.33 | Naikrovek | beek: nothing, i just say that sometimes when i don't know what someone is talking about |
| 16:46.59 | beek | Naikrovek: http://en.wikipedia.org/wiki/Silent_Running |
| 16:47.27 | Naikrovek | bruce dern |
| 16:47.47 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 16:49.22 | rickross | hi all, I have 10 Polycom phones in an office in NC that need to connect to an asterisk server in TX - until now we have used NAT from the NC router, and it has worked. We've just set up a VPN tunnel from NC to TX, though, and would like to force the phones to use that tunnel while leaving other computers and devices in NC on their normal NAT connections. Is this a reasonable thing to do? |
| 16:52.07 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 16:52.23 | Naikrovek | rickross: yes |
| 16:52.31 | Naikrovek | rickross: i do this to my phones in india (some of them) |
| 16:52.36 | Naikrovek | rickross: i'm in illinois |
| 16:52.42 | rickross | thanks, Naikrovek |
| 16:52.57 | Naikrovek | all you get from it is encrypted voice traffic, and slightly easier administration |
| 16:53.00 | rickross | do we need to make any specific config changes to the asterisk server in TX? |
| 16:53.18 | rickross | or do we just tell our phones in NC to use the tunnel as their gateway address? |
| 16:53.53 | Naikrovek | rickross: depends on your firewall, if the phones in NC can contact the * box it should work fine |
| 16:54.05 | Naikrovek | rickross: will just need to reconfigure the phone |
| 16:54.27 | rickross | well, the * box is bound to its public IP address |
| 16:54.48 | rickross | can * bind to multiple addresses? (or is this a non-issue) |
| 16:55.50 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 16:56.14 | rickross | I don't think * will be automatically listening for SIP on the tunnel IP address |
| 16:56.20 | Naikrovek | rickross: can the phones in NC contact the * server over the vpn? |
| 16:56.32 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 16:56.52 | Naikrovek | you can give the * box an internal IP address and it'll be fine |
| 16:57.05 | rickross | we don't know yet - we have just managed to get the VPN tunnel up, and local computers in NC can ping the remote side of the tunnel (on the * box) |
| 16:57.18 | Naikrovek | the * server is the other side of the VPN? |
| 16:59.10 | rickross | Naikrovek, we have other users who must connect from their own locations - it must remain accessible on its public IP |
| 16:59.17 | rickross | yes, * is on the other side |
| 16:59.30 | Naikrovek | it can be on a public IP and a private IP |
| 16:59.52 | rickross | ahh, so * can bind on multiple IPs? |
| 17:00.00 | Naikrovek | i know asterisk is ON the other side, is asterisk box THE OTHER VPN ENDPOINT |
| 17:00.05 | Naikrovek | rickross: of course |
| 17:00.20 | Naikrovek | it can listen on 0.0.0.0 (all interfaces) or just one or whatever |
| 17:00.35 | rickross | we cannot let it listen on 0.0.0.0 |
| 17:00.47 | Naikrovek | i'm just sayign it can |
| 17:00.51 | Naikrovek | i'm not telling you to |
| 17:00.54 | rickross | and I thought I remembered reading that you can only specify one listen address |
| 17:01.06 | Naikrovek | well mine listens on two |
| 17:01.13 | Naikrovek | one public, one private |
| 17:01.21 | rickross | nice - could you paste your sip.cfg line for that? |
| 17:01.43 | Naikrovek | i could put the whole thing on a single private interface and use my ASA to route incoming SIP packets to it, even |
| 17:02.25 | rickross | interesting thought |
| 17:02.42 | Naikrovek | rickross: i use trixbox (because I inhereted it) and it would not be wise for me to post any configs from it, because they woudl mean nothing to anyone, not even another trixbox user |
| 17:02.43 | rickross | but we have no ASA in this location, just a dedicated server at theplanet |
| 17:04.29 | Naikrovek | putting it behind my ASA is something i'm going to do, i do not like dual homing servers like that |
| 17:04.32 | Naikrovek | previous admin loved it |
| 17:06.25 | wcselby | wow |
| 17:06.36 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 17:06.36 | wcselby | somehow I just made it into my client's commercial |
| 17:06.42 | Naikrovek | nice |
| 17:08.04 | rickross | hmm, well I just set a phone into the address space of the tunnel, and I told it to use the tunnel's far end as its gateway |
| 17:08.13 | rickross | it did not register on * |
| 17:08.24 | rickross | and it could not contact the boot server |
| 17:08.28 | JT | what type of vpn are we talking about here? |
| 17:08.49 | rickross | it is OpenVPN (I believe it is version 2.1.x) |
| 17:09.30 | JT | ah |
| 17:09.30 | rickross | the OpenVPN server is on the same box as *, and the NC client is trying to access via a tunnel from a local dd-wrt router |
| 17:10.32 | rickross | computers on the NC side can ping the far end of the tunnel, but we are not sure whether the server end of the tunnel will rote to further destinations |
| 17:11.06 | wcselby | rickross - is your * server only listening on the public IP? |
| 17:11.42 | wcselby | rickross - either that or 0.0.0.0? |
| 17:11.57 | rickross | wcselby, it is on the public IP |
| 17:12.09 | wcselby | that's why you can't register to the openVPN ip |
| 17:12.30 | wcselby | rickross - you need to add udpbindaddr=vpn.ip.add.ress to the sip.conf file |
| 17:12.52 | rickross | ok, and it will allow more than one such line? |
| 17:13.14 | geneticx_wrk | hi. is the wildcard TDM400P a good option for an asterisk box? |
| 17:13.42 | Chainsaw | geneticx_wrk: Yes, if you want between 1 and 4 analog lines and have a PCI slot. (Note you also need a molex power connector) |
| 17:14.07 | wcselby | rickross - i've never tried, but I think it should work |
| 17:14.18 | wcselby | rickross - best way to find out is to just test it :) |
| 17:14.28 | rickross | I will do so now and let you know |
| 17:14.36 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:14.49 | rickross | I thought I remembered Russell saying it cannot have more than one, but I hope I am wrong |
| 17:15.00 | geneticx_wrk | Chainsaw: awesome. Have you ever used this card before? |
| 17:15.24 | Chainsaw | geneticx_wrk: Yes, I am using one in our fax server at the moment. |
| 17:15.38 | Chainsaw | geneticx_wrk: (With one FXO module and the other 3 module slots unpopulated) |
| 17:16.42 | *** join/#asterisk rocksfrow (~kyle@static-66-16-158-235.dsl.cavtel.net) |
| 17:16.58 | leifmadsen | rickross: you can either listen to all, or one address |
| 17:17.09 | wcselby | well there you go |
| 17:17.13 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 17:17.30 | geneticx_wrk | Chainsaw: cool. do you have a hardware echo cancellation module installed ? or you haven't had any issues with echo |
| 17:17.53 | Chainsaw | geneticx_wrk: I only use it for faxing so I have no need for echo cancellation. |
| 17:18.24 | geneticx_wrk | Chainsaw: I see. Ok, thanks for your help |
| 17:18.30 | wcselby | you could always listen to all, but then block traffic on port 5060 to the IP's you don't want it to use using IPTABLES or something |
| 17:18.39 | rickross | leifmadsen, so it is all of them or only one of them, but not two of them? |
| 17:18.50 | coppice | there isn't much point in hardware EC for just 4 channels |
| 17:19.01 | rickross | we'll put it on all then, to test the VPN thing |
| 17:19.16 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 17:19.25 | Chainsaw | geneticx_wrk: You're welcome. Are you in the UK by chance? |
| 17:19.36 | leifmadsen | rickross: correct, you can't just pick 2 of 3 interfaces to listen on for example |
| 17:20.13 | geneticx_wrk | Chainsaw: nope. Florida here =D |
| 17:20.23 | rocksfrow | if anybody can PLEASE Help me, i have a live customer service system down...it's a PRI and it was working perfectly...i previously did plenty of restarts and everything came back on fine..i just unplugged in the box and plugged it back in..now i'm getting congestion errors |
| 17:20.27 | Chainsaw | geneticx_wrk: Ah, good. Then there's no bugs to worry about :) |
| 17:20.31 | rocksfrow | i've tried resetting everything multiple times... |
| 17:20.46 | paulc | rocksfrow: Are the calls hitting your box? congestion inbound or outbound? |
| 17:20.47 | rocksfrow | is there something that i can clear or something?? |
| 17:20.52 | rocksfrow | congestino outbound |
| 17:20.53 | Kobaz | rocksfrow: call your provider |
| 17:20.57 | rocksfrow | inbound calls aren't hitting |
| 17:21.02 | rocksfrow | atleast i can't see from asterisk -rvvvvv |
| 17:21.10 | ManxPower-work | rocksfrow, My guess is that you upgraded the kernel and did not recompile the zaptel/dahdi kernel modules |
| 17:21.17 | rocksfrow | ManxPower-work, no. |
| 17:21.20 | ManxPower-work | rocksfrow, are you using Zaptel or DAHDI? |
| 17:21.24 | rocksfrow | i'musing dahdi |
| 17:21.26 | rocksfrow | i didnt make any upgrades |
| 17:21.32 | rocksfrow | everything was working perfectly |
| 17:21.36 | rocksfrow | until repoewring |
| 17:21.37 | geneticx_wrk | Chainsaw: bugs on the card or drivers for UK users? |
| 17:21.38 | rocksfrow | repowering |
| 17:21.41 | ManxPower-work | rocksfrow, what does "dahdi_cfg -vvv" give you. (pastebin the output) |
| 17:21.43 | rocksfrow | all channels look great |
| 17:21.48 | Chainsaw | geneticx_wrk: Buggy channel driver in the Asterisk core. |
| 17:21.51 | rocksfrow | ManxPower-work, no errors. |
| 17:21.58 | Chainsaw | geneticx_wrk: But it gets triggered by that card daily. |
| 17:21.59 | rocksfrow | ManxPower-work, dahdi show channels has te same output as before the restart |
| 17:22.06 | Kobaz | like i said.. call your provider if you are absolutly certain nothing changed and you are sure that the box itself is okay |
| 17:22.21 | ManxPower-work | rocksfrow, pastebin the output of "cat /proc/dahdi/1" |
| 17:22.25 | rocksfrow | Kobaz, thanks. |
| 17:22.32 | Kobaz | rocksfrow: either they'll tell you 'whoops, your line is down'... or they'll tell you... fix your server |
| 17:22.35 | rocksfrow | ManxPower-work, dont have time to waste time bro... |
| 17:22.38 | rocksfrow | calling the provider |
| 17:22.49 | rocksfrow | if they give heads up |
| 17:22.51 | geneticx_wrk | Chainsaw: and why does it only affect UK users? |
| 17:23.00 | rocksfrow | ill start debugging other things..but everything checks uout and looks IDENTICAL to before the restart |
| 17:23.02 | rocksfrow | no alarms |
| 17:23.04 | rocksfrow | al is good on my side |
| 17:23.07 | ManxPower-work | rocksfrow, I wish you the BEST of luck. Good thing you didn't run that cat command or I could have given you something specific to tell them. |
| 17:23.20 | rocksfrow | ManxPower-work, ill run it..sorry |
| 17:23.21 | rocksfrow | let me run it |
| 17:23.35 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 17:23.42 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 17:23.48 | rocksfrow | ManxPower-work, whooaaa |
| 17:23.49 | rocksfrow | they all show in use |
| 17:23.53 | rocksfrow | is thats what you were looking for? |
| 17:24.00 | Kobaz | no, that's normal |
| 17:24.03 | rocksfrow | oh.. |
| 17:24.09 | ManxPower-work | rocksfrow, that means "connected to Asterisk" not "active calls" |
| 17:24.17 | rocksfrow | oh okay |
| 17:24.18 | rocksfrow | then looks good |
| 17:24.23 | rocksfrow | was there anything elseu were looking for? |
| 17:24.25 | rocksfrow | timing slips: |
| 17:24.25 | rocksfrow | 2 |
| 17:24.29 | Chainsaw | geneticx_wrk: Because British Telecom runs automated line tests. |
| 17:24.31 | ManxPower-work | rocksfrow, paste us the line that has the B8ZS on it |
| 17:24.45 | Chainsaw | geneticx_wrk: A sequence of events occurs that is apparently unique to the UK market. |
| 17:24.45 | rocksfrow | Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) B8ZS/ESF |
| 17:24.45 | rocksfrow | Timing slips: 2 |
| 17:24.49 | ManxPower-work | rocksfrow, next go into the asterisk cli and type "pri debug span 1" |
| 17:25.11 | rocksfrow | okay debug enabled |
| 17:25.11 | ManxPower-work | rocksfrow, a timing slip would just cause a glitch in audio. |
| 17:25.12 | rocksfrow | now what? |
| 17:25.19 | rocksfrow | try calling out? |
| 17:25.24 | Kobaz | sure |
| 17:25.29 | ManxPower-work | rocksfrow, try to make some calls, put the cli output on pastebin.ca |
| 17:25.38 | rocksfrow | mn |
| 17:25.41 | rocksfrow | output looks identical |
| 17:25.44 | rocksfrow | congestion error from channel |
| 17:25.46 | Katty | ugah. |
| 17:25.52 | Katty | time to go to sam's club |
| 17:25.54 | ManxPower-work | rocksfrow, start pastebining or stop asking |
| 17:25.56 | geneticx_wrk | Chainsaw: humm..interesting. |
| 17:26.04 | Katty | someone come with me to carry stuff |
| 17:26.07 | rocksfrow | <PROTECTED> |
| 17:26.11 | rocksfrow | man..i have to call cavalier |
| 17:26.14 | rocksfrow | shit is DOWN right now |
| 17:26.14 | Kobaz | pastebin.ca |
| 17:26.19 | Kobaz | stop panicing |
| 17:26.20 | ManxPower-work | rocksfrow, that is not a pri debug of span 1 |
| 17:26.22 | Kobaz | and start pasting |
| 17:26.26 | rocksfrow | ok ok..lol |
| 17:26.59 | ManxPower-work | rocksfrow, you are going to be down for a couple of hours if this is a provider problem. Lets confirm it is a provider problem before you have them start screwing up your lines. |
| 17:27.04 | Chainsaw | geneticx_wrk: https://issues.asterisk.org/view.php?id=14163 |
| 17:27.19 | rocksfrow | http://pastebin.ca/1817475 |
| 17:27.37 | rocksfrow | i dont see how its provider..i mean it was literally working before i restarted the damned computer |
| 17:27.45 | rocksfrow | but damn.. |
| 17:27.57 | ManxPower-work | rocksfrow, call your provider. Say "I have no traffic on my D-channel. Fix it." |
| 17:28.00 | rocksfrow | i restarted the server a few times the other night to verify all is well after a repoweruping up |
| 17:28.05 | ManxPower-work | Chances are they will reset your port and all will be well. |
| 17:28.05 | rocksfrow | ManxPower-work, really? |
| 17:28.14 | rocksfrow | okay |
| 17:28.15 | rocksfrow | let me call |
| 17:28.19 | ManxPower-work | rocksfrow, a call should generate at least a page of debug stuff. |
| 17:28.27 | rocksfrow | okay |
| 17:29.08 | [TK]D-Fender | Where are the configs to match? |
| 17:29.24 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 17:29.49 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 17:30.37 | rickross | well, I set the sip bind address to 0.0.0.0 and see it listening when I do "netstat -l" on the server |
| 17:30.43 | geneticx_wrk | Chainsaw: you have PRI lines for your day-to-day voice calls? |
| 17:30.50 | rickross | but after a reboot, the NC phone, still doesn't connect to * |
| 17:31.18 | rickross | I may have something borked on the server end of the VPN (maybe it needs some briding or routing config) |
| 17:31.19 | [TK]D-Fender | rickross: And I see no SIP DEBUG from your attempts nor any configs |
| 17:31.45 | Chainsaw | geneticx_wrk: BRI, but I don't use Digium hardware in production anymore. |
| 17:32.29 | Chainsaw | geneticx_wrk: A bundle of three terminating on a Patton 4634. |
| 17:32.58 | Kobaz | Chainsaw: sangoma ftw |
| 17:33.37 | Chainsaw | Kobaz: I'm rather pleased with the separation now, I'll just have everything talking SIP to Asterisk. |
| 17:33.56 | Kobaz | that works too |
| 17:34.25 | Kobaz | i think we'll wind up moving to pure hardware tdm gateways quite soon, even though it's more expensive |
| 17:35.08 | Chainsaw | Kobaz: Yes, it's nicer to have failover configured on there so you can work around a downed Asterisk instance. |
| 17:35.24 | Kobaz | i have that going nicely, using two asterisk boxes |
| 17:35.48 | Kobaz | and one t1 card in each machine |
| 17:36.12 | Chainsaw | Ah, okay. That would do it. |
| 17:36.30 | *** join/#asterisk tkrn (~tkrn@216.196.213.246) |
| 17:36.46 | tkrn | need some help fellas, any body know where to get a SIP FAX Client Windows DriveR? |
| 17:36.56 | Kobaz | #voip ? |
| 17:37.28 | rocksfrow | ManxPower-work, with debug mode on..calls still are completed right? |
| 17:37.40 | Kobaz | rocksfrow: why wouldn't they be? |
| 17:37.44 | ManxPower-work | rocksfrow, yes. |
| 17:37.46 | rocksfrow | i reset all the hardware and now i'm getting different output |
| 17:37.48 | rocksfrow | not the same as i pasted |
| 17:37.52 | rocksfrow | but calls still not completing |
| 17:37.56 | Kobaz | keep pasting |
| 17:38.40 | ManxPower-work | rocksfrow, you should be seeing stuff like this with pri debug: http://pastebin.ca/1817495 |
| 17:40.09 | rocksfrow | http://pastebin.ca/index.php |
| 17:40.14 | rocksfrow | oops |
| 17:40.24 | rocksfrow | http://pastebin.ca/1817498 |
| 17:40.57 | geneticx_wrk | Chainsaw: nice. Does the Patton 4634 come with it's own drivers for asterisk? |
| 17:41.36 | Kobaz | geneticx_wrk: why would you need drivers? it's sip |
| 17:41.43 | rocksfrow | ManxPower-work, http://pastebin.ca/1817498 has more output |
| 17:42.02 | rocksfrow | its like its calling..butnot |
| 17:42.07 | rocksfrow | bc..i wont get any message |
| 17:42.14 | rocksfrow | it'll just sit quietly |
| 17:42.20 | geneticx_wrk | Kobaz: true. |
| 17:42.26 | rocksfrow | telco is gonna call me back in 15mins, they're testing the circuit |
| 17:42.36 | rocksfrow | ManxPower-work, Kobaz does that log say anything to you? |
| 17:42.56 | rocksfrow | there is some random at the end from something checking their vm |
| 17:43.04 | ManxPower-work | rocksfrow, notice how all that debug is SENT data, not received data. |
| 17:43.27 | rocksfrow | ManxPower-work, yes..thats interesting |
| 17:43.30 | ManxPower-work | rocksfrow, Ah. Then stop screwing with Asterisk or the telco may claim the problem is your equipment. |
| 17:43.37 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 17:43.44 | Kobaz | rocksfrow: either the line is fscked, or your card fried |
| 17:44.06 | rocksfrow | card fried?! oh no |
| 17:44.11 | ManxPower-work | did you tell them your D-channel is down? If not, chances are the wrong people are testng the line. |
| 17:44.12 | rocksfrow | wouldnt it be giving some sort of alarm? |
| 17:44.28 | Kobaz | rocksfrow: depends |
| 17:44.44 | bmoraca_work | is there anyway to disable the "!" command in the asterisk console? |
| 17:44.53 | rocksfrow | the card appears to be fine....it configures fine..and channels build out fine |
| 17:45.06 | rocksfrow | but..shit i do have a second card in the old server i could probably pop in |
| 17:45.13 | rocksfrow | that card is brand new though |
| 17:45.52 | ManxPower-work | rocksfrow, do nothing until the telco is calls you back |
| 17:46.03 | rocksfrow | ManxPower-work, right |
| 17:46.04 | Kobaz | rocksfrow: like it said... it all depends |
| 17:46.10 | rocksfrow | ManxPower-work, what's interesting is.. |
| 17:46.11 | Kobaz | rocksfrow: i've seen some weird failures |
| 17:46.14 | rocksfrow | before resetting the PRI that last time |
| 17:46.18 | Kobaz | rocksfrow: but most likly it's your provider |
| 17:46.19 | rocksfrow | i was just getting the congetsion error |
| 17:46.27 | rocksfrow | then after resetting the most recent time |
| 17:46.31 | rocksfrow | im getting this different output |
| 17:46.34 | rocksfrow | whats up with that? lol |
| 17:46.35 | rocksfrow | get me? |
| 17:46.41 | rocksfrow | does that confirm even more that its the telco? |
| 17:46.54 | Kobaz | it confirms that something is wrong somewhere |
| 17:46.59 | rocksfrow | lol |
| 17:47.06 | Kobaz | i'm serious |
| 17:47.10 | Kobaz | you wont know until the telco checks |
| 17:47.28 | geneticx_wrk | Chainsaw: you know of any VoIP Gateways like the 4630 but for PRI/? |
| 17:47.47 | Kobaz | if they say "everything looks fine here", then it's probably your box |
| 17:48.11 | Kobaz | but if you have a good provider... generally they call you when it's down |
| 17:49.07 | bmoraca_work | geneticx_wrk: check out the Adtran TA900 |
| 17:49.09 | dddh | hm, 729 is supported? |
| 17:49.21 | rocksfrow | Kobaz, ta900 is what i have i think |
| 17:49.24 | rocksfrow | oops |
| 17:49.32 | rocksfrow | Kobaz, there is an fxo port in the back of the adtran |
| 17:49.36 | rocksfrow | is that for testing?? |
| 17:49.58 | bmoraca_work | rocksfrow: it's for failover or general use as an FXO port |
| 17:50.36 | rocksfrow | bmoraca_work, do you have a ta900? |
| 17:50.44 | bmoraca_work | rocksfrow: i have several |
| 17:50.47 | rocksfrow | bmoraca_work, ever have any issues after powering off and back on? lol |
| 17:51.19 | bmoraca_work | rocksfrow: only if someone forgot to commit the changes |
| 17:51.45 | rocksfrow | ...? |
| 17:51.46 | idespinner | Ive had good luck with audiocodes pri gateways |
| 17:51.58 | idespinner | have about 15 in production, never an issue |
| 17:52.24 | idespinner | but there is a steep learning curve on configuring |
| 17:52.24 | bmoraca_work | rocksfrow: who set it up? telco or you? |
| 17:52.37 | bmoraca_work | idespinner: they're also more expensive than the Adtrans and less featureful |
| 17:52.46 | Kobaz | idespinner: haha, tell me about it... audiocodes have configuration insanity |
| 17:52.50 | rocksfrow | bmoraca_work, the asterisk box? |
| 17:53.01 | bmoraca_work | rocksfrow: no, the TA900 and what are you using it for? |
| 17:53.08 | geneticx_wrk | Chainsaw: thanks |
| 17:53.10 | rocksfrow | nah the telco set it up |
| 17:53.15 | *** join/#asterisk smooth_penguin (~smoove@59.95.10.201) |
| 17:53.22 | rocksfrow | but..i meant after powering down the asterisk box |
| 17:53.24 | Kobaz | are you still on hold? |
| 17:53.28 | geneticx_wrk | bmoraca_work: ok, I will thanks. |
| 17:53.29 | rocksfrow | Kobaz, hes calling me back |
| 17:53.34 | rocksfrow | he said 15m |
| 17:54.00 | bmoraca_work | rocksfrow: oh. unlikely to cause a problem, but if the TA900 also went down, it could be because they forgot to commit their changes |
| 17:54.29 | rocksfrow | bmoraca_work, that would suck...the ta900 did not go down originally though |
| 17:54.43 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:55.14 | rocksfrow | so frusrated, i reset this thing multiple times on friday to confirm everything would be cool after repowering |
| 17:55.36 | bmoraca_work | rocksfrow: is your PRI just not coming up on the asterisk box? |
| 17:55.47 | rocksfrow | bmoraca_work, no traffic from d channel |
| 17:55.55 | rocksfrow | card configures channels show.. |
| 17:56.06 | rocksfrow | everything looks identical to how it was before i restarted |
| 17:56.13 | rocksfrow | some sort of communication error.. |
| 17:56.18 | bmoraca_work | rocksfrow: is it giving you an alarm or is there just nothing on your d channel? |
| 17:56.30 | rocksfrow | the card isnt showing an alarm atleast |
| 17:56.38 | rocksfrow | all status lights on the telco hardware look normal |
| 17:57.46 | rocksfrow | inbound calls give busy signal |
| 17:58.04 | bmoraca_work | and "pri intense debug span 1" shows absolutely nothing? |
| 17:58.27 | bmoraca_work | shows no traffic what so ever? |
| 17:58.36 | rocksfrow | ManxPower-work already looked over the log |
| 17:58.48 | rocksfrow | said no traffic on d-channel |
| 17:58.57 | rocksfrow | i swear i think its the t1 box |
| 17:59.05 | rocksfrow | the line comes into the t1 box i guess, then into the pri box |
| 17:59.13 | *** join/#asterisk jilbert (~eXtra_Ric@77.30.210.176) |
| 17:59.15 | rocksfrow | i think that initial box is buggy |
| 17:59.21 | rocksfrow | i had a simliar issue previously and it just came back up |
| 17:59.47 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 17:59.51 | rocksfrow | what does timeout looking for release mean? |
| 18:00.06 | Kobaz | it's not getting data back |
| 18:00.07 | ManxPower-work | rocksfrow, it means "telco ignored us" |
| 18:00.31 | rocksfrow | okay, figured |
| 18:00.32 | rocksfrow | damn man |
| 18:00.35 | rocksfrow | isnt this some shit |
| 18:00.47 | *** part/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 18:00.55 | bmoraca_work | could be a bad or misconfigured adtran |
| 18:01.02 | ManxPower-work | rocksfrow, you must be new to telecom |
| 18:01.24 | rocksfrow | yah |
| 18:01.37 | TheDavidFactor | I've got a dual-homed * server (eth0 public ip, eth1 private ip) and when I try to call * from an internal phone * replies to all internal SIP messages on the external interface from the external ip, what am I missing? |
| 18:01.56 | ManxPower-work | stuff randomly breaks all the time. I swear sometimes it seems like the ILEC just randomly screws up lines. |
| 18:02.12 | bmoraca_work | TheDavidFactor: the common sense not to expose your PBX on a public interface |
| 18:02.14 | ManxPower-work | TheDavidFactor, remove bindip/bindaddr |
| 18:02.15 | *** join/#asterisk rickross (~rickross@supporter/active/rickross) |
| 18:03.31 | jaskew | bmoraca_work: That's just mean. Correct, but still mean ;) |
| 18:03.57 | TheDavidFactor | ManxPower-work: that did it, thanks! |
| 18:04.34 | TheDavidFactor | bmoraca_work: we've got stupid firewalls that don't like SIP or RTP so we have to expose our * servers until we replace our firewalls |
| 18:04.50 | TheDavidFactor | which is high on our priority list |
| 18:05.02 | ManxPower-work | TheDavidFactor, or you could turn off the sip ALG/nat on your firewalls |
| 18:05.02 | bmoraca_work | what kind of "stupid firewalls"? |
| 18:05.16 | ManxPower-work | you'll have to turn it off on most any firewall |
| 18:05.17 | rocksfrow | ManxPower-work, question.. |
| 18:05.26 | TheDavidFactor | some flavor of sonicwall |
| 18:05.28 | bmoraca_work | ManxPower-work: i've never had luck doing that on sonicwalls...but, yes, it works on most others |
| 18:05.31 | rocksfrow | so..i did 'amportal restart', and i get the same congestion error |
| 18:05.36 | rocksfrow | any clue why? |
| 18:05.45 | ManxPower-work | rocksfrow, I don't do GUIs |
| 18:05.54 | hardwire | Corydon76-lap: re my voicemail bug (whardier) it appears as though the arch word size is not being taken into effect in the 1.6.2.3-rc2 code for the thread stack size. It seems to be set at 8, statically. and I appear to be a near duplicate of bug #14932 |
| 18:06.32 | bmoraca_work | rocksfrow: congestion typically means the other side rejected your call, i believe. so it's consistent with what you're seeing. |
| 18:06.46 | hardwire | Corydon76-lap: oh nm.. seanbright replaced WORDSIZE with sizeof(void *) |
| 18:07.17 | rocksfrow | bmoraca_work, when i turn insense debugging on..cli constantly outputs the same message as idle |
| 18:07.22 | rocksfrow | unnumbered frame? |
| 18:07.24 | rocksfrow | is that normal |
| 18:07.49 | rocksfrow | like constantly outputting that same message |
| 18:07.51 | *** part/#asterisk k5tux (~RussW_K5T@tempest.bluecows.com) |
| 18:10.09 | rocksfrow | i wish i could reset the first box the telco line hits |
| 18:10.19 | rocksfrow | i guess the thing gets power through the telco line |
| 18:11.09 | *** join/#asterisk HorizonXP (~xitij@76-10-156-87.dsl.teksavvy.com) |
| 18:11.27 | *** join/#asterisk smooth_penguin (~smoove@59.95.10.141) |
| 18:11.31 | HorizonXP | hey, i'm having trouble getting my asterisk server to use my new DID |
| 18:11.33 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:11.35 | rocksfrow | hey guys, can you check out: http://pastebin.ca/1817550 |
| 18:11.44 | rocksfrow | that message is outputting at asterisk cli over and over |
| 18:11.57 | HorizonXP | i had it set up before with another DID which was working before |
| 18:12.25 | TheDavidFactor | ManxPower-work: I'm sorry I spoke too quickly earlier, it's still sending it out the wrong interface. Do you need to see my sip.conf? |
| 18:13.20 | ManxPower-work | TheDavidFactor, can you do a quitck "service iptables stop" (or whatever is used to turn that off on your Asterisk box. If that solves the problem, then your firewall is NATing the response packets for your internal clients. |
| 18:13.33 | ManxPower-work | (the firewall on the ASTERISK box, of course) |
| 18:14.48 | HorizonXP | it seems to be spawning calls from that DID on its own accord; it doesn't do it when i dial the number, but seems to do it repeatedly on its own |
| 18:16.09 | p3nguin_ | "it seems to" ??? What is spawning calls, and how do you spawn calls FROM a DID? |
| 18:16.44 | *** join/#asterisk rickross (~rickross@supporter/active/rickross) |
| 18:17.09 | TheDavidFactor | ManxPower-work: I don't think that's the problem, because iptables -t nat -L lists no rules; and I'm a little hesitant to mess with the firewall rules because I'm working on a remote server. |
| 18:17.16 | rocksfrow | can anybody explain what this message is? http://pastebin.ca/1817550 |
| 18:18.18 | TheDavidFactor | it might have helped to have mentioned this earlier, but I'm running * 1.6.2 trunk |
| 18:18.34 | leifmadsen | TheDavidFactor: you mean asterisk 1.6.2 branch |
| 18:18.42 | leifmadsen | TheDavidFactor: trunk is trunk |
| 18:18.51 | TheDavidFactor | yes, sorry there is only one trunk :-) |
| 18:18.55 | leifmadsen | :D |
| 18:19.19 | rocksfrow | bmoraca_work, ManxPower-work ...any clue what this is? <leifmadsen> TheDavidFactor: you mean asterisk 1.6.2 branch |
| 18:19.21 | rocksfrow | oops... |
| 18:19.23 | TheDavidFactor | I'm doing my best to sow FUD every where I go |
| 18:19.26 | rocksfrow | http://pastebin.ca/1817550 |
| 18:19.36 | leifmadsen | TheDavidFactor: well then good hustle :) |
| 18:19.55 | TheDavidFactor | btw, congrats dude! |
| 18:20.10 | bmoraca_work | rocksfrow: it's probably a keep-alive message your box is sending over the PRI. |
| 18:20.39 | rocksfrow | okay so unumbered isnt a bad thing |
| 18:20.57 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 18:21.26 | bmoraca_work | probably not...though my internal q.931 parser is on the fritz, so that may not be correct |
| 18:21.40 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 18:21.41 | pa | hi |
| 18:21.54 | pa | anyone has experience with linux nf_conntrack_sip and nf_nat_sip modules? |
| 18:22.06 | Qwell | pa: Don't use them |
| 18:22.08 | rocksfrow | alright the telco was supposed to call me back in 15min..not 40 |
| 18:22.08 | pa | i am not sure what options i have to use |
| 18:22.09 | rocksfrow | lol |
| 18:22.26 | pa | Qwell, but without i cant connect to my asterisk server from behind a nat |
| 18:22.26 | ManxPower-work | pa: use Asterisk's NAT support, not any other NAT support for SIP. |
| 18:22.32 | HorizonXP | p3nguin_: well, i can make outgoing calls from my SIP softphone that's connected to the asterisk server. spontaneously, it will receive calls. it's not doing it right now though. |
| 18:22.37 | ManxPower-work | ~answers |
| 18:22.38 | infobot | answers is probably Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 18:22.38 | Qwell | With it, things will be very broken. |
| 18:22.46 | ManxPower-work | pa:read the NAT stuff CAREFULLY |
| 18:22.49 | HorizonXP | so let's ignore that. when I try to call my DID, i don't see it on the asterisk console, nor the SIP softphone |
| 18:22.50 | Qwell | pa: You just need to learn to setup NAT config in Asterisk |
| 18:23.02 | pa | ManxPower-work, i tried to put nat=yes in my sip.conf, and now i can login into my sip server,but i cant hear anything |
| 18:23.20 | ManxPower-work | pa: stop being an idiot and read the docs we sent you to. |
| 18:23.30 | p3nguin_ | horizonxp: Did you register to the ITSP correctly? Did you run a SIP debug to see if the call is hitting Asterisk at all? |
| 18:23.33 | pa | ok |
| 18:23.35 | pa | let me see |
| 18:23.36 | HorizonXP | p3nguin_: oh wait, correction: i can see it on the console now |
| 18:23.44 | ManxPower-work | nat=yes is for REMOTE CLIENTS behind NAT. |
| 18:23.51 | pa | yes |
| 18:23.54 | pa | thats my case |
| 18:24.04 | ManxPower-work | pa:I thought your ASTERISK server was behind NAT. |
| 18:24.19 | pa | no, my asterisk is on public ip |
| 18:24.23 | pa | client is behind a nat |
| 18:24.39 | p3nguin_ | You don't need to configure NAT stuff on the client network. |
| 18:24.55 | ManxPower-work | pa: then it should work just fine as long as you do not enable SIP NAT on the NAT router and do not enable SIP NAT on the phones. |
| 18:25.00 | HorizonXP | p3nguin_: yeah, sip debug shows it. ok, so my exntesions.conf is right, just need to forward it to my sip phone |
| 18:25.17 | ManxPower-work | pa: do you have a firewall on the Asterisk server? |
| 18:25.21 | p3nguin_ | horizonxp: Just because sip debug shows it does not mean your extensions are right. |
| 18:25.24 | pa | well, i use x-lite as client, and did not configure anything |
| 18:25.37 | pa | ManxPower-work, yes, i do |
| 18:25.41 | pa | and i opened the 5060 |
| 18:25.52 | ManxPower-work | pa: what about the audio ports? |
| 18:25.55 | p3nguin_ | pa: And the RTP range, too? |
| 18:26.06 | pa | mmh.. not sure. |
| 18:26.09 | bmoraca_work | psssh...i don't need no stinkin RTP ports |
| 18:26.20 | Corydon76-lap | "What's RTP for?" |
| 18:26.24 | ManxPower-work | the RTP ports are listed in /etc/asterisk/rtp.conf. You have to open those up to. |
| 18:26.27 | pa | i mean, it was working just fine with clients not behind a nat |
| 18:26.35 | HorizonXP | p3nguin_: no, you're right it doesn't. but the fact that i was now able to forward it to my SIP phone to have it answered, does. :-) |
| 18:26.36 | pa | ok, i check |
| 18:26.57 | p3nguin_ | horizonxp: This means you have completed the project? |
| 18:29.00 | pa | can i narrow the RTP range? or should i keep it so broad? |
| 18:29.28 | jameswf | Qwell, any reason a clean install of *NOW with yum update + *1.6 conversion would not have dahdi_tool installed? |
| 18:29.35 | p3nguin_ | pa: You need 2 ports per call. |
| 18:29.46 | pa | oh ok |
| 18:29.52 | pa | and these are udp ports, right? |
| 18:29.57 | pa | rtp i mean |
| 18:29.58 | p3nguin_ | pa: So if you aren't doing 5000 calls at the same time, you can narrow it. |
| 18:30.08 | p3nguin_ | Yes, they are UDP. |
| 18:30.26 | rocksfrow | ManxPower-work, still around? |
| 18:30.28 | Corydon76-lap | You need 4 ports per call, if you're doing video |
| 18:31.00 | Qwell | jameswf: mmm, nope |
| 18:31.16 | p3nguin_ | corydon76-lap: one each way for audio, one each way for video? or two each way for video itself? |
| 18:31.23 | jameswf | :( i have no dahdi_tool I find it mildly annoying... |
| 18:31.57 | pa | however was it normal that everything was working without RDP ports open, when my client wasnt behind a nat? |
| 18:32.44 | p3nguin_ | pa: Client networks do not need any ports forwarded. If your server has a firewall hiding the ports, they have to be uncovered. |
| 18:33.34 | pa | p3nguin_, yes, i mean, my firewall blocked those ports on the server, but everything worked fine when the client was not behind a nat |
| 18:33.45 | pa | now my client is behind a nat, and i cant hear anything anymore |
| 18:33.46 | Corydon76-lap | one each way for video |
| 18:34.04 | *** join/#asterisk blaines (~blaines@67.130.168.2) |
| 18:34.15 | Corydon76-lap | One port each way for each separate data stream |
| 18:34.15 | p3nguin_ | pa: Seems strange to be able to cover the ports and connections would still be possible. |
| 18:34.31 | ManxPower-work | pa: That is so unlikely I doubt anyone here believes you. |
| 18:34.34 | pa | now it works fine |
| 18:34.40 | pa | well |
| 18:34.43 | Corydon76-lap | You could have audio, video, and text, and that would require 6 ports per call |
| 18:34.50 | pa | now i opened the rdp range, and it works fine |
| 18:34.56 | ManxPower-work | like maybe you modified your firewall at one point. |
| 18:35.01 | rocksfrow | ManxPower-work, do you think it would be worth trying to use the second span on the card instead of the first? |
| 18:35.05 | pa | well i did it now |
| 18:35.12 | ManxPower-work | or maybe the firewall sofware was updated at one pooint. |
| 18:35.21 | ManxPower-work | rocksfrow, I gave you my diagnosis. Take it or leave it. |
| 18:35.36 | p3nguin_ | Is there any chance that it could have been allowing the reciprocating port as a RELATED connection? |
| 18:35.41 | rocksfrow | ManxPower-work, your diagnosis was to call the telco, i did..he just returned my call saying everything looks fine on their side! |
| 18:35.47 | HorizonXP | p3nguin_: this means I've solved my problem, yes :) |
| 18:35.52 | *** join/#asterisk lesouvage (~lesouvage@82.73.69.76) |
| 18:35.53 | ManxPower-work | rocksfrow, it sucks to be you. |
| 18:35.53 | rocksfrow | hes looking into something else and calling me back in 5 he said.. |
| 18:35.59 | rocksfrow | ManxPower-work, damn man are you serious? |
| 18:36.01 | p3nguin_ | horizonxp: Wonderful! |
| 18:36.12 | *** join/#asterisk smooth_penguin (~smoove@59.95.18.170) |
| 18:36.14 | *** join/#asterisk Arcu (~tilde@173.49.38.18) |
| 18:36.22 | pa | maybe it's xlite who does something strange? |
| 18:36.32 | ManxPower-work | rocksfrow, say "I need a tech with a T-Byrd dispatched". Make sure the tech can make and receive calls using the T-Byrd. |
| 18:36.43 | pa | or my version of asterisk which is bit-something? |
| 18:36.50 | rocksfrow | ManxPower-work, okay. |
| 18:36.52 | pa | (for zapata) |
| 18:37.18 | pa | i am 200% sure that those ports were closed, and that of course i could call |
| 18:37.25 | jameswf | Qwell, may be a bug I just did rpm -qpl dahdi-tools-2.2.1-1_centos5.i386.rpm dahdi_tool is not there |
| 18:37.40 | Qwell | not in the RPM? |
| 18:37.45 | jameswf | nope |
| 18:37.56 | Qwell | lets see |
| 18:38.43 | rickross | ok, we now have 10 local phones on the client side of the VPN, each talking to the * sip server on the other end, and we definitely had more than twice as many conversations working simultaneously as we had been able to conduct before the VPN |
| 18:39.23 | Qwell | jameswf: It's possible that package was built without newt-dev installed |
| 18:39.46 | Qwell | ah hah |
| 18:40.26 | pa | well, however thanks : ) |
| 18:40.30 | Qwell | jameswf: okay, give me a bit. I think I see what's wrong |
| 18:41.02 | jameswf | Qwell, np not in a hurry if I was I would just build it :) |
| 18:42.41 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 18:44.11 | Kobaz | heh |
| 18:45.41 | rickross | We can now do 10 simultaneous calls across the VPN, but we began to degrade with only 3 simultaneous calls before the VPN - does this imply that Time-Warner cable (or someone else in the middle) is capping voip traffic? |
| 18:46.19 | p3nguin_ | It certainly seems illogical. |
| 18:46.27 | Katty | hi. |
| 18:46.27 | p3nguin_ | the scenario, that is. |
| 18:46.31 | rickross | why illogical? |
| 18:47.07 | p3nguin_ | It seems like bandwidth is bandwidth, so cramming calls over a VPN would actually use more of it than calls not going across a VPN. |
| 18:47.14 | *** join/#asterisk s14ck (~s14ck@190.142.78.144) |
| 18:47.21 | Katty | has anyone been asking about connecting asterisk to a toshiba system |
| 18:47.25 | *** join/#asterisk smooth_penguin (~smoove@59.95.18.170) |
| 18:47.33 | Katty | HELLO SMOOTH OPERATOR |
| 18:47.36 | rickross | we had observed that we could use almost our full 2 mbps upstream for non-voip traffic, but we started to degrade around 300 kbps of voip traffic |
| 18:47.37 | Qwell | throws a box of sudafed at Katty |
| 18:47.49 | Katty | :> |
| 18:47.59 | Katty | puts it in the Emerency Supply Stash |
| 18:48.10 | rickross | we started to suspect that maybe some kind of bandwidth limiting or traffic shaping was in effect |
| 18:48.13 | Qwell | wonders if he closed his lens yesterday |
| 18:48.17 | Katty | Qwell: are you feeling better todays? |
| 18:48.21 | Qwell | Katty: nope |
| 18:48.21 | *** join/#asterisk yamahataxx (maxxxim@host-static-109-185-146-130.moldtelecom.md) |
| 18:48.25 | Katty | Qwell: bummer :< |
| 18:48.28 | rickross | so we set up a vpn and routed all our voip traffic to the * server across the vpn |
| 18:48.29 | yamahataxx | there is any sip client for asterisk that can make calls to a specific list of phones, in order to play some audio file and to hangup the line? |
| 18:48.30 | Katty | Qwell: i will send BB to give you kisses. |
| 18:48.41 | Qwell | O.o |
| 18:48.50 | p3nguin_ | rickross: That's crazy. Try usign iperf in udp mode on voip ports, then again on non-voip ports. |
| 18:48.55 | Katty | Qwell: ferret kisses make everything better. |
| 18:49.28 | rickross | now we seem to be able to use all of our upstream bandwidth for voip calls with nearly no degradation until we start peaking near the max |
| 18:49.32 | ariel_ | yamahataxx: none needed you can setup your own call files |
| 18:49.51 | Naikrovek | rickross: that's how it's supposed to be - no degredation until you reach max |
| 18:49.52 | rickross | p3nguin_: I am unfamiliar with iperf |
| 18:50.09 | p3nguin_ | rickross: It's easy to figure out. I trust that you can do it. |
| 18:50.23 | p3nguin_ | rickross: If you can set up VoIP over VPN, you can surely use iperf. |
| 18:50.25 | Katty | you can doooo eeet |
| 18:50.28 | rickross | Naikrovek: we always thought so, but apparently someone between us and our * server was capping or shaping |
| 18:50.41 | Naikrovek | rickross: some ISPs don't like voip |
| 18:50.47 | Naikrovek | even some that don't sell phone service |
| 18:50.55 | rickross | p3nguin_: I will need to read about iperf |
| 18:51.07 | Naikrovek | it could also be a QoS thing; their links may be busy with other stuff, and voip may be low on priority list |
| 18:51.10 | p3nguin_ | naikrovek: That sounds like DoS to me. And that's against the law. |
| 18:51.12 | rickross | I would love to have a clear, straightforward demonstration that this is occurring |
| 18:51.23 | yamahataxx | ariel_> where can i read about this? |
| 18:51.31 | Naikrovek | p3nguin_: he should call his ISP and find out what's up |
| 18:51.42 | rickross | Naikrovek, we will |
| 18:51.46 | Naikrovek | okay |
| 18:51.48 | Naikrovek | let us know what they say |
| 18:51.58 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 18:52.03 | Naikrovek | there are tools out there you can use to determine if your voip is being throttled |
| 18:52.10 | rickross | our understanding is that Time-Warner claims NOT to be doing anything like DPI or traffic shaping |
| 18:52.11 | Naikrovek | at least i think there are |
| 18:52.15 | Naikrovek | there should be if there isn't |
| 18:52.31 | Naikrovek | rickross: perhaps the place where your server is colocated is doing the shaping |
| 18:52.36 | rickross | but the apparent evidence suggests that someone is! |
| 18:52.51 | rickross | Naikrovek - good point |
| 18:53.05 | rickross | we don't know where in the middle it happens, but we are certain that it does |
| 18:53.19 | p3nguin_ | VPN traffic could be high in priority? |
| 18:53.23 | rickross | I doubt theplanet.com does any shaping |
| 18:53.28 | p3nguin_ | But VoIP is lower? |
| 18:53.34 | Naikrovek | p3nguin_: good point |
| 18:53.35 | rickross | they have lots of people running voip servers |
| 18:53.54 | p3nguin_ | Voice and video should always be higher than anything else, right? |
| 18:53.58 | Naikrovek | p3nguin_: now he's doing voip over vpn so it should all appear to be vpn to any ISP |
| 18:54.42 | Naikrovek | that may be it |
| 18:54.51 | Naikrovek | they've simply prioritized vpn over voip |
| 18:54.53 | rickross | Naikrovek: that is our understanding. Unless there's some way to infer from 20 ms rtp packet timing, they shouldn't be able to identify what we are sending across the vpn. |
| 18:54.55 | p3nguin_ | I would think voice/video should be high, and interactive traffic should be slightly lower. Bulk traffic would be lower than everything else. |
| 18:55.44 | rickross | Naikrovek: ftp also moves at full speed on the upstream channel. The only capping we have observed is our SIP calling |
| 18:56.01 | rocksfrow | ManxPower-work, got a minute? |
| 18:56.04 | Naikrovek | rickross: what voice codec are you using |
| 18:56.12 | rocksfrow | telco called me back..he said he can see the PRI status going up, then down..up - down |
| 18:56.14 | rocksfrow | over and over... |
| 18:56.21 | rickross | in this case we are using G.722 to our * |
| 18:56.21 | p3nguin_ | flapping? |
| 18:56.39 | rickross | I believe it uses about 90 kbps per call |
| 18:56.46 | Naikrovek | rickross: g722 okay. i /think/ that uses less bw than g711 right? |
| 18:56.47 | TheDavidFactor | ManxPower-work: ok, I feel stupid :-) I didn't have the routes configured correctly. should have tried ping a lot sooner :-S |
| 18:57.17 | rickross | I don't know the g711 overhead, but I think g722 is slightly higher |
| 18:57.38 | rickross | but we were crapping out at a small fraction of our actual available upstream bandwidth |
| 18:57.47 | ManxPower-work | rocksfrow, you may have to try rebuilding dahdi, libpri, and Asterisk. |
| 18:57.50 | p3nguin_ | rickross: http://www.ciscosystems.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml |
| 18:57.55 | rickross | degradation started at around 300 kbps of voip upstream |
| 18:58.05 | rocksfrow | ManxPower-work, rebuilding!? |
| 18:58.10 | rocksfrow | ie reinstalling?? |
| 18:58.20 | ManxPower-work | rocksfrow, yes. Should take you all of 15 mins. |
| 18:58.26 | rickross | p3nguin_: thx |
| 18:58.36 | Naikrovek | rickross: you'll just have to find some sip testing tools and measure what's going on |
| 18:58.40 | Katty | can i borrow someone to test a pop3/smtp account |
| 18:58.41 | ManxPower-work | rickross, I have seen version mismatches causing similar problems. |
| 18:58.43 | rocksfrow | ManxPower-work, do you think i should try using the second span on my card? |
| 18:58.45 | rocksfrow | worth a shot, or no? |
| 18:58.57 | rickross | 87.2 kbps per call qualifies as "approximately 90 kbps" to me |
| 18:59.03 | ManxPower-work | rocksfrow, I have no opinion on the matter or I would have responded. |
| 18:59.10 | rocksfrow | ManxPower-work, right.. |
| 18:59.14 | rocksfrow | damn man this is fucking rough |
| 18:59.15 | rocksfrow | wtf. |
| 19:00.30 | rickross | ManxPower-work: I don't know what versions you mean? These are the same phones talking to the same * server. The difference is whether or not we route over the vpn tunnel. |
| 19:01.04 | rickross | ManxPower: is ther something else you think could be in play in this context? |
| 19:01.18 | Katty | any volunteers to help me test this pop3/smtp account |
| 19:01.29 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:01.36 | Naikrovek | Katty: busy or i'd be happy to |
| 19:01.43 | Naikrovek | working on three laptops atm |
| 19:01.46 | Katty | k |
| 19:01.50 | p3nguin_ | katty: I guess I will. |
| 19:01.53 | Katty | you don't have to |
| 19:01.59 | Katty | dave might be able to help |
| 19:02.07 | rickross | I'd love to know how to use wireshark or some traffic generator to simulate this. It would allow us to eliminate a lot of variables and focus on what the ISP is doing. |
| 19:02.10 | p3nguin_ | Either way. |
| 19:02.23 | Katty | p3nguin_: -> |
| 19:02.24 | p3nguin_ | rickross: iperf |
| 19:02.41 | rickross | p3nguin_: I will read about it after lunch - thx |
| 19:03.39 | Naikrovek | rickross: sipp to create sip connections and calls, really |
| 19:03.39 | Naikrovek | SIPp |
| 19:03.40 | *** join/#asterisk maszlo (~reckenrod@65.223.240.146) |
| 19:03.40 | rickross | Naikrovek: I will look into that one, too |
| 19:04.19 | rickross | thx, guys - bbiaw |
| 19:04.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:06.18 | maszlo | I have recently have been smacked with everyones phone getting the reorder tone when trying to make a call. checking the logs I got app_dial.c: Unable to forward voice or dtmf can anyone point me in the direction of where to start? |
| 19:06.29 | maszlo | the system was functional for over a year |
| 19:06.51 | seanbright | hardwire: you there? |
| 19:06.59 | Naikrovek | maszlo: what changed recently |
| 19:07.24 | Naikrovek | maszlo: firewall? recent reboot? |
| 19:07.53 | wcselby | Katty - need help with that email still? |
| 19:07.58 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 19:08.07 | maszlo | nothing, phone calls functions up until about an hour ago |
| 19:08.27 | Katty | wcselby: sure |
| 19:08.29 | Katty | wcselby: -> |
| 19:08.41 | maszlo | I did attempt a reboot after not seeing anything off from dmesg |
| 19:09.05 | maszlo | am wondering if should call the provider or where to look here |
| 19:09.10 | wcselby | maszlo - what's your setup? |
| 19:10.42 | rocksfrow | is there anywhere I can configure the PRI as n1-2? |
| 19:11.05 | rocksfrow | i dont understand what he meant by that |
| 19:11.17 | rocksfrow | but the telco told me to confirm i'm configured as N1-2 |
| 19:11.51 | hardwire | seanbright: indeed |
| 19:11.51 | maszlo | we are using a rhino pri into the system, verizon service |
| 19:12.08 | seanbright | hardwire: uploaded a patch to your issue |
| 19:12.23 | mazpe | does it make a difference in performance or anything to use exten => _1NXXNXXXX, 1, .... exten => _1NXXNXXXX, 2, .... vs exten => _1NXXNXXXX, 1, ..... exten => _1NXXNXXXX, n, ..... |
| 19:12.46 | mazpe | i other words using 'n' vs actually numerating each extension |
| 19:13.30 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 19:13.35 | [TK]D-Fender | mazpe: No. its gets parsed out on load |
| 19:14.02 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 19:14.09 | mazpe | n just takes it in the order listed i assume. |
| 19:16.58 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.35 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 19:19.00 | rocksfrow | the telco called me back and told me he can see the d-channel going up and down over and over... |
| 19:19.04 | rocksfrow | he says thats my fault |
| 19:19.11 | rocksfrow | why can't i see this happening within my logs then? |
| 19:19.12 | p3nguin_ | lol |
| 19:19.24 | rocksfrow | is there something i'm missing here? |
| 19:19.39 | p3nguin_ | You should have retorted with, "No you!" |
| 19:19.41 | rocksfrow | techs supposed to be coming out..just afraid hes giong to get here and say everything is OK |
| 19:19.48 | rocksfrow | p3nguin_, lol |
| 19:20.17 | rocksfrow | what could possibly be causing the d-channel to keep dropping like that? |
| 19:20.29 | rocksfrow | could a bad t1 card do that?..wouldn't there be some sort of log? |
| 19:21.43 | rocksfrow | i'm so lost... |
| 19:21.54 | wcselby | rocksfrow - i know this may not help you, but I recently had a client that had a bad cable plugged between their t1 demarc and their t1 card in their asterisk server |
| 19:22.06 | wcselby | if you touched it slightly, it would go into an alarm state |
| 19:22.17 | wcselby | swapped out with a new cable, and it's been peachy ever since |
| 19:22.20 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 19:22.23 | rocksfrow | damn.. |
| 19:22.26 | rocksfrow | well, i'm not showing any alarms |
| 19:22.33 | rocksfrow | green status on both the pri, and my card |
| 19:23.43 | outtolunc | rocksfrow: sounds like the tech that told one of my clients his new pbx/t1 card was the issue and wanted to charge them $400 to do a site visit. I jumped on and seen the issue, and 'magically' they were able to find/and fix the issue. |
| 19:24.47 | ecrane | rocksfrow: Despite the reliability of telco equipment, sometimes a good 'old reboot can fix a problem. That includes the telco side... if it's ok to take the circuit down ask them to do a 'remove and restore'. Or see if you can loop on the circuit from your end and have them do testing.... |
| 19:25.12 | ecrane | rocksfrow: They probably have test sets that are way more expensive then yours. (No offense...) |
| 19:25.26 | rocksfrow | ecrane, he remotely ran a test |
| 19:25.31 | ecrane | rocksfrow: remove/restore would be on the d-channel stuff. |
| 19:25.39 | ecrane | ran a test on the circuit? |
| 19:25.48 | rocksfrow | i guess |
| 19:25.51 | rocksfrow | okay so remove/restore? |
| 19:26.04 | rocksfrow | sounds like this telco guy is just as much of an idiot as i am |
| 19:26.17 | rocksfrow | he said he was running a test yes |
| 19:26.24 | rocksfrow | and he said he can connect to the adtran fine |
| 19:26.33 | rocksfrow | and..can see the pri state going up and back down |
| 19:26.34 | rocksfrow | over and over |
| 19:26.37 | ecrane | couldn't hurt.. as long as taking down the traffic is ok. if he did LOOP it on your side and test then yeah, like wcselby said, could be a patch cable from the adtran. |
| 19:26.54 | rocksfrow | wouldn't a bad patch cable give an alarm |
| 19:27.00 | rocksfrow | im getting all green status lights |
| 19:27.03 | *** part/#asterisk pfn (pfnguyen@socal.hanhuy.com) |
| 19:27.05 | ecrane | " I can connect to the adtran " is not the same as loop and running bit pattern tests (eg. BERT tests) |
| 19:27.09 | rocksfrow | the lights will go from green to red when i unlpug, but then back to green |
| 19:27.18 | rocksfrow | ecrane, right.. |
| 19:28.22 | rocksfrow | outtolunc, ecrane just frustrating bc this is after physically moving the server |
| 19:28.30 | rocksfrow | wish i never touched the fucking thing lol |
| 19:28.40 | rocksfrow | everything looks identical as before the reboot.. |
| 19:28.41 | ecrane | I'm just saying, from a telco perspective, they are often hesitant to type in commands to disable then re-enable the d-channel on their switch, but I have seen that clear problems on occasion. No guarantee ;<. |
| 19:29.20 | rocksfrow | outtolunc, the signalling? |
| 19:29.29 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 19:29.45 | rocksfrow | outtolunc, this all was working perfectly before unlpugging it and plugging it back in...doesnt that confirm that configuration is fine? |
| 19:29.52 | rubberneck | Anyone have skill with polycom phones? |
| 19:29.52 | rocksfrow | if nothing was changed after it working for 3 days straight? |
| 19:30.45 | rubberneck | When i get calls on my polycom phones the caller ID shows in this format "sip:<extension>@Ipaddress" is there a way to change this in the xml file somewhere? |
| 19:31.16 | rocksfrow | ecrane, when the tech gets out here...he shouldn't be able to make calls out right.. |
| 19:31.22 | outtolunc | rocksfrow: if you are saying the server moved, nothing else changed (same site, same server, same card, same software, same circuit...) then did the 'distance' between the server and the niu change (increase over 100') |
| 19:31.28 | rocksfrow | i'm just afraid he's going to be able to call out yet there still be an issue on their side? |
| 19:31.44 | rocksfrow | outtolunc, nope, like 2 feet further away lol |
| 19:32.00 | rocksfrow | literally nothing has changed... |
| 19:32.01 | Naikrovek | rubberneck: are you using asterisk or just phone-to-phone calls |
| 19:32.11 | rubberneck | Naikrovek: asterisk |
| 19:32.18 | outtolunc | rocksfrow: then i agree, change the cable |
| 19:32.19 | Naikrovek | then there is a configuration issue within asterisk |
| 19:32.23 | Naikrovek | but i dunno what it is |
| 19:32.32 | rocksfrow | hrm..i can try making my own |
| 19:32.39 | rocksfrow | the cable is special |
| 19:32.56 | rocksfrow | outtolunc, so you think its possible that the cable is causing the issues but..still displaying a green status? |
| 19:33.03 | rocksfrow | i'm open for anything..let me go get the peices |
| 19:33.50 | outtolunc | it is possible, i add an issue once where the cable was causing the NIU which had an issue where it would do a local loop on cpe signal loss |
| 19:34.07 | ruben23 | hi, any problem, all my asterisk CLI display only white text, no color purple or blue, on agi scripts, i cant identify.. |
| 19:34.11 | outtolunc | so yeah, a cable 'can' cause the d channel to bounce |
| 19:34.26 | outtolunc | without causing an error on the card |
| 19:35.29 | outtolunc | iirc, it is only the RX side that shows continuity on the card |
| 19:35.53 | rocksfrow | okay im going to try to duplicate this cable |
| 19:36.38 | rocksfrow | hey question |
| 19:36.39 | rocksfrow | with this cable |
| 19:36.43 | rocksfrow | does it matter which end is plugged in? |
| 19:36.49 | rocksfrow | since they are crossed i mean |
| 19:37.01 | rocksfrow | dont think so..right? |
| 19:37.04 | outtolunc | no |
| 19:37.09 | rocksfrow | k |
| 19:37.17 | outtolunc | as long as 'both' ends are wired the same as original |
| 19:37.53 | outtolunc | and that 'that' is the original working one.. and someone didn't go 'oh lookie.. a new cable.. lets use this one' |
| 19:39.54 | ManxPower-work | Remember a crossover T-1 cable is NOT the same as a crossover Ethernet. |
| 19:42.11 | rocksfrow | no, using identical cable |
| 19:42.18 | *** part/#asterisk maszlo (~reckenrod@65.223.240.146) |
| 19:42.20 | rocksfrow | i really doubt the cables bad..but i want to rule everything out |
| 19:42.25 | rocksfrow | would be a great suprise |
| 19:42.30 | ruben23 | hi, any problem, all my asterisk CLI display only white text, no color purple or blue, on agi scripts, i cant identify.. |
| 19:42.32 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 19:44.00 | bmoraca_work | ruben23: it's broken...you have to throw it away now |
| 19:44.24 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.12.93) |
| 19:44.52 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 19:45.44 | wcselby | rubberneck - i think there's a sitting in the sip.cfg file that controls that |
| 19:45.49 | wcselby | a setting* |
| 19:47.08 | bmoraca_work | imperial measurements are so funny... 1.5 L = 1 qt, 1 pt and 2.7 fl oz |
| 19:48.09 | idespinner | how do you control terminal colors? |
| 19:48.22 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 19:48.28 | idespinner | ive got an AA50 that never resizes when using VI. its pretty annoying |
| 19:48.36 | wcselby | ruben23 - how are you accessing the cli? |
| 19:48.47 | wcselby | ruben23 - as in, what command to open your cli? |
| 19:49.20 | [TK]D-Fender | rubberneck: It happens where a call comes in from a registration that was against 1 IP, but the call oroginates from another (typical of multiple-subnet environments" |
| 19:49.43 | [TK]D-Fender | rubberneck: Try to ensure that both hosts will be the same if the server has IP's on both subnets. |
| 19:50.08 | ruben23 | asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 19:50.49 | *** join/#asterisk blaines (~blaines@67.130.168.2) |
| 19:52.21 | *** join/#asterisk blaines (~blaines@67.130.168.2) |
| 19:52.29 | *** join/#asterisk knctrnl (~aembrey@76.164.169.130) |
| 19:53.03 | knctrnl | is there anyway to execute an additional script when say dialplan reload is executed? |
| 19:57.10 | wcselby | ruben23 - what version of asterisk are you running? |
| 19:57.37 | rubberneck | [TK]D-Fender: Thanks |
| 20:00.27 | ruben23 | asterisk - 1.4.27 |
| 20:01.08 | leifmadsen | knctrnl: #exec |
| 20:01.11 | ruben23 | my asterisk works, its just the CLI is all white text.. |
| 20:01.31 | leifmadsen | ruben23: start asterisk with -n (no color) |
| 20:01.31 | rocksfrow | so i made a new cable |
| 20:01.37 | rocksfrow | no changes. :( |
| 20:02.15 | ruben23 | leifmadsen: asterisk -n..? |
| 20:02.23 | bmoraca_work | rocksfrow: if the problem was the cable, your T1 would not say it was up |
| 20:02.31 | leifmadsen | ruben23: yes... -n ... |
| 20:02.45 | leifmadsen | ruben23: won't work with -r -- needs to be done when you start the asterisk process I believe |
| 20:02.59 | rocksfrow | bmoraca_work, thats what i figured..but others suggested so it was worth a shot |
| 20:03.00 | leifmadsen | (i.e. modify your init script, or add it when you run "asterisk" |
| 20:03.02 | leifmadsen | ) |
| 20:03.08 | rocksfrow | bmoraca_work, telco is saying shit is fine |
| 20:03.21 | rocksfrow | my customer service has been down for 4 hours now |
| 20:03.42 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 20:03.43 | bmoraca_work | rocksfrow: tell them to do a trace from their end while you place a call to see if they actually see the call come across the PRI |
| 20:03.58 | rocksfrow | bmoraca_work, the telco says they can see the pri state going up and back down |
| 20:04.01 | rocksfrow | over and over and over |
| 20:04.04 | knctrnl | leifmadsen: so #exec without any extension. I want it to do it on dialplan reload. not when an particular extension is dialed |
| 20:04.12 | rocksfrow | and i see a message with debug intense on |
| 20:04.14 | rocksfrow | over and over and over |
| 20:04.17 | rocksfrow | while he says he sees that |
| 20:04.22 | leifmadsen | knctrnl: check the documentation to see what #exec does |
| 20:04.22 | rocksfrow | outputing 'unumbered frame' |
| 20:04.32 | leifmadsen | knctrnl: sounds like you don't understand how it works |
| 20:04.39 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.173.128.dsl.dyn.forthnet.gr) |
| 20:05.01 | bmoraca_work | rocksfrow: do they see the call come across? |
| 20:05.19 | Naikrovek | bmoraca_work: it's probably flapping faster than that. |
| 20:05.21 | p3nguin_ | knctrnl: #exec, not Exec(). |
| 20:05.25 | Naikrovek | rocksfrow: is it a T1 or .. you said PRI |
| 20:05.26 | outtolunc | giggles |
| 20:05.36 | leifmadsen | p3nguin_: thanks :) |
| 20:05.46 | rocksfrow | Naikrovek, pri.. |
| 20:06.03 | Naikrovek | rocksfrow: have you asked in #cisco? |
| 20:06.06 | bmoraca_work | rocksfrow: what model PRI card do you have? |
| 20:06.09 | Naikrovek | or #networking |
| 20:06.15 | rocksfrow | i have the digium te220 |
| 20:06.26 | rocksfrow | let me remind you guys this was all working |
| 20:06.28 | rocksfrow | 100% |
| 20:06.40 | rocksfrow | until i unplugged in the server, moved it..plugged it back in |
| 20:06.42 | bmoraca_work | have you tried to replace the card and see if that makes a difference? |
| 20:06.52 | rocksfrow | the t1 card? |
| 20:06.55 | bmoraca_work | yes |
| 20:07.03 | rocksfrow | no, i have no tried that yet..bc the other card does nto have EC on it |
| 20:07.05 | bmoraca_work | also, how far away from the Adtran is the PBX? |
| 20:07.16 | rocksfrow | like 3 feet |
| 20:07.24 | bmoraca_work | rocksfrow: EC is irrelevant in attempting to debug whether or not the card even WORKS |
| 20:08.10 | bmoraca_work | also, an EC card is not needed when tying to a TA900, as the TA900 has EC in it that is superior to any digium card |
| 20:08.17 | rocksfrow | bmoraca_work, right..the card isnt identical |
| 20:08.19 | rocksfrow | its a te205 |
| 20:08.21 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 20:08.29 | rocksfrow | (the one i have to test with) |
| 20:08.54 | bmoraca_work | rocksfrow: so install it and modprobe it. it's close enough that it shouldn't matter. in fact, i think it probably uses the same dahdi module |
| 20:09.24 | bmoraca_work | yep, it does |
| 20:09.33 | rocksfrow | okay..so shut it down |
| 20:09.35 | rocksfrow | swap this card |
| 20:09.35 | bmoraca_work | te205 and te220 use the same dahdi module |
| 20:09.35 | rocksfrow | then reboot? |
| 20:09.42 | bmoraca_work | yep |
| 20:09.44 | rocksfrow | okay.. |
| 20:09.49 | rocksfrow | bb in 2 mins |
| 20:09.52 | knctrnl | so i can do EXEC System(php script.php) ? |
| 20:12.01 | leifmadsen | knctrnl: no.... |
| 20:12.11 | leifmadsen | knctrnl: #exec /path/to/myfile.php |
| 20:12.30 | leifmadsen | knctrnl: anything passed back to STDOUT from the script will be loaded into the dialplan |
| 20:13.01 | knctrnl | what if i dont want it loaded into the dialplan? |
| 20:13.08 | knctrnl | i just want to execute the script |
| 20:13.11 | leifmadsen | knctrnl: then don't pass it to STDOUT |
| 20:13.21 | knctrnl | gotchat |
| 20:13.26 | leifmadsen | knctrnl: just avoid passing anything to STDOUT... seems pretty trival |
| 20:13.40 | leifmadsen | like I said, anything passed back to STDOUT, including "nothing" |
| 20:15.07 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 20:16.17 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 20:16.48 | hardwire | Has anybody seen asterisk suddenly hit a high load and become unusable after one or many attempts to dial a DND SIP device through local channels and a direct SIP dial (no options) |
| 20:17.10 | Kobaz | so, if a sangoma says short circuit detected when you plug in a t1... and then keeps cycleing between short detected, not detected when you unplug it.... would it be safe to assume the card is bad? |
| 20:17.14 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:17.34 | giesen | Kobaz: either that or a wiring issue on your pri |
| 20:17.38 | bmoraca_work | Kobaz: or your cabling is bad |
| 20:17.40 | hardwire | Dial(Local/0012@dundi-priv-local) -> Goto(sip-dial,0012,1) -> Dial(SIP/0012) |
| 20:17.49 | Kobaz | well the cabling is fine |
| 20:17.59 | Kobaz | the card also complains about going out of sync with the pci bus |
| 20:18.04 | bmoraca_work | Kobaz: you certified it? it works with other hardware? |
| 20:18.39 | Kobaz | Mar 1 15:07:43 demo3 kernel: wanec1: The H100 slave has lost its framing on the bus! |
| 20:18.41 | bmoraca_work | hardwire: perchance one of the local channels isn't being properly hungup? what model of phone? |
| 20:18.42 | Kobaz | Mar 1 15:07:43 demo3 kernel: wanec1: The CT_C8_A clock behavior does not conform to the H.100 spec! |
| 20:18.57 | Kobaz | bmoraca_work: yeah the cabling works with other hardware |
| 20:19.57 | Kobaz | i've had problems with this board before when hooked up to a different system |
| 20:20.06 | Kobaz | it would only take this one cable |
| 20:20.18 | Kobaz | any other cable (correct cables), it would complain of a short |
| 20:20.31 | Kobaz | i don't know what i did with the one cable that worked |
| 20:21.28 | rocksfrow | bmoraca_work, so..the other card is pci..new one is pci x |
| 20:21.30 | rocksfrow | :-/ |
| 20:21.35 | rocksfrow | i tried moving the card down one slot |
| 20:21.39 | rocksfrow | for the hell of it, since i had it open\ |
| 20:21.55 | bmoraca_work | rocksfrow: shouldn't matter, unless your system doesn't have any PCI slots |
| 20:22.17 | rocksfrow | bmoraca_work, doesnt |
| 20:22.27 | bmoraca_work | well that's a problem |
| 20:22.31 | rocksfrow | heh |
| 20:22.38 | bmoraca_work | you don't have another asterisk box around you could try? |
| 20:23.16 | rocksfrow | well i have the old box..but its not confiured correctly for the PRI.. |
| 20:23.21 | rocksfrow | i guess i could try to reconfigure that one |
| 20:23.41 | bmoraca_work | it would probably be a good idea to test as much as possible |
| 20:24.10 | rocksfrow | yeah..hopefully that tech makes it out |
| 20:24.23 | rocksfrow | im just going to hit my head on a wall if he gets out here and makes outbound calls fine |
| 20:24.29 | *** join/#asterisk Chodorenko (~chodorenk@86.57.250.150) |
| 20:24.33 | rocksfrow | so lost |
| 20:24.43 | Chodorenko | Hello All |
| 20:25.00 | rocksfrow | OMGGGGG |
| 20:25.02 | rocksfrow | DUUUUUUDE |
| 20:25.05 | rocksfrow | its working bmoraca_work |
| 20:25.08 | wcselby | lol |
| 20:25.14 | wcselby | after moving it down one slot? |
| 20:25.19 | rocksfrow | yes..wtf? |
| 20:25.27 | wcselby | card needed to be reseated maybe? |
| 20:25.38 | rocksfrow | jesus |
| 20:25.39 | rocksfrow | inboud/outbound |
| 20:25.41 | rocksfrow | all working |
| 20:25.41 | Naikrovek | welcome to hardware |
| 20:25.43 | bmoraca_work | rocksfrow: what kind of system is it? could be 100 things that caused the problem |
| 20:25.43 | rocksfrow | WTF! lol |
| 20:25.49 | wcselby | when you moved the box, maybe the card got messed up a bit |
| 20:25.52 | rocksfrow | im happy but so pissed right now, haha |
| 20:25.53 | Naikrovek | the reboot could have done it |
| 20:25.58 | rocksfrow | wcselby, i was thinking that... |
| 20:25.58 | Naikrovek | it could have come unseated a bit |
| 20:26.01 | wcselby | Naikrovek - he rebooted a few times |
| 20:26.05 | Naikrovek | oh i missed that |
| 20:26.07 | rocksfrow | i rebooted plenty |
| 20:26.08 | Naikrovek | reseat then |
| 20:26.10 | rocksfrow | but.. |
| 20:26.11 | rocksfrow | reseat |
| 20:26.11 | wcselby | yeah |
| 20:26.13 | rocksfrow | damn bro wtf |
| 20:26.16 | rocksfrow | thats def possible i mean |
| 20:26.17 | rocksfrow | pci x1 |
| 20:26.17 | Naikrovek | it happens |
| 20:26.21 | rocksfrow | is such a small plug you know? |
| 20:26.22 | wcselby | lol |
| 20:26.25 | Naikrovek | heh |
| 20:26.26 | Naikrovek | well |
| 20:26.31 | wcselby | if it comes out just slight |
| 20:26.34 | wcselby | slightly |
| 20:26.36 | Naikrovek | i ALWAYS reset my cards after i move a box, just in case |
| 20:26.42 | Naikrovek | lessons just like this one in my past |
| 20:26.46 | wcselby | haha |
| 20:26.47 | wcselby | wow |
| 20:27.03 | wcselby | i don't have that kind of dedication |
| 20:27.09 | rocksfrow | dude |
| 20:27.10 | rocksfrow | i mean.. |
| 20:27.13 | rocksfrow | should i try putting it back |
| 20:27.14 | rocksfrow | in the old slot? |
| 20:27.17 | wcselby | i'll bring it back up and do that stuff if something isn't working right |
| 20:27.18 | bmoraca_work | rocksfrow: i would try putting it back, yes |
| 20:27.21 | rocksfrow | yeah.. |
| 20:27.23 | rocksfrow | let me go do that |
| 20:27.30 | rocksfrow | first let me cancel the tech coming out X0 |
| 20:27.32 | rocksfrow | ugh.... |
| 20:27.50 | bmoraca_work | rocksfrow: if it doesn't work, you may need to get some warranty support on that server...you did buy a teir 1 server with warranty, right? |
| 20:28.02 | *** join/#asterisk P1ersson (~P1ersson@213-64-217-60-no50.tbcn.telia.com) |
| 20:30.28 | radcliff | I have a SIP-provider who sends calls into my asterisk box from two different IP-addresses, random each time, how can I add two "host" parameters in sip.conf for this provider? |
| 20:31.32 | kaldemar | you can't. either make two peers with different ip addresses or match by some other means. |
| 20:31.51 | radcliff | how do I match by "other means" ? |
| 20:31.59 | Chodorenko | recently discovered in the release of asterisk 1.4.29.1 a strange error, it for no reason at interrupts the connection for 20 seconds |
| 20:32.18 | Chodorenko | after 20 -25 seconds |
| 20:34.20 | p3nguin_ | radcliff: It's probably easiest to just use two peer definitions. That's how I do it for an ITSP which sends calls from two IP addresses. |
| 20:34.46 | radcliff | p3nguin_: ok, I'll do that then, thanks! |
| 20:34.47 | radcliff | :) |
| 20:36.12 | rocksfrow | okay now is the real test..if this shit continues to work, lol |
| 20:36.21 | rocksfrow | bmoraca_work, yes..brand new server...brand new warranty |
| 20:36.32 | rocksfrow | the more i think about it |
| 20:36.33 | Chodorenko | https://issues.asterisk.org/view.php?id=16932 |
| 20:36.39 | rocksfrow | i did have some trouble getting the cable out when i moved |
| 20:36.53 | rocksfrow | maybe i just popped it a little bit..but that just blows my mind of how why the card would appear to be working though |
| 20:37.05 | rocksfrow | real test is if it works when the server comes up now (back in the original slot) |
| 20:37.27 | *** join/#asterisk aandrade (~aandrade@189.114.181.92.dynamic.adsl.gvt.net.br) |
| 20:38.21 | *** join/#asterisk aces1up (~blah@wsip-24-234-80-23.lv.lv.cox.net) |
| 20:39.27 | rocksfrow | wow.. |
| 20:39.30 | aces1up | i know this isn't an asterisk questions but anyone here familiar with merlin magix systems or have worked with them in the past? |
| 20:39.31 | rocksfrow | bmoraca_work, working..in the same pci slot |
| 20:39.40 | rocksfrow | how the HELL is this possible? |
| 20:39.47 | rocksfrow | lol |
| 20:39.53 | rocksfrow | what a day.. |
| 20:40.07 | rocksfrow | thank you ALL for your help, its very very much appreciated |
| 20:42.34 | *** join/#asterisk crazy_penguin (~crzp@unaffiliated/crazypenguin/x-000001) |
| 20:43.15 | *** join/#asterisk nny (~Scott@64.203.239.83) |
| 20:43.42 | nny | quick q, under voicemail.conf delete=yes/no or delete=0/1 I assume 0 = no and 1 yes? |
| 20:44.00 | rocksfrow | boolean |
| 20:44.10 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 20:44.11 | nny | k thanks |
| 20:44.19 | superbeef | yo |
| 20:44.26 | nny | made sense just checking, brain fart |
| 20:45.09 | superbeef | I'm tracking down DTMF errors.. any idea what DTMF Exception on 12 could possibly imply? [http://pastebin.com/g0BcwtKF] |
| 20:47.27 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 20:50.17 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 20:53.42 | rocksfrow | so is there somewhere to control/prevent my welcome message from being a half second into it when i call it? or is that my phone? |
| 20:54.00 | rocksfrow | sometimes when i call, the initial recording will alrady be playing partially..get me? |
| 20:54.01 | giesen | rocksfrow: add a half second of silence to it |
| 20:54.16 | *** join/#asterisk eppigy (~Dave@216-139-241-102.aus.us.siteprotect.com) |
| 20:54.18 | eppigy | hello |
| 20:54.20 | eppigy | i am dave |
| 20:54.56 | wcselby | rocksfrow - add a Wait(2) to wait 2 seconds, before you start playing your recording |
| 20:55.06 | giesen | or that =) |
| 20:55.23 | [TK]D-Fender | No, you want Silence/2 |
| 20:55.23 | wcselby | or a Wait(1) for 1 second, etc etc |
| 20:55.28 | *** join/#asterisk dinesh___ (~dinesh@77-58-221-165.dclient.hispeed.ch) |
| 20:55.59 | [TK]D-Fender | Playback(silence/2) will give you 2 seconds o have the rtp come up... you need the audio path established. A Wait() won't do that |
| 20:56.09 | leifmadsen | wcselby: Wait(1) won't really do anything because it doesn't play audio (and thus doesn't answer the channel) |
| 20:56.14 | leifmadsen | points at [TK]D-Fender |
| 20:56.23 | jblack | I've often seen answer and then a wait 1. |
| 20:56.25 | leifmadsen | because I don't want to meet your mom! I just want ... |
| 20:56.27 | wcselby | well, he needs to Answer() the channel first :P |
| 20:56.37 | leifmadsen | wcselby: well yes, but you didn't say that :) |
| 20:56.44 | leifmadsen | you must be pedantic my son |
| 20:57.10 | wcselby | leifmadsen - I'm sorry, I didn't want to give him everything.... ;) |
| 20:58.00 | leifmadsen | heh |
| 20:58.05 | leifmadsen | that's what she said |
| 20:58.14 | wcselby | <PROTECTED> |
| 20:58.24 | rocksfrow | giesen, yeah i was thinking about juts rerecording them and pausing a little at the beginning |
| 20:58.44 | rocksfrow | does anybody else use grandstream phones? |
| 20:59.06 | rocksfrow | when making outbound calls theres always like a 2 second delay before it connects..and before i see it in asterisk CLI |
| 20:59.15 | rocksfrow | is that the phone? or just the server taking some time to handle it |
| 20:59.21 | rocksfrow | i'm figuring its the phones.. |
| 20:59.45 | idespinner | rocksfrow, sounds like the digitmap on the phones |
| 21:03.13 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 21:04.32 | rocksfrow | idespinner, digitmap, hrm |
| 21:06.28 | *** join/#asterisk trentcreek (~kvirc@red1.cs.panam.edu) |
| 21:11.44 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 21:13.35 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 21:15.34 | *** join/#asterisk nny (~Scott@64.203.239.83) |
| 21:16.00 | dinesh___ | hi all, I've got a fairly simple problem described here http://codepad.org/xf14oW97 , could someone please give me some hints? |
| 21:16.05 | nny | anyone willing to throw me a hint or two, setting up queues.conf, want it to do a sequential ring, skipping anyone who is on the phone |
| 21:16.11 | dinesh___ | I didn't find anything really useful in hours :/ |
| 21:17.25 | [TK]D-Fender | dinesh___: That already looks fine. Show us where it FAILS |
| 21:17.46 | dinesh___ | well when I call my number it says the number is invalid |
| 21:18.22 | [TK]D-Fender | dineWhere do i see this? |
| 21:18.27 | dinesh___ | but if I put some more basic rules such as Answer() , MP3Player('test.mp3') and Hangup() it works perfectly |
| 21:18.45 | kfife | Her |
| 21:18.52 | kfife | fatfingered--sorry. |
| 21:19.29 | nny | i assume strategy=linear for the sequential part, any advice on setting up queues to skip people on the phone? |
| 21:19.44 | wcselby | nny - strategy=rrmemory ? |
| 21:20.14 | nny | wcselby: thinking if they call out too though |
| 21:20.25 | nny | wcselby: so if they have an active channel, skip them, etc |
| 21:20.39 | dinesh___ | ok I added some -vvvvv |
| 21:20.49 | dinesh___ | [Mar 1 21:26:50] WARNING[7813]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) |
| 21:20.50 | [TK]D-Fender | dinesh___: SIP DEBUG <--------- |
| 21:20.53 | nny | wcselby: but yeah rmmemory is better than linear it seems |
| 21:21.03 | [TK]D-Fender | dinesh___: Ok, that failed to answer the call... |
| 21:21.10 | [TK]D-Fender | dinesh___: So how is that "bad"? |
| 21:21.25 | [TK]D-Fender | dinesh___: if they don't answer... what then? |
| 21:21.26 | dinesh___ | so that might be because no one is registered to "home" ? |
| 21:21.32 | [TK]D-Fender | dinesh___: correct |
| 21:21.58 | wcselby | nny - set call-limit to 1 on each phone in the queue? |
| 21:22.01 | kfife | Are any efforts underway to to incorporate a sligtly more sophisticated database functions like say, a 'simple table in to Asterisk without having to add the additional cost and complexity of ODBC and an external DB? |
| 21:22.06 | nny | i am thinking I have to set penalties in my dialplan when a call hits the extension or they dial out, is this over thinking the process? |
| 21:22.17 | dinesh___ | okay, and you know why my mobile phone provider returns "number invalid" when I try to call it (as a consequence of this) |
| 21:22.19 | wcselby | kfife - you mean like AstDB>? |
| 21:22.20 | nny | wcselby: well I don't want to cripple the phones for other uses |
| 21:22.23 | dinesh___ | when obviously the number is valid |
| 21:22.25 | kfife | correct. |
| 21:22.33 | [TK]D-Fender | kfife: MySQL() |
| 21:22.45 | nny | wcselby: there are reasons why the phone would have 2 lines open, just not for the queue part |
| 21:22.53 | kfife | even that's a bit 'fat'. |
| 21:22.54 | [TK]D-Fender | kfife: there is also AstDB, or SQLIte, etc |
| 21:23.15 | [TK]D-Fender | dinesh___: where do I see the failed call? |
| 21:23.29 | kfife | SQLlite is more my speed--I'm just trying to avoid needing to store many tuples just so I can look things up in reverse. |
| 21:23.37 | dinesh___ | well that's what i get on my mobile phone when i try to call the number provided by "sip.backbone.ch" |
| 21:23.39 | [TK]D-Fender | dinesh___: maybe you should do MORe than just kill the call failing the dial. |
| 21:23.52 | [TK]D-Fender | dinesh___: like doing Busy() afterwards |
| 21:23.58 | kfife | but correct me if I'm wrong, I have to use SQLite with FUNC_ODBC |
| 21:24.01 | dinesh___ | ah, okay thanks a lot |
| 21:24.29 | [TK]D-Fender | kfife: No, it has its own res_ module |
| 21:24.33 | kfife | Wasn't there an 'fork' early on in ast devel by some zealots who wanted SQLite |
| 21:24.36 | nny | wcselby: i was thinking penalties in the dialplan added when they call out or recieve a call directly of some kind |
| 21:24.37 | kfife | Cool. |
| 21:24.43 | kfife | Why can't I see that. |
| 21:24.58 | kfife | Is that an add-on |
| 21:25.07 | [TK]D-Fender | kfife: Because maybe you're missing the libs that would have allowed ti to be built in the first place? |
| 21:25.19 | [TK]D-Fender | kfife: and IIRC it is in addons. |
| 21:25.23 | dinesh___ | yeah thanks so much it's working [TK]D-Fender :) |
| 21:25.32 | [TK]D-Fender | dinesh___: you're welcome |
| 21:25.49 | kfife | [TK]D-Fender: fair enough, but normally I would see it along with a remark about missing dependencies. |
| 21:26.43 | wcselby | nny - I think leifmadsen was working on a method to detect device state, and not call queuemembers who's device state was not available |
| 21:26.48 | [TK]D-Fender | kfifok, checkout time, bbiab |
| 21:26.50 | wcselby | from a while back |
| 21:27.36 | kfife | res_config_sqlite? |
| 21:27.55 | kfife | I though that was for real-time asterisk for storing the dialplan? Am I mistaken ?> |
| 21:28.51 | nny | wcselby: IIRC he mentioned it, I'll have to see what's going on with it |
| 21:30.14 | *** join/#asterisk simplydrew (~simplydre@pool-74-97-190-109.prvdri.fios.verizon.net) |
| 21:34.20 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 21:43.36 | Kobaz | hmm |
| 21:43.52 | Kobaz | getting overruns on a t1 interface... what would that cause? |
| 21:45.21 | *** join/#asterisk puzzled_ (~foobar@puzzled.xs4all.nl) |
| 21:45.44 | wcselby | nny - what version of asterisk are you running? |
| 21:45.45 | *** join/#asterisk matt_d (~matt@70.134.98.183) |
| 21:45.57 | *** part/#asterisk matt_d (~matt@70.134.98.183) |
| 21:46.10 | *** join/#asterisk matt_d (~matt@70.134.98.183) |
| 21:47.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:47.21 | rocksfrow | grandstream indeed has a setting, 'no key entry timeout' |
| 21:47.36 | rocksfrow | default is 4 seconds, which i think is hihg |
| 21:47.37 | rocksfrow | high** |
| 21:48.35 | [TK]D-Fender | rocksfrow: And as soon as you shorten it... you'll think it's low |
| 21:48.47 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 21:48.50 | [TK]D-Fender | GS doesn't HAVE a "dialplan" |
| 21:48.55 | rocksfrow | [TK]D-Fender, lol...i made it 2 seconds |
| 21:49.03 | [TK]D-Fender | ~gs |
| 21:49.04 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:49.09 | rocksfrow | hahahahaa |
| 21:49.10 | rocksfrow | NICE! |
| 21:49.22 | rocksfrow | fortunately, i didn't buy the shit...previous sysadmin did |
| 21:49.36 | rocksfrow | wow.. |
| 21:49.42 | rocksfrow | interesting to see that straight from asterisk though |
| 21:49.44 | rocksfrow | never knew... |
| 21:49.49 | rocksfrow | polycom the way to go? |
| 21:50.00 | Naikrovek | i love polycom myself |
| 21:50.06 | rocksfrow | i almost ordered one of the ip 6000's for our conf. room |
| 21:50.08 | rocksfrow | thing is badass, lol. |
| 21:50.19 | Naikrovek | if you're new to them the setup can be confusing, but after you get them setup they're a dream |
| 21:50.31 | *** join/#asterisk alexx1523 (~abirmingh@sea02-v600-nat.marchex.com) |
| 21:50.33 | Naikrovek | and i'm always willing to help with configuration for anyone who needs it |
| 21:50.53 | rocksfrow | i'm assuming they have a simliar web interface? |
| 21:51.00 | rocksfrow | similar* |
| 21:51.02 | Naikrovek | their web interface sucks and you don't want to use it |
| 21:51.11 | Naikrovek | it's good for seeing how they're configured but not for configuration |
| 21:51.16 | rocksfrow | really? do you telnet into them or something? |
| 21:51.16 | Nugget | telnet is eeeeeeevil! |
| 21:51.20 | rocksfrow | ssh? |
| 21:51.22 | rocksfrow | loool |
| 21:51.29 | Naikrovek | even for a single phone i would recommend setting up an FTP server and pointing the phone to it |
| 21:51.30 | rocksfrow | that's gotta be a bot |
| 21:51.41 | Naikrovek | he's not a bot, but he's got an autoresponder for telnet |
| 21:51.47 | rocksfrow | haha, nice |
| 21:52.03 | rocksfrow | does anybody make any sweet software for managing an office full of them? |
| 21:52.13 | Naikrovek | i have some scripts i can send you |
| 21:52.32 | Naikrovek | i add a small line to a text file then run a perl script to generate the configs |
| 21:52.49 | Naikrovek | then turn the phone(s) on and the phones get the FTP information from DHCP, contact the FTP server, and provision themselves |
| 21:52.49 | rocksfrow | hrm..interesting |
| 21:53.07 | Naikrovek | takes 10 seconds for me, and about 15 for the end user (but only due to firmware upgrade time) |
| 21:53.15 | Naikrovek | 15 minutes, i mean |
| 21:53.40 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 21:54.15 | rocksfrow | i haven't had any issues with my GS's at all though.. |
| 21:54.18 | rocksfrow | other than feature limits |
| 21:54.22 | rocksfrow | but as far as funtionality |
| 21:54.55 | nny | wcselby: 1.6.2 |
| 21:55.06 | nny | wcselby: sorry was afk for a sec |
| 21:55.13 | Naikrovek | rocksfrow: i had some grandstreams and they wouldn't hang up the call when you put the receiver down, they woudlnt' ring half the time |
| 21:55.20 | Naikrovek | one of them died about 3 weeks after i got it |
| 21:55.23 | nny | ~grandstrem |
| 21:55.25 | nny | gah |
| 21:55.25 | Naikrovek | never again yo |
| 21:55.28 | nny | fail |
| 21:55.28 | rocksfrow | ~gs |
| 21:55.29 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:55.30 | nny | ty |
| 21:55.30 | rocksfrow | lol.. |
| 21:55.34 | nny | lol |
| 21:55.38 | nny | ~pollycommunist |
| 21:55.41 | Naikrovek | yes |
| 21:55.42 | Naikrovek | i am |
| 21:55.44 | nny | ok i give up, damn you bot! |
| 21:55.50 | rocksfrow | haha |
| 21:55.52 | nny | that was one of my favorite ones |
| 21:55.52 | Naikrovek | ~polycommunist |
| 21:55.53 | infobot | A polycommunist is someone who believes Polycom phones can do no wrong.. that Polycom's are so over and above anything else, that what you are using is surely crap, and the mere fact you mention another brand name is of great insult to the channel, the community, and the world. They may also be getting a 10% kickback. |
| 21:56.02 | nny | hahhaha |
| 21:56.04 | rocksfrow | hahaha |
| 21:56.05 | Naikrovek | ooh i should get a kickback |
| 21:56.13 | nny | <3 infobot |
| 21:56.17 | Naikrovek | i preach polycom all the time |
| 21:56.39 | rocksfrow | well |
| 21:56.45 | Naikrovek | i love our polycom phones |
| 21:56.52 | rocksfrow | my phone actually has frozen up on me a couple times |
| 21:56.55 | *** join/#asterisk matt_d (~matt@70.134.98.183) |
| 21:56.57 | rocksfrow | i thought that was bc of the PoE though |
| 21:57.29 | rocksfrow | Naikrovek, does polycom have something similar to the gs handytones? |
| 21:57.36 | rocksfrow | i absolutely love those handytones, lol |
| 21:57.40 | Naikrovek | i dunno what a handy tone is |
| 21:57.43 | nny | handytones? |
| 21:57.52 | Naikrovek | wireless portable sip phone? yes they do |
| 21:57.53 | rocksfrow | yeah ip to fxs |
| 21:57.56 | Naikrovek | oh |
| 21:57.59 | rocksfrow | no... |
| 21:58.00 | Naikrovek | no |
| 21:58.06 | Naikrovek | they don't have ATAs |
| 21:58.07 | rocksfrow | oh really? |
| 21:58.20 | rocksfrow | things are sweet |
| 21:58.37 | rocksfrow | i use em for fax |
| 21:58.42 | nny | handytones? http://i48.photobucket.com/albums/f234/jklapp/handy.jpg |
| 21:59.05 | rocksfrow | no, http://www.voiptraders.co.uk/assets/206/Grandstream%20Handy%20Tone%20286%20-%20300x300.jpg |
| 21:59.05 | rocksfrow | lol |
| 21:59.29 | nny | ahh |
| 21:59.33 | Naikrovek | linksys has some ATAs that people talk about all the time in here, Polycom has no FXS adapters |
| 21:59.46 | rocksfrow | so ATA = ?? |
| 21:59.53 | rocksfrow | ata is the same thing as an fxs adapter? |
| 21:59.57 | rocksfrow | what's ATA mean |
| 21:59.57 | Naikrovek | analog telephone adapter |
| 22:00.00 | rocksfrow | ah ok |
| 22:00.03 | nny | actually the gs atas are reasonable afaik |
| 22:00.10 | *** join/#asterisk Geminizer (~whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
| 22:00.10 | nny | just the phones aren't so hot |
| 22:00.29 | rocksfrow | are you guys running your alarms through the pbx? |
| 22:00.33 | rocksfrow | or have a dedicated line for an alarm? |
| 22:00.34 | p3nguin_ | ~ata |
| 22:00.35 | infobot | i heard ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 22:00.36 | rocksfrow | analog line |
| 22:00.41 | nny | analog lines here |
| 22:00.46 | nny | try to avoid crossing em |
| 22:01.00 | rocksfrow | nny, yeah..the old sysadmin has our alarm on a handytone |
| 22:01.03 | rocksfrow | and..honestly thats wack |
| 22:01.09 | Geminizer | Hello guys.. does WaitExten(10) mean "fall through 10 seconds after not receiving any input", or "fall through 10 seconds from receiving the lass digit entry" |
| 22:01.30 | rocksfrow | i've actually read that you can run into insurance problems if you were to get say..robbed while your phone server was down, leaving the alarm disabled |
| 22:02.08 | rocksfrow | nny, we have a dsl line..i was thinking i could probably use that line for the alarm as well? |
| 22:02.14 | Geminizer | e.g. if I am entering a 16 digit number, does WaitExten(10) mean I have 10 seconds to enter all 16 digits, or does it mean the pause between digit presses cannot be equal to or greater than 10 seconds ? |
| 22:02.23 | rocksfrow | i guess i should call the telco to confirm that..but i would think if i put the alarm on a filter it should be cool to do that |
| 22:02.30 | p3nguin_ | geminizer: After a BackGround(), WaitExten() waits the specified amount of time for input. If no input has been received after the time, it exits and the dialplan continues. |
| 22:02.52 | Naikrovek | if your alarm is so important, get a dedicated system (phone line) for it |
| 22:03.07 | p3nguin_ | rocksfrow: I suppose you have ATM machines, PIN numbers, and PCB boards, too. |
| 22:04.28 | rocksfrow | p3nguin_, huh? |
| 22:04.43 | p3nguin_ | "we have a dsl line" |
| 22:04.50 | p3nguin_ | The L in DSL means Line. |
| 22:04.57 | rocksfrow | ohhhhh hahaah |
| 22:04.58 | rocksfrow | smartass |
| 22:04.59 | rocksfrow | lol |
| 22:05.17 | p3nguin_ | :) |
| 22:05.29 | p3nguin_ | Just sayin' |
| 22:05.32 | rocksfrow | Naikrovek, of course the alarm is important :-p |
| 22:05.47 | rocksfrow | but i think sharing the DSL is normal? |
| 22:05.52 | Qwell | p3nguin_: I connected my NIC card in my ATM machine to my DSL line, but it isn't accepting my PIN number. Maybe we need to replace the PCB board? |
| 22:05.55 | rocksfrow | just gotta put one of those filters on it |
| 22:06.10 | rocksfrow | i guess i should call my telco and verify that's cool |
| 22:06.30 | p3nguin_ | qwell: My sentiments, exactly. |
| 22:06.36 | rocksfrow | Qwell, suck it |
| 22:06.45 | rocksfrow | lol |
| 22:07.05 | nny | quick q, whats the best way to have asterisk only try a queue once and then jump to the next item in my context? |
| 22:07.22 | rocksfrow | timeout destination? |
| 22:07.35 | rocksfrow | and skip if full? |
| 22:07.36 | wcselby | nny - set a timeout |
| 22:07.38 | nny | well |
| 22:07.54 | nny | i thought i did ha one sec |
| 22:07.56 | wcselby | Queue(queue_name,,,,120) will ring the queue for 120 seconds, then go to the next step in your dialplan |
| 22:08.17 | wcselby | alright, I think it's time for me to head out |
| 22:08.20 | wcselby | have fun |
| 22:08.24 | wcselby | o/ |
| 22:09.47 | Geminizer | p3nguin_, so you are saying if no extension match occurs within x seconds (as specified by WaitExten(x)), then it will fall through ? |
| 22:10.03 | Geminizer | regardless if digits are entered or not |
| 22:10.27 | [TK]D-Fender | Geminizer: incomplete match = INVALID extension |
| 22:10.29 | p3nguin_ | geminizer: If you enter digits which are invalid extensions, you've still entered digits. |
| 22:10.49 | p3nguin_ | It's just waiting for input. |
| 22:10.56 | nny | hmm i tried timeout=seconds and Queue(something,r,,7) and it keeps trying the first queue over and over after 7 seconds? |
| 22:11.06 | *** join/#asterisk matt_d (~matt@70.134.98.183) |
| 22:11.15 | Geminizer | ok, got it... thanks |
| 22:11.46 | nny | my dialplan has the queue(something,r,,7) queue(something2,r,,7) |
| 22:11.54 | nny | but it sticks on something and doesn't jump to something2 |
| 22:12.15 | p3nguin_ | You could always use the 'n' option for Queue(). |
| 22:12.38 | [TK]D-Fender | nny: count your parameters.. that app's has changed over the versions |
| 22:13.49 | nny | [TK]D-Fender: roger was one , short heh |
| 22:16.40 | *** join/#asterisk Jhirley (~Jhirley@adsl-145-4-166.mia.bellsouth.net) |
| 22:16.51 | nny | [TK]D-Fender: hmm odd, still stays in the first queue after 7000 ms, let me PB my stuff, probably pebkac as usual |
| 22:17.24 | [TK]D-Fender | nny: queue timeout onl;y gets checked after agents have stopped ringing. |
| 22:17.44 | [TK]D-Fender | nny: its a "lowest common denominator" issue |
| 22:18.01 | nny | yeah it says nobody picked up, but then just starts over instead of jumping to next in context |
| 22:18.57 | nny | http://pastebin.org/99659 |
| 22:19.12 | nny | ignore any oddness, just testing right now, both after hours and normal do the same thing |
| 22:19.54 | nny | sorry included queues.conf http://pastebin.org/99660 |
| 22:20.48 | *** join/#asterisk danj1980 (~dan@91.110.3.94) |
| 22:21.26 | [TK]D-Fender | nny: set the timeout to 10 in queues.conf |
| 22:21.32 | [TK]D-Fender | nny: and the retry=5 |
| 22:21.52 | danj1980 | Hi, has anyone had problems with Polycom phones randomly rebooting during a call? |
| 22:22.31 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 22:22.32 | [TK]D-Fender | nny: and from what I can see you should be doing a nested dial w/ local channels to time-delay some memebers instead of using queues |
| 22:25.22 | nny | [TK]D-Fender: yeah was considering that as wel (if you mean Dial(SIP/100&SIP/101&SIP/102) etc |
| 22:25.33 | nny | [TK]D-Fender: was just trying ot make it.. cleaner with queues.conf |
| 22:26.56 | idespinner | danj1980, try upgrading the polycom firmware. Ive heard of it with older firmware |
| 22:27.53 | danj1980 | idespinner: Its using the latest firmware, but still reboots. Polycom want a wireshark taken from the phone, but the phone is on my client's network which I cant control. |
| 22:28.31 | idespinner | danj1980, unless someone is manually rebooting it... your probably gonna need a packet capture to see whats causing it to reboot |
| 22:30.47 | danj1980 | idespinner, can it be a packet capture from our asterisk server? even though its not on the same network? |
| 22:31.41 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 22:32.21 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 22:32.51 | ManxPower-work | danj1980, What specific firmware version? |
| 22:33.46 | ManxPower-work | The only reason I know of for 3.1.x or 3.2.x to cause phones to reboot is if you screwed up the config files |
| 22:34.04 | ManxPower-work | Oh, there is one bug having to do with LLDP support that could cause reboots. |
| 22:34.11 | ManxPower-work | (that's only in 3.1, I think) |
| 22:34.18 | ManxPower-work | ..er... in 3.2 |
| 22:35.15 | danj1980 | 3.2.2.0477 |
| 22:35.21 | Chainsaw | And selecting https in DHCP option 66 causes the SIP application to reboot in the latest firmware on a Polycom 670. |
| 22:35.40 | Chainsaw | But that's immediate, with a 0x4020 error. Not during a call. |
| 22:35.51 | danj1980 | Im not using DHCP to autoconfig. I manually entered the provisioning server. |
| 22:35.58 | ManxPower-work | yup, you either have the LLDP bug or you messed up your config files or the messed up the provisioning server address |
| 22:36.01 | idespinner | danj1980, A pcap on the asterisk side may or may not help. its hit or miss. A pcap of the polycom is gonna be a hit for sure... |
| 22:36.31 | idespinner | if danj1980 has a ticket open with polycom, i'm guessing they already reviewed his config... |
| 22:36.35 | idespinner | may be some new bug |
| 22:36.40 | ManxPower-work | danj1980, what do the phone logs say for the phone that is rebooting? |
| 22:36.49 | *** join/#asterisk lesouvage (~lesouvage@82.73.69.76) |
| 22:36.58 | danj1980 | ManxPower-work: 1sec. i'll open them up. |
| 22:37.20 | lesouvage | Is there a max in the number of peers that can be created automatically with autocreatepeer=yes in the [general] part of sip.conf? |
| 22:38.05 | danj1980 | ManxPower-work, 0225121757|app1 |*|00|Not recognized argument. |
| 22:38.36 | ManxPower-work | there you go |
| 22:39.09 | danj1980 | Yep. already sent over to polycom. |
| 22:39.25 | danj1980 | But they need a wireshark before they can progress the issue. |
| 22:39.40 | danj1980 | The client wont give me access to their server or network. |
| 22:41.18 | danj1980 | Actually, theres quite a lot logged of info logged at the same time index |
| 22:42.18 | nny | [TK]D-Fender: not clear ont he diff between retry and timeout... |
| 22:42.50 | dinesh___ | is there a way to give set the timeout for the "qualify" parameter of sip.conf ? |
| 22:42.58 | dinesh___ | i believe it's like the ping/pong mechanism used on irc |
| 22:43.37 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 22:44.13 | danj1980 | ManxPower-work, can the wireshark be logged from any computer on the network? |
| 22:44.46 | danj1980 | How would I do a wireshark "at the polycom phone"? |
| 22:45.40 | Chainsaw | danj1980: You stick a switch between the phone and the rest of the network. |
| 22:45.53 | Chainsaw | danj1980: Configure a switch port for mirroring, stick a PC/laptop with wireshark on it. |
| 22:46.05 | ManxPower-work | you need to either do the packet capture on the provisioning server, set up your switch to "port mirror" or remove the switch and replace it with a hub. |
| 22:47.59 | *** join/#asterisk k5tux (~RussW_K5T@tempest.bluecows.com) |
| 22:48.15 | [TK]D-Fender | dinesh___: Qualify IS a timeout value |
| 22:48.19 | k5tux | Is it normal for ztdummy to cause the clock on an Asterisk VM to go to hell in a handbasket? |
| 22:48.27 | danj1980 | ManxPower-work, the provisioning server was not being contacted at the time of the reboot. the phone was in a call at the time. |
| 22:49.03 | ManxPower-work | danj1980, then you need replace "provisioning server" in my statement with "asterisk server" |
| 22:49.21 | danj1980 | ok, thanks. |
| 22:53.54 | *** join/#asterisk dunkoh (~dunkoh@MW-ESR1-72-49-37-45.fuse.net) |
| 22:55.05 | lesouvage | Yesterday I asked about autogenerating sip entries for an OpenBTS based solution based on log data of failed registration attempts. autocreatepeer=yes is doing the trick. It is working great but I have no idea for how many to register sim cards. |
| 22:57.01 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 22:58.47 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 22:59.22 | ManxPower-work | LOL! "Connected to Asterisk UNKNOWN__and_probably_unsupported currently running on pbx (pid = 12620)" |
| 22:59.24 | *** join/#asterisk [netman] (~netman@193.153.154.9) |
| 23:00.18 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 23:00.34 | lesouvage | I didn't expect that something out of the box would be available, nice surprise. |
| 23:03.41 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 23:06.16 | *** join/#asterisk Carlos_PHX (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 23:08.11 | jaytee | has anyone ever used Vonage as an ITSP with Asterisk? |
| 23:08.37 | ManxPower-work | jaytee, you just need a special account with Vonage |
| 23:08.52 | jaytee | they sell plain SIP termination? |
| 23:09.21 | jaytee | or do you have to kludge their adapters? |
| 23:09.32 | p3nguin_ | Why not sniff the normal traffic and apply the credentials to your Asterisk configuration? |
| 23:09.56 | ManxPower-work | My info *is* quite old, things could have changed. As I understand it their unlimited offer is limited to only their locked adapters. You can add a "softphone account", which will work with Asterisk but does NOT have unlimited calling. |
| 23:10.06 | p3nguin_ | I'm pretty sure they use regular SIP on their adaptors. |
| 23:10.21 | ManxPower-work | I imagine google will know more current info. |
| 23:10.54 | jaytee | ManxPower-work and p3nguin_ thanks! |
| 23:16.41 | *** join/#asterisk engrxyz (~jkjkjk@host81-143-50-92.in-addr.btopenworld.com) |
| 23:19.52 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 23:20.29 | adnc | hello, what role is res_odbc.c playing and would i need it if i only wanted cdr with sqlite database recording? |
| 23:24.33 | *** join/#asterisk dwery (~dwery@nslu2-linux/dwery) |
| 23:24.53 | dwery | hello. does asterisk support native ISDN (PRI/BRI) call deflection? |
| 23:29.05 | leifmadsen | wow, adnc waited a whole 7 minutes |
| 23:29.24 | p3nguin_ | Ping timeout |
| 23:29.40 | leifmadsen | ah |
| 23:29.43 | p3nguin_ | I assume he left without his own consent. |
| 23:29.56 | leifmadsen | I didn't notice that part :) maybe his network doesn't permit idling :) |
| 23:31.36 | ManxPower-work | dwery, in 1.4 BRI w/BRI-Stuff patches. PRI in 1.4, 2BCT is supported on one specific switch (DMS?), in 1.6 I thnk 2BCT is supposed to be supported on national 2, but I'm not sure. |
| 23:32.27 | phix | leifmadsen: amazing |
| 23:33.46 | dwery | ManxPower-work: ty. it seems DAHDISendCallreroutinfFacility is what I'd nee |
| 23:33.47 | dwery | d |
| 23:37.04 | dinesh___ | hm, I'm playing an mp3 from my asterisk server to an x-lite client |
| 23:37.07 | dinesh___ | the quality is pretty bad |
| 23:37.24 | dinesh___ | should I disallow some codecs, or is it automatically going to pick up the best one ? |
| 23:37.42 | dinesh___ | right now i allowed them all |
| 23:38.05 | leifmadsen | dinesh___: it picks the least cost based on what is allowed |
| 23:38.25 | leifmadsen | i.e. if no transcoding is required, then it will pick that first |
| 23:38.28 | dinesh___ | cost in terms of bandwidth , or cpu ? |
| 23:38.29 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 23:38.30 | leifmadsen | order does matter in sip.conf |
| 23:38.33 | leifmadsen | dinesh___: CPU |
| 23:38.53 | dinesh___ | hm, okay, and isn't there some kind of codec agreement between the client and the server ? |
| 23:39.18 | dinesh___ | like browsers support several encryption methods and go for the strongest one |
| 23:40.05 | dzup_ | if i encrypt all sip comunications the load on bandwidth get heavier right? |
| 23:40.23 | *** join/#asterisk doneir (~cbrunker@appenp.lnk.telstra.net) |
| 23:41.09 | *** part/#asterisk Corydon76-lap (~Corydon76@nat/digium/x-qlfuqrygeqjcqayb) |
| 23:41.57 | dlynes | Is there a way to do math in the dialplan? |
| 23:42.19 | doneir | i've been trying to find out about reserving DAHDI channels. For example, asterisk setup has 2 lines, Asterisk is setup to dial a specific number at a specified time (via .call file), when the user picks up, the user tells asterisk to dial another number and connect for a conference. However, if the second line is taken up during this process by an incoming call, this conference will fail |
| 23:42.27 | dlynes | i.e. if I want to create a macro that takes the number of rings, instead of seconds, multiply that number by 5, and then pass it on to the dial app? |
| 23:42.49 | doneir | so i'm looking to reserve/block a channel from incoming calls until specified by the dial plan |
| 23:42.54 | tzafrir | doneir, can you use dialing by a group? |
| 23:42.59 | tzafrir | e.g. DAHDI/g1 |
| 23:43.34 | doneir | yep, that's what i'm currently doing, it's actually 10 lines, and we need to have all open for various activities, so i can't split up incomign and outgoing into groups |
| 23:43.50 | p3nguin_ | leifmadsen: If the codec with the least cost is picked, how does order matter? |
| 23:43.51 | doneir | there could be 7 incoming and in use, and 2 outgoing in conferences |
| 23:44.07 | doneir | i've looked at DEVICE_STATE, but this can not be set on DAHDI channels |
| 23:44.30 | doneir | is there a way to set an 'off hook' mode on a channel? |
| 23:44.43 | p3nguin_ | leifmadsen: or vice versa -- if the order of the listed codecs matter, how is the codec with least cost chosen? |
| 23:45.04 | *** join/#asterisk QbY (~QbY@c-24-126-145-123.hsd1.ga.comcast.net) |
| 23:45.12 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.12.93) |
| 23:45.15 | QbY | Does anyone know how LD works in Japan? |
| 23:45.18 | doneir | the only other way i could possibly do it, is by creating an external call handling app (using a DB or such) |
| 23:45.30 | doneir | but if there's a simple way i'm currently missing, that would be preferred :) |
| 23:45.38 | dinesh___ | well if they have all the same cost, then the order matters p3nguin_, i'd suppose |
| 23:45.58 | p3nguin_ | G.729 and G.711u, for example. |
| 23:46.03 | p3nguin_ | Same cost? |
| 23:46.42 | p3nguin_ | If the originating channel is G.729, isn't the least expensive codec for the other leg going to be G.729 as well? |
| 23:47.00 | p3nguin_ | <PROTECTED> |
| 23:47.33 | p3nguin_ | But I just tested that theory, listing g729 first and ulaw second... and it chose ulaw. |
| 23:47.40 | phix | p3nguin_: G.729a has a licence fee too as well a CPU overhead compared with G.711u |
| 23:48.24 | phix | p3nguin_: of course G.711u uses more bandwidth |
| 23:49.15 | p3nguin_ | If the call comes in using g729, and g729 is an allowed codec on the phone where the call is going, no transcoding needs to be done, making it a lesser cost. |
| 23:50.36 | p3nguin_ | Order does not seem to make a lick of difference in my test scenario. |
| 23:52.10 | leifmadsen | dlynes: MATH() function? |
| 23:52.48 | leifmadsen | dlynes: expressions can also do simple math: Set(RESULT=$[5 + 7]) |
| 23:52.58 | p3nguin_ | If I do not allow=ulaw at all, then (as expected) g729 is chosen... but only under that circumstance. |
| 23:53.55 | leifmadsen | p3nguin_: imagine both ends say "I can support ulaw, alaw, and g729" then the order should matter in that case with the first listed the preferred |
| 23:54.26 | leifmadsen | p3nguin_: if one end supports g729 and the other ulaw and g729, then g729 should be the ideal since there would be no transcoding involved |
| 23:54.53 | p3nguin_ | leifmadsen: In this test case, the order does not matter at all -- if ulaw is allowed, ulaw is being chosen, regardless of order. |
| 23:54.59 | leifmadsen | p3nguin_: it's been forever since I've tested, so version might matter too. I'd have to look at the SDP and all that for various clients |
| 23:55.08 | p3nguin_ | Using 1.4.29. |
| 23:55.10 | dinesh___ | anyway looks pretty much like x-lite only supports ulaw and perhaps alaw |
| 23:55.28 | leifmadsen | it won't support g.729 for sure |
| 23:58.15 | dinesh___ | the usual order would be G.729, G.726, ulaw, alaw, and exclude G.723 ? |
| 23:58.52 | dinesh___ | that's all the codecs supported by the linksys SPA2102 |
| 23:58.56 | p3nguin_ | I would kind of like to make that work right... I prefer to use g729 from my provider and if I only allow g729 on the peers, then no transcoding needs to be done. However, I would like to use ulaw between peers on the local network, so I want to also allow ulaw for those peers. |
| 23:59.04 | leifmadsen | p3nguin_: using 1.6.2 branch order matters for me |
| 23:59.24 | leifmadsen | allow=ulaw first causes ulaw to be used, and allow=g729 first causes .g729 prompts to be used |
| 23:59.31 | leifmadsen | p3nguin_: you have both sound files installed right? |
