| 00:00.10 | mvanbaak | jaytee: 08 - Sex is my religion |
| 00:01.16 | hardwire | I hope it's not some home music mvanbaak recorded on through some sort of cryptomic. |
| 00:02.28 | mvanbaak | no |
| 00:02.42 | mvanbaak | checkout the name on wikipedia |
| 00:03.07 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 00:03.13 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279394325.dsl.bell.ca) |
| 00:04.26 | hardwire | can't seem to find them on wikipedia |
| 00:05.35 | hardwire | I wear crampons all the time tho :) |
| 00:10.10 | jaytee | game 7's on, be back later |
| 00:10.21 | jaytee | Go Red Sox!!!! |
| 00:11.39 | hardwire | mvanbaak: testing the warez now. |
| 00:12.28 | mvanbaak | be my guest |
| 00:12.30 | hardwire | mvanbaak: got /incoming/ ? |
| 00:12.36 | mvanbaak | yup |
| 00:12.46 | hardwire | ever heard Oar? |
| 00:12.53 | mvanbaak | nope |
| 00:12.58 | hardwire | joo may like |
| 00:13.11 | hardwire | I'm digging track 8 |
| 00:14.12 | mvanbaak | :) |
| 00:16.46 | hardwire | O.A.R. was good brain music for me a while ago |
| 00:16.50 | hardwire | mainly one track |
| 00:16.59 | hardwire | otherwise.. tool and techno fill in the gaps |
| 00:17.06 | hardwire | when I sing.. I get more done |
| 00:17.08 | hardwire | so Tool |
| 00:19.12 | jaytee | mvanbaak, you've got a great movie collection |
| 00:19.24 | mvanbaak | jaytee: thank you :) |
| 00:19.36 | mvanbaak | I like it |
| 00:19.42 | mvanbaak | I seen 90% of it |
| 00:19.55 | mvanbaak | the other 10% is still in my TODO |
| 00:19.57 | jaytee | you don't have Mallrats though |
| 00:20.25 | hardwire | sigh |
| 00:20.29 | mvanbaak | jaytee: there is an /incoming/ directory ;-) |
| 00:20.37 | hardwire | stares at jaytee until he makes sense |
| 00:20.41 | hardwire | OH ITS AN ASTRONAUT! |
| 00:21.12 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 00:22.04 | jaytee | my favorite part of Clerks is when Dante's girlfriend screws the dead homeless guy in the employee bathroom |
| 00:22.16 | jaytee | and still thought it was Dante |
| 00:22.37 | mvanbaak | yup |
| 00:22.39 | mvanbaak | lol |
| 00:23.40 | *** join/#asterisk LiNeTuX_Home (n=LiNeTuX@253.238.95.24.cfl.res.rr.com) |
| 00:25.58 | hardwire | can apparently pump a few ulaw channels to mvanbaak |
| 00:26.03 | hardwire | I hate my home dsl |
| 00:26.17 | mvanbaak | hardwire: 8/1 mbit here |
| 00:26.18 | mvanbaak | :) |
| 00:26.27 | hardwire | ass |
| 00:26.33 | mvanbaak | 39 euro's a month |
| 00:26.45 | hardwire | I pay $90/mo for 768k/768k |
| 00:26.48 | hardwire | shared |
| 00:26.52 | *** join/#asterisk pg1054 (n=pg1054@unaffiliated/pg1054) |
| 00:27.01 | mvanbaak | and next weekend I'm moving to my new house where I'll have 20/2 mbit |
| 00:27.08 | mvanbaak | for 19 euro/month |
| 00:27.09 | hardwire | punches mvanbaak |
| 00:27.24 | hardwire | thats it.. screw alaska |
| 00:27.32 | hardwire | moves to mvanbaak's spare room |
| 00:27.56 | mvanbaak | actually, the place above me is free in februari |
| 00:28.08 | [TK]D-Fender | hardwire: I'd rather restrict it to a few choice individuals ;) |
| 00:28.33 | hardwire | [TK]D-Fender: rawr? |
| 00:28.47 | hardwire | are you mvanbaak's landlord ? :) |
| 00:28.54 | [TK]D-Fender | hardwire: Rather than screwing the entire state ;) |
| 00:28.59 | hardwire | oh.. hah |
| 00:29.17 | hardwire | I've been doing a loooot of research |
| 00:29.34 | hardwire | I'm forcing my company to be move involved with the fiber between states, and in town in Anchorage |
| 00:29.47 | hardwire | soon... SOON.. Anchorage will have my company providing MAN services |
| 00:29.55 | [TK]D-Fender | hardwire: Can you see Russia from your house? |
| 00:30.05 | hardwire | [TK]D-Fender: why do people keep asking me that?! |
| 00:30.11 | jaytee | he's been smoking too much Matanuska Thunderfuck |
| 00:30.24 | hardwire | Matanuska ThunderFunk. |
| 00:30.46 | pg1054 | I've found & read how to install additional sound files, etc. Simple & clear. I'm unclear whether I *need* all formats (ulaw, alaw, GSM, etc) for some reason, or only for the formats that I codecs that I use between * <-> trunks? |
| 00:30.47 | *** join/#asterisk lunaphyte (n=lunaphyt@unaffiliated/lunaphyte) |
| 00:30.53 | jaytee | that may be what it's called now but not what they called it in the 70's |
| 00:30.57 | hardwire | there's a cool band up here called Alaska Thunder Funk (ATF) |
| 00:31.21 | lunaphyte | hi :) |
| 00:31.31 | hardwire | two funky black guys with a really nerdy white band. |
| 00:31.45 | pg1054 | (oops ... that's "or only for the codecs that I use between" ...) |
| 00:31.54 | hardwire | ok mvanbaak.. check out incoming/O.A.R./Risen |
| 00:31.58 | hardwire | song 8 is my favorite |
| 00:32.03 | mvanbaak | hardwire: thank you |
| 00:32.15 | hardwire | you may not like it.. but hey.. not like you don't have the bandwidth |
| 00:32.27 | hardwire | I haven't listened to this album in a long while |
| 00:32.35 | *** join/#asterisk jeev (n=email@unaffiliated/jeev) |
| 00:32.51 | jaytee | hardwire, do you own an oosik? |
| 00:32.58 | mvanbaak | hardwire: I'll run it through easytag and move it to /Music/ |
| 00:33.08 | jeev | wankers |
| 00:33.12 | hardwire | jaytee: my oosik isn't owned.. it's for everybody to share. |
| 00:33.19 | jaytee | hahaha |
| 00:33.26 | mvanbaak | ok all |
| 00:33.30 | hardwire | night. |
| 00:33.35 | jaytee | nite mvanbaak |
| 00:33.45 | hardwire | mvanbaak: it's pre musicbrainzed |
| 00:33.50 | hardwire | err.. post |
| 00:33.51 | mvanbaak | download the file "/Music/OpenBSD/song40.mp3" |
| 00:34.37 | mvanbaak | can also be found here: ftp://ftp.openbsd.org/pub/OpenBSD/songs/song40.mp3 |
| 00:34.45 | hardwire | it has a song? |
| 00:34.59 | hardwire | mvanbaak: we all thought you were going to bed :) |
| 00:35.12 | mvanbaak | listen to it |
| 00:35.26 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-8c5ced2ee094cc65) |
| 00:35.36 | mvanbaak | and I'm going to bed in 2cm (that's what's left of my last smoke of today) |
| 00:36.44 | hardwire | my ears! |
| 00:36.52 | mvanbaak | lol |
| 00:37.24 | mvanbaak | http://www.openbsd.org/lyrics.html#40 |
| 00:37.51 | mvanbaak | "OpenBSD 4.0 CD2 track 2 is an |
| 00:37.52 | mvanbaak | uncompressed copy of this song." |
| 00:41.00 | lunaphyte | i have a pap2t that i'm successfully using for calls through asterisk to a term. provider, but am having trouble making calls between sip endpoints connected to asterisk. the calls will ring the proper device, but i have no audio. i've set verbosity to 90, but don't see errors. where can i look for clues? |
| 00:41.04 | drmessano | So a 37 year old operating system using a 2000 year old song for its theme? |
| 00:41.07 | drmessano | Fitting |
| 00:41.51 | hardwire | sung by drunken polkaists |
| 00:42.00 | mvanbaak | whehehehehehe |
| 00:45.28 | mvanbaak | ok, I'm off to bed |
| 00:45.30 | mvanbaak | latero all |
| 00:45.35 | mvanbaak | have fun with my FTP |
| 00:45.36 | mvanbaak | ;) |
| 00:46.15 | jeev | someone DoS it |
| 00:46.26 | jaytee | STFU douche |
| 00:46.32 | jeev | lick my sack |
| 00:46.54 | drmessano | jeev: Everybody here hates you, more than they hate me. How does that feel? |
| 00:46.59 | jaytee | goes to get his microscope |
| 00:47.16 | jaytee | drmessano, I don't hate you at all |
| 00:47.23 | jeev | why do you need your microscope? have a chip in your tire of your rollerblade(s) ? |
| 00:47.26 | drmessano | jaytee: shoosh |
| 00:47.46 | jaytee | jeev, so I can actually find that sack |
| 00:47.50 | jeev | 0_o |
| 00:47.52 | jeev | it's RIGHT THERE! |
| 00:48.04 | jaytee | it's probably tinier than Ahmadickinyourassijads |
| 00:48.12 | jeev | ahmadinejad > bush |
| 00:48.40 | drmessano | It's a little inside joke that his girlfriend has nicknamed her "hooha" the hadron particle accelerator, for obvious reasons |
| 00:48.53 | jaytee | drmessano, I think they've locked down both threads in the forum now |
| 00:50.30 | jeev | needs prunes |
| 00:50.38 | drmessano | fantastic |
| 00:54.06 | hardwire | jeev: do you boil them at all? |
| 00:55.00 | jaytee | drmessano, have you seen this? it's interactive. http://www.palinaspresident.us/ |
| 00:55.07 | drmessano | Yes |
| 00:55.16 | drmessano | "Where did russia go?" <-- classic |
| 00:55.24 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-37-238.pskn.east.verizon.net) |
| 00:55.34 | jaytee | "Learnin is hard!" |
| 00:56.04 | jaytee | they add new stuff everyday |
| 00:56.43 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 00:57.28 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:02.43 | hardwire | the red phone stalls my browser |
| 01:03.18 | lunaphyte | given a user who might be registered via differing methods from time to time (sip one time, iax another, etc) is it sensible to simply Dial() their various endpoints in order, or is there a more appropriate method? |
| 01:03.48 | hardwire | lunaphyte: I'd find a way to set a variable on "last registered" |
| 01:03.55 | hardwire | or use a ringall |
| 01:04.07 | hardwire | dial(tech/blah&tech/blah) |
| 01:04.30 | hardwire | for users like that @ my work.. I force them to use an agent channel |
| 01:04.39 | hardwire | it makes life less annoying for everybody |
| 01:05.20 | lunaphyte | can the ringall approach be combined with a mechanism to know if an endpoint is currently registered? e.g. only dial endpoints that are currently registered/reachable? |
| 01:06.16 | [TK]D-Fender | lunaphyte: Go see if regexten exists for each channel driver you need |
| 01:07.03 | [TK]D-Fender | lunaphyte: I know it does for SIP |
| 01:07.35 | lunaphyte | channel driver refers to sip / iax / sccp, etc? |
| 01:08.30 | hardwire | lunaphyte: if the channel doesn't exist.. nothing happens |
| 01:08.33 | hardwire | or if it's not registered |
| 01:08.36 | hardwire | if it is.. it rings it. |
| 01:08.38 | jeev | boil what |
| 01:08.39 | hardwire | it won't error out |
| 01:08.44 | lunaphyte | i see. |
| 01:08.45 | hardwire | jeev: dried prunes |
| 01:08.47 | jeev | no man |
| 01:08.50 | jeev | i took a crap juts now |
| 01:08.51 | jeev | anyway |
| 01:09.01 | hardwire | ew |
| 01:09.01 | jeev | i named the big one after jaytee |
| 01:09.04 | hardwire | hah |
| 01:09.16 | jeev | HAHAHAHA http://www.palinaspresident.us/ "baby names" hahahahahahha |
| 01:09.33 | jaytee | click on the empty frames on the wall |
| 01:09.34 | lunaphyte | hardwire: what is it regarding using an agent channel that makes life less annoying for everybody? |
| 01:09.47 | jeev | lol |
| 01:09.55 | jaytee | and the panels on the front of the desk |
| 01:10.00 | hardwire | lunaphyte: our employees want to be reached wherever they are |
| 01:10.13 | hardwire | so we have tons of "devices" that are nothing but channels for agents to register on |
| 01:10.30 | jeev | hahahah |
| 01:10.31 | hardwire | if they register on a device, then it rings that device |
| 01:10.31 | jeev | maverick |
| 01:10.44 | hardwire | lunaphyte: that's all done through the dialplan |
| 01:11.00 | hardwire | every night at midnight I reset the agents to a default |
| 01:11.04 | jeev | north koreastan hahahahaha |
| 01:11.18 | lunaphyte | hardwire: oh - so it's more of an approach of the person using the device registering, so-to-speak, as opposed to the physical device being used registering? |
| 01:11.31 | hardwire | lunaphyte: exactly |
| 01:11.34 | hardwire | register a ton of devices |
| 01:11.41 | hardwire | but you don't need to put them in the dialplan |
| 01:11.47 | lunaphyte | that sounds really desirable. |
| 01:12.00 | hardwire | it.. IS.. really desirable |
| 01:12.01 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:12.02 | lunaphyte | heh |
| 01:12.22 | hardwire | another goofy thing you can do is use dynamic queue members |
| 01:12.40 | hardwire | that way they can register from multiple devices |
| 01:12.40 | lunaphyte | i'll have to read a bit then. i see agents.conf. |
| 01:12.44 | hardwire | mah |
| 01:12.47 | hardwire | use queues.conf |
| 01:12.50 | hardwire | trust me |
| 01:12.50 | lunaphyte | oh? |
| 01:12.56 | lunaphyte | ok. |
| 01:12.58 | hardwire | agents are teh sweet |
| 01:13.04 | hardwire | but the digium folks are phasing it out |
| 01:13.11 | hardwire | at least they SAY they are |
| 01:13.19 | hardwire | BUT THEY HAVE YET TO BACK IT UP! |
| 01:13.35 | lunaphyte | queue ~= agents then, or did i misunderstand your last comment? |
| 01:14.12 | hardwire | queues are normally used when you want a ton of people answering an incoming caller |
| 01:14.19 | hardwire | but it can be used similarly to agents |
| 01:14.22 | lunaphyte | i see. |
| 01:14.35 | hardwire | use either or |
| 01:14.40 | hardwire | agents is easy to set up |
| 01:14.46 | hardwire | queues is easy to expand |
| 01:15.15 | lunaphyte | does the phone being used need to explicitly support agents? |
| 01:15.22 | lunaphyte | how does one "log in"? |
| 01:15.37 | hardwire | there is dialplan logic for logging an agent in |
| 01:15.45 | hardwire | check out voip-info.org and chan_agent |
| 01:15.55 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:15.57 | lunaphyte | yeah, looking as we're talking :) |
| 01:16.03 | hardwire | the phone just needs to be callable from the asterisk system |
| 01:16.06 | hardwire | it can even be a mobile number |
| 01:16.13 | lunaphyte | hmm. |
| 01:16.14 | hardwire | it's up to you to program how that dials out |
| 01:16.26 | hardwire | but in most cases it groks the incoming channel tech and ID perfectly |
| 01:16.58 | lunaphyte | when i sit down in front of a phone that isn't "mine" though - how do i tell asterisk i'm there? |
| 01:17.30 | hardwire | dial the agent login extension that you create |
| 01:17.38 | lunaphyte | ah, i see. |
| 01:18.45 | lunaphyte | so, i could in theory, call my own extension, and then possibly provide a passcode or whatever, then it knows the device i've just done this from should be where calls are sent (maybe oversimplified)? |
| 01:18.46 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:19.30 | hardwire | whats with the dinosaur? |
| 01:19.49 | lunaphyte | me? |
| 01:19.51 | jeev | hardwire, how's alaska |
| 01:20.12 | hardwire | full of pricks |
| 01:20.22 | lunaphyte | hardwire: you live in ak? |
| 01:20.24 | jeev | all 680,000 are pricks? |
| 01:20.33 | lunaphyte | whereabout? |
| 01:20.35 | jeev | lunaphyte, he's a Sarah Palin supporter |
| 01:20.39 | lunaphyte | heh |
| 01:20.44 | jeev | just kidding |
| 01:21.10 | lunaphyte | nothing wrong with wasilla. as many freaks there as anywhere else. :) |
| 01:21.27 | jeev | lol |
| 01:21.39 | hardwire | lunaphyte: Anchorage/Eagle River |
| 01:21.42 | lunaphyte | hardwire: i grew up there. i haven't been back for a few years though. |
| 01:22.04 | *** join/#asterisk freakazoid0223 (n=mattc@68.238.189.158) |
| 01:22.04 | hardwire | ran away screaming? |
| 01:22.06 | lunaphyte | anchorage/ bayshore |
| 01:22.48 | hardwire | lunaphyte: where are you now? |
| 01:23.04 | jeev | he's in Boston |
| 01:23.38 | *** join/#asterisk salzh (n=Administ@58.247.195.160) |
| 01:24.10 | hardwire | whyfor? |
| 01:24.32 | lunaphyte | boston? yuck. no, in sw michigan these days. |
| 01:25.23 | hardwire | sounds about the same. |
| 01:26.07 | lunaphyte | hardwire: not too much screaming. i was just out of jr high, so it wasn't up to me. i actually get pretty nostalgic if i think too much about it :) |
| 01:26.15 | hardwire | lunaphyte: I'm a maverick. |
| 01:26.24 | lunaphyte | haha. |
| 01:26.46 | lunaphyte | i think the list of who's not is probably shorter these days. |
| 01:27.26 | lunaphyte | you mean sw mich sounds about the same as ak? |
| 01:27.37 | hardwire | yeh |
| 01:27.53 | hardwire | probably less chinese food tho. |
| 01:27.57 | lunaphyte | that's probably a fair argument, in some respects. |
| 01:28.15 | hardwire | probably fewer mavericks |
| 01:28.19 | hardwire | like.. I dunno.. |
| 01:28.22 | lunaphyte | actually, i live out in the boondocks and have 3 places w/in pickup distance. |
| 01:28.31 | hardwire | a year ago word got out that I'm a real "maverick" |
| 01:28.37 | hardwire | which I assumed was a nice way of saying "prick" |
| 01:28.39 | hesco | is there a goto function or other means for moving a call from one context to another? |
| 01:28.57 | lunaphyte | you grow up there or end up there? |
| 01:29.05 | hardwire | end up |
| 01:29.06 | hardwire | from Colorado |
| 01:29.58 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:30.06 | lunaphyte | ah |
| 01:30.40 | hardwire | but.. I had a hard time fitting in with all the microsofties up here |
| 01:31.01 | lunaphyte | ugh. |
| 01:31.04 | hesco | ok, I see it now, Goto, who woulda thought |
| 01:31.21 | lunaphyte | that's a challenge pretty much anywhere. |
| 01:31.24 | hardwire | and I seemed more adept at helping people get out of the humdrum microworld and do other things |
| 01:31.52 | hardwire | so I got coined a "maverick" haha.. for being an ass in meetings when people tout their awesome and standing up for the little guys |
| 01:31.56 | hardwire | anyways.. I thought it was funny |
| 01:32.08 | hardwire | so I posted a resume on craigslist a while back called "I.T. Maverick" :) |
| 01:32.14 | lunaphyte | gtfo. |
| 01:32.25 | hardwire | and then all the maverick stuff started up with the election stuffins. |
| 01:32.29 | hardwire | it was pretty funny to me. |
| 01:32.37 | lunaphyte | well, i hear there's possibly an opening for governor :p |
| 01:33.00 | hardwire | I am a maverick.. it's a shoe-in |
| 01:33.26 | lunaphyte | are the hickels still a big name around there in politics? |
| 01:34.31 | hardwire | not really |
| 01:35.10 | lunaphyte | i guess that was longer ago then i remember, now that i'm thinking about it. |
| 01:36.09 | hardwire | you an old fart? |
| 01:36.34 | lunaphyte | some days it seems like it, but not really - 32. |
| 01:36.42 | hardwire | 29 |
| 01:37.05 | lunaphyte | how long you been up there? |
| 01:37.09 | hardwire | 5 or so |
| 01:37.13 | hardwire | grabbed a girl and headed north |
| 01:37.16 | hardwire | like the ancient caveman |
| 01:37.41 | hesco | alright! success with that test. Time for a break. back later. |
| 01:37.44 | lunaphyte | ha. funny. i was in ny before moving here, and did the same thing. grabbed a girl and got the f out. |
| 01:39.43 | lunaphyte | iirc, '01 was when i was last up there. |
| 01:39.45 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 01:40.01 | ZX81 | hi all, I'm seeing a really weird problem and wondered if anyone could help: |
| 01:40.08 | hardwire | maybe |
| 01:40.11 | ZX81 | I have a machine with realtime mysql |
| 01:40.22 | ZX81 | if I disable realtime no problem |
| 01:40.32 | ZX81 | if enabled the machine locks hard after starting the call |
| 01:41.25 | ZX81 | no problems with cdr_mysql.so - the records are stored |
| 01:41.25 | ZX81 | I've taken the mysql realtime db from a functioning machine |
| 01:41.26 | ZX81 | Asterisk 1.4 svn |
| 01:41.26 | lunaphyte | hardwire: anyway, thanks for the agent/queue clues and the chat. |
| 01:41.26 | lunaphyte | i'm off to do some reading. |
| 01:41.26 | ZX81 | no crash, no debug - /proc/sys/kenel/panic set to 5 but it doesn't reboot |
| 01:41.56 | ZX81 | if I cause realtime to fail to connect to mysql (i.e. wrong password) it works |
| 01:42.05 | ZX81 | so, no crash |
| 01:42.15 | hardwire | wowsa |
| 01:42.21 | ZX81 | :) |
| 01:42.31 | hardwire | did you file a bug? |
| 01:42.34 | hardwire | cause yeh.. |
| 01:42.35 | hardwire | bug. |
| 01:42.35 | ZX81 | nah not yet |
| 01:42.50 | ZX81 | have seen two similar probs in the last few days |
| 01:43.03 | ZX81 | might just go back through revs of asterisk-addons to check it all |
| 01:43.30 | hardwire | ZX81: tried ODBC instead? |
| 01:43.35 | ZX81 | heh |
| 01:43.36 | ZX81 | :) |
| 01:43.36 | hardwire | nobody likes supporting the mysql stuff |
| 01:43.43 | ZX81 | yeah I know |
| 01:43.48 | hardwire | just do it |
| 01:43.50 | hardwire | DO IT |
| 01:43.51 | ZX81 | I can fix it myself though |
| 01:43.52 | hardwire | DOO IIIIIT |
| 01:43.55 | ZX81 | if I find the problem |
| 01:43.56 | ZX81 | :) |
| 01:44.06 | hardwire | word |
| 01:44.47 | hardwire | I.. officially.. cannot concentrate. |
| 01:44.49 | ZX81 | I probably should move to sqllite3 so I can embed - but meh - too much code to change :) |
| 01:44.54 | ZX81 | heh same |
| 01:45.55 | ZX81 | should have done a make clean before scp'ing the older version across :) |
| 01:46.25 | hardwire | you sir.. are a dirty man. |
| 01:46.31 | ZX81 | lol yeps |
| 01:46.44 | ZX81 | oh well - will do once its finsihed scping |
| 01:48.48 | ZX81 | oh ffs - using the version from the other machine - same spec h/w etc - crashes too - so, either asterisk or debian etch |
| 01:49.56 | *** join/#asterisk Hadi- (n=Hadi@CPE001310492769-CM001a668ee8b2.cpe.net.cable.rogers.com) |
| 01:50.29 | Hadi- | hello... my question is a little off topic but can someone assist me with setting up a linksys dial plan in the provision template? |
| 01:51.21 | drmessano | I use this (*x.|x.) |
| 01:52.53 | ZX81 | :) |
| 01:53.03 | ZX81 | what if someone wants to dial #123 |
| 01:53.04 | ZX81 | :) |
| 01:53.23 | drmessano | x. |
| 01:53.32 | ZX81 | thought x was just for numbers? |
| 01:53.55 | drmessano | Actually, it is |
| 01:53.57 | ZX81 | (*x,|x.|#x.) |
| 01:53.58 | ZX81 | :) |
| 01:54.01 | ZX81 | but meh |
| 01:54.04 | ZX81 | who dials #123 |
| 01:54.05 | ZX81 | :) |
| 01:54.36 | drmessano | # isn't needed |
| 01:54.40 | ZX81 | nah |
| 01:54.56 | hardwire | ZX81: ME |
| 01:54.59 | hardwire | err .. me |
| 01:55.00 | drmessano | I do need to add something to my dialplans |
| 01:55.04 | WimpMan | Someone wo wants to cancel CF? |
| 01:55.15 | ZX81 | isn't that *numnum |
| 01:55.38 | hardwire | umnumnum |
| 01:55.47 | ZX81 | :) |
| 01:56.06 | drmessano | I need to add to my dialplans a failover if they dont dial in 5 seconds |
| 01:56.16 | drmessano | Since they fast-busy otherwise |
| 01:56.20 | drmessano | A nice Alison message |
| 01:56.52 | WimpMan | Maybe a sponsored ad for some coffee? |
| 01:57.05 | drmessano | Holy crap |
| 01:57.27 | drmessano | Now I know where Google Adsense for PBX's is going to fit in |
| 01:57.32 | drmessano | YES |
| 01:58.09 | drmessano | Ok, time to go write some code so I can sell it to google and retire next week |
| 02:00.05 | *** join/#asterisk mog (n=mog@nat/digium/x-2a252c56b22b9bad) |
| 02:00.05 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:00.16 | De_Mon | I tried to write an extension that begain with a #, all sorts of problems occurred! don't use # |
| 02:01.18 | WimpMan | They didn't behave any different than others for me. |
| 02:01.33 | WimpMan | Would be bad if they did. |
| 02:02.20 | De_Mon | default dial pattern doesn't like it |
| 02:05.52 | [TK]D-Fender | <PROTECTED> |
| 02:05.53 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 02:06.01 | [TK]D-Fender | You can dial whatever you feel like. |
| 02:06.22 | WimpMan | Unless you have a crappy(ly configured) UA. |
| 02:06.57 | *** join/#asterisk Hadi-- (n=Hadi@CPE001310492769-CM001a668ee8b2.cpe.net.cable.rogers.com) |
| 02:07.00 | Hadi-- | ok sorry |
| 02:07.02 | Hadi-- | here is the problem |
| 02:07.20 | Hadi-- | I need to add the following to my XML file so it can be provisioned via tftp |
| 02:07.34 | Hadi-- | <411:14162352332> |
| 02:07.44 | Hadi-- | but I can not use "<" or ">" |
| 02:08.00 | Hadi-- | because its causing an error as thats a code in the XML template |
| 02:08.02 | hardwire | so don't tell your equipment to do it right. |
| 02:08.09 | hardwire | err |
| 02:08.13 | hardwire | so.. yeh.. tell it not to do that |
| 02:08.17 | Hadi-- | well XML has all sorts of < and > in it |
| 02:08.17 | hardwire | what phone? |
| 02:08.27 | Hadi-- | its an ATA PAP2 |
| 02:08.28 | hardwire | lol |
| 02:08.47 | hardwire | why are you telling it to use <411:....> ? |
| 02:08.57 | hardwire | why not 411-..... |
| 02:09.00 | WimpMan | Hadi--: Ever wanted to display < or > on a web page? |
| 02:09.11 | Hadi-- | here is the issue |
| 02:09.12 | Hadi-- | <Dial_Plan_2_>(*xx.|[23456789]11S0|0|[2-9]xx[2-9]xxxxxxS0|[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|011xxxxxxxxxxxx.)</Dial_Plan_2_> |
| 02:09.21 | Hadi-- | thats the line in the XML config |
| 02:09.23 | Hadi-- | on the tftp |
| 02:09.28 | Hadi-- | but I need to add that line |
| 02:09.42 | Hadi-- | <411:14162352332> |
| 02:10.04 | Hadi-- | but b/c of the "<"... its causing an error with the code |
| 02:10.04 | hardwire | walks off |
| 02:10.22 | drmessano | Dude |
| 02:10.27 | drmessano | You dont put < > around it |
| 02:10.29 | drmessano | ( ) |
| 02:11.07 | Hadi-- | hum |
| 02:11.09 | drmessano | and <411:14162352332> would go in [ ] |
| 02:11.17 | Hadi-- | so we can use (411:14162352332) |
| 02:11.24 | drmessano | No, at the ends |
| 02:11.34 | drmessano | hang on |
| 02:12.49 | Hadi-- | ok |
| 02:13.05 | drmessano | Something like that doesnt work ----> (*xx.|<411:14162352332>|[23456789]11S0|0|[2-9]xx[2-9]xxxxxxS0|[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|011xxxxxxxxxxxx.) |
| 02:13.40 | Hadi-- | nope |
| 02:13.47 | Hadi-- | the < > around the 411 |
| 02:13.48 | drmessano | What is the problem? |
| 02:13.56 | Hadi-- | will cause an error and it will not download the xml |
| 02:13.59 | Hadi-- | from the tftp |
| 02:14.40 | drmessano | Oh, hang on |
| 02:14.45 | drmessano | I know the answer |
| 02:15.03 | drmessano | Lemme go find the code.. Was braindead for a moment |
| 02:16.22 | Hadi-- | ok thanks |
| 02:17.04 | frogonwheels | Hadi--: regardless of whether your dialplan is correct, you can use < &rt; or the corresponding unicode version.. |
| 02:17.09 | frogonwheels | > |
| 02:17.11 | frogonwheels | duh |
| 02:17.24 | *** join/#asterisk Alton2 (n=alton@69.45.126.86) |
| 02:18.00 | jameswf-home | my 1411 is '1411' => 1. Dial(SIP/16416418004664411@sip.tollfreegateway.com) [pbx_config] |
| 02:18.06 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 02:18.59 | Hadi-- | pl |
| 02:19.02 | Hadi-- | ok |
| 02:19.03 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 02:19.08 | Hadi-- | so I can have the following |
| 02:19.18 | drmessano | Ok |
| 02:19.27 | drmessano | < and > for the open and close |
| 02:19.29 | hardwire | tollfreegateway.com |
| 02:19.30 | hardwire | ? |
| 02:19.32 | frogonwheels | Hadi--: possibly < > |
| 02:19.41 | drmessano | This is how Linksys spits them out |
| 02:20.10 | jameswf-home | hardwire: yep freeway to goog411 |
| 02:20.14 | hardwire | nice |
| 02:20.21 | Hadi-- | <Dial_Plan_2_>(*xx.|[23456789]11S0|0|[2-9]xx[2-9]xxxxxxS0|[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|011xxxxxxxxxxxx.|<411:14162352332>)</Dial_Plan_2_> |
| 02:20.24 | drmessano | <411:14162352332> |
| 02:20.28 | jameswf-home | then it connects you to anyone free |
| 02:20.30 | drmessano | No |
| 02:20.38 | drmessano | Why do people not read? |
| 02:20.42 | Hadi-- | ok i noticed now |
| 02:20.44 | drmessano | < > |
| 02:20.45 | Hadi-- | ;) |
| 02:20.47 | Hadi-- | lets try |
| 02:21.26 | Hadi-- | brb in a sec |
| 02:21.35 | drmessano | jameswf-home: I use goog411 too |
| 02:21.41 | drmessano | via tollfreegateway |
| 02:21.42 | Alton2 | I keep seeing this unique ID in the log in version 1.4.22: SIP/192.168.50.135-b77211a0 I wonder if this isn't some sort of bug, since calls are kinda disappearing on me. |
| 02:22.24 | hardwire | is it an active channel? |
| 02:22.31 | hardwire | like.. at all times? |
| 02:22.42 | Alton2 | This is on a calling cards system, |
| 02:22.49 | hardwire | what system? |
| 02:22.53 | *** part/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 02:22.55 | Alton2 | the system keeps getting calls with that ID. I rolled back to 1.4.21 and it works fine. |
| 02:23.00 | Alton2 | One I wrote. |
| 02:23.19 | hardwire | cool :) |
| 02:23.31 | Alton2 | Just bringing it up since it seems like something's odd there. Not sure if it's common knowledge. |
| 02:23.41 | hardwire | I'm guessing that caller keeps calling, and never killing the attempt to call you |
| 02:23.49 | hardwire | but that's just me. |
| 02:23.54 | hardwire | somebody smarter should answer |
| 02:24.13 | Alton2 | Well, that's the IP of the routes that's sending the call. However, the characters after the ID shouldn't appear again; each call id should be unique. |
| 02:24.19 | Alton2 | of the router |
| 02:24.35 | file | a channel name is unique at the time it is active, but not unique across all time |
| 02:24.38 | hardwire | I'm unsure of how uniqueid's in asterisk work |
| 02:24.47 | file | it's a combination of IP address and memory address |
| 02:24.49 | hardwire | I thought it was generated against the tech/extension |
| 02:24.52 | hardwire | ah.. ok :) |
| 02:25.05 | file | it is possible for the memory address to get used again, thus it can show up again |
| 02:25.07 | Alton2 | I understand, although I haven't seen it duplicated over and over like this. |
| 02:25.11 | Alton2 | Hmm, interesting. |
| 02:25.35 | hardwire | if you have a relatively efficient and busy app.. it should show up again, often :) |
| 02:25.52 | Alton2 | Well, what happens is that the script appears to die while playing a voice prompt, right in the middle of executing. |
| 02:26.03 | Alton2 | This is using PHPAGI by the way. |
| 02:26.25 | Alton2 | Like I say, not sure why it broke with this particular version. |
| 02:26.29 | jeev | could it be that we have been this way before, i know you dont think that i aint trying! |
| 02:26.38 | hardwire | is your PHP referencing unique id's as a primary key? |
| 02:26.44 | Alton2 | nope |
| 02:26.45 | hardwire | that would bring the call to a halt. |
| 02:26.57 | Alton2 | no, it's as generic as can be. |
| 02:27.02 | jeev | http://www.youtube.com/watch?v=CENJLApAqnU&feature=related = hardwire |
| 02:27.12 | hardwire | damnit jeev.. don't palinroll me |
| 02:27.45 | jeev | lol |
| 02:27.50 | jeev | i dunno why ilike that song so much |
| 02:27.55 | jeev | i want to play it on piano now |
| 02:27.55 | hardwire | is that you? |
| 02:27.57 | hardwire | haha |
| 02:28.01 | hardwire | IT IS ISN'T IT |
| 02:28.06 | hardwire | why are you seranading me?! |
| 02:28.11 | Nugget | http://www.youtube.com/watch?v=07kO9TtHYzQ <-- Terry is back! |
| 02:28.12 | jeev | lol no, it isn't |
| 02:31.40 | *** join/#asterisk Hadi- (n=Hadi@CPE001310492769-CM001a668ee8b2.cpe.net.cable.rogers.com) |
| 02:31.42 | Hadi- | ok |
| 02:31.46 | Hadi- | seems like its not working |
| 02:31.56 | Hadi- | its not downloading the file from the tftp |
| 02:32.05 | Hadi- | this is the line im using |
| 02:32.24 | Hadi- | <Dial_Plan_2_>(*xx.|[23456789]11S0|0|[2-9]xx[2-9]xxxxxxS0|[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|011xxxxxxxxxxxx.|<411:14162352332>)</Dial_Plan_2_> |
| 02:34.09 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 02:35.15 | jameswf-home | heh Palin http://www.hulu.com/watch/39808/saturday-night-live-update-palin-rap |
| 02:36.53 | jeev | palin is wack |
| 02:36.55 | jeev | i'd sex her up though |
| 02:37.06 | jeev | i'm sure a lot of people here wouldn't mind losing their virginity to her. (hahahaha) |
| 02:37.40 | jaytee | she probably uses an oosik as a dildo |
| 02:37.50 | jeev | what's an oosik |
| 02:38.05 | jeev | ew |
| 02:38.12 | jeev | jaytee, how often you rollerblade |
| 02:38.16 | jaytee | the Inuit word for the petrified penis bone from a walrus |
| 02:38.34 | jaytee | jeev, not as often as you masturbate |
| 02:38.36 | jeev | i'm considering going back to school |
| 02:38.39 | jeev | i want to do something crazy |
| 02:40.02 | drmessano | Learn Asterisk.. that would be nuts and unlike you |
| 02:40.24 | *** part/#asterisk Alton2 (n=alton@69.45.126.86) |
| 02:41.56 | jeev | hey there delilah, take your clothes off and sit on my oosik |
| 02:44.03 | tzanger | heh |
| 02:44.09 | jaytee | the Inuit use them to club seals to death |
| 02:44.19 | hardwire | http://gawker.com/5065718/important-news-the-first-minute-of-that-sarah-palin-porn |
| 02:45.14 | hardwire | you couldn't club a seal to death with an oosik |
| 02:45.17 | jaytee | I clicked to watch a YouTube vid called "Who's Nailin Palin" and almost halfway it switches to the Rick Astley video. |
| 02:45.22 | jaytee | hardwire, yes you could |
| 02:45.27 | jeev | heh |
| 02:45.29 | hardwire | jaytee: prove it. |
| 02:45.33 | jaytee | it's about the size of a small baseball bat |
| 02:45.47 | hardwire | jaytee: I've worked in prime seal clubbing territory |
| 02:45.55 | hardwire | the use huge clubs that are 8+ feet long |
| 02:46.01 | hardwire | seals don't go down easy |
| 02:46.13 | jeev | 8+ foot cock? |
| 02:46.17 | hardwire | club |
| 02:46.22 | jaytee | some oosiks are over 5 feet, record one was 8 and fetched over 8000 bucks at auction |
| 02:46.26 | hardwire | the oosiks are just fun little toys found on the beach |
| 02:46.41 | jeev | jaytee, i think that was the same oosik used to FUCK everyone in the US this past month |
| 02:46.44 | hardwire | jaytee: yeh.. I wouldn't club a seal with an $8k oosik |
| 02:46.53 | jaytee | the one I bought is about 3 1/2 feet long |
| 02:47.00 | hardwire | and hows the missus? |
| 02:47.06 | jaytee | lol |
| 02:47.51 | hardwire | snags a banana from the kitchen |
| 02:48.01 | jeev | close your eyes |
| 02:48.11 | jeev | i've been walking around with a 1 ct stone in my pocket since london |
| 02:48.14 | jeev | i forgot to give it back |
| 02:48.18 | jeev | and very expensive pink diamonds |
| 02:48.59 | hardwire | confused. |
| 02:49.18 | jeev | i went to london |
| 02:49.22 | jeev | my friend is a big jeweler |
| 02:49.33 | jaytee | and you're a thief |
| 02:49.39 | jeev | no i'm not |
| 02:49.42 | jeev | you're gay |
| 02:49.44 | jeev | cause you rollerblade |
| 02:49.51 | jeev | i'm a thief cause i still have it in my pocket (cause i held it for him) |
| 02:50.04 | [TK]D-Fender | Jewels : If you can't grow a pair, have one cut for her. |
| 02:50.33 | [TK]D-Fender | ZING! |
| 02:50.38 | jeev | i told my gf, she's so careless |
| 02:50.45 | jeev | i'm gonna give her a 25 cent ring from the gumball machine |
| 02:50.52 | sdaniels | anyone ever get a Cisco 7941g loaded up with SIP and registered to *? |
| 02:51.05 | jaytee | look, whatever my sexual preference is and whatever recreational exercise I happen to like has no bearing on the fact that you are going to end up in a "pound me in the ass" prison if you keep stealing people's gemstones. |
| 02:52.19 | jeev | lol |
| 02:52.20 | jeev | i didn't steal it |
| 02:52.42 | [TK]D-Fender | "Diamonds are forever, cubic zirconium is for Ms Right Now" |
| 02:52.46 | drmessano | sdaniels: Im sure lots have |
| 02:53.09 | jeev | exactly |
| 02:53.40 | jaytee | there's a company in Massachusetts that can make man made diamond gems that can't be told from natural diamonds even under a 100X magnification. |
| 02:54.08 | jeev | yea, they started that shit a while back |
| 02:54.10 | jaytee | their address is a secret, they only have a P.O. box and they've gotten death threats |
| 02:54.15 | jeev | i deal with a lot of athletes |
| 02:54.20 | jeev | i've noticed that athletes are fagballs |
| 02:54.29 | jeev | they pay someone 35k for TRASH gemstones |
| 02:54.38 | jeev | when they could've paid 50k and gotten something like 2000% better quality |
| 02:54.46 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 02:54.53 | jeev | jaytee, seen black hawk down ? |
| 02:54.57 | jaytee | they're really looking more to industrial uses, sheets of diamond or machine parts made of diamond. |
| 02:55.06 | jaytee | jeev, yeah I've got the DVD |
| 02:55.11 | jeev | man, i love that movie |
| 02:55.14 | jeev | i feel like watching it |
| 02:55.17 | jaytee | so? |
| 02:55.18 | jeev | i'm listening to the song at the end |
| 02:55.19 | jeev | :> |
| 02:55.23 | jaytee | ok |
| 02:55.27 | jaytee | I forget the song |
| 02:55.40 | jeev | listen to that shit |
| 02:56.27 | jaytee | I like the song at the end of Shooter |
| 02:56.38 | jeev | dood |
| 02:56.41 | jeev | SHOOTER was a great movie |
| 02:56.44 | jeev | are you gonna accept my dcc? |
| 02:56.48 | jaytee | I did |
| 02:57.04 | jeev | wtf |
| 02:57.04 | hardwire | shooter is a good movie |
| 02:57.07 | jeev | is it coming through? |
| 02:57.19 | hardwire | are you DCC'ing a movie to jaytee? |
| 02:58.23 | jeev | no |
| 02:58.25 | jaytee | jeev, try again. something went haywire at this end |
| 02:58.29 | jeev | whatg is this, 1999 XDCC days ? |
| 02:59.30 | jeev | http://x.jeev.net/dl/ |
| 02:59.33 | jeev | download from there |
| 03:00.34 | jeev | ? |
| 03:00.35 | jaytee | what language is that in? French? Farsi? |
| 03:01.08 | jeev | french i think |
| 03:01.21 | jeev | the black hawk down soundtrack is amazing. |
| 03:02.00 | drmessano | Polls are just a fancy way of systematically predicting what is going to happen |
| 03:02.08 | jeev | get the other song i put there, i should upload the whole album |
| 03:02.13 | jeev | 30 seconds left for still |
| 03:02.32 | drmessano | OMFG |
| 03:02.36 | drmessano | are those words? |
| 03:03.09 | jeev | jaytee, like it ? |
| 03:03.17 | jaytee | yeah |
| 03:03.25 | jaytee | it's really nice |
| 03:03.30 | jeev | how about still |
| 03:03.38 | jaytee | even though I don't know what they're saying |
| 03:04.12 | jaytee | I just started listening to that. it sounds arabic |
| 03:04.43 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:06.33 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:06.50 | jaytee | jeev, did you get the DCC invite? |
| 03:07.56 | jeev | no |
| 03:07.58 | jeev | what's the point |
| 03:08.02 | jaytee | ah crap it timed out |
| 03:08.07 | jeev | it really matters what they're saying? |
| 03:08.12 | jeev | if you dont understand it, you shouldn't listen ? |
| 03:08.14 | [TK]D-Fender | I work in french and can't make more than 2 words out in that piece... |
| 03:08.20 | jaytee | no, I liked it anyways |
| 03:08.32 | jeev | Fender, everyone here speaks english and can't understand anything you say when it comes to asterisk! |
| 03:08.37 | jaytee | just the tonality makes it worth it |
| 03:09.05 | [TK]D-Fender | jeev: Well they're just slow.. this piece is UNINTELLIGIBLE! |
| 03:09.26 | [TK]D-Fender | jeev: I'm jsut waiting for Enya to overdub it |
| 03:09.49 | jeev | fender, that's like 10 years old |
| 03:09.50 | jeev | lol |
| 03:09.55 | jeev | or 6 or whatever |
| 03:09.58 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:10.34 | jeev | jaytee, download the other two songs |
| 03:11.36 | jaytee | hey! I like Enya! ..... oh, wait! I'm gay and rollerblade........never mind. |
| 03:11.43 | jeev | enya sucks |
| 03:12.00 | jeev | people think my fascination for Yanni's Live at the Acropolis == gayism but is not. |
| 03:12.58 | jaytee | is listening to Enya - On Your Shore [21:09 (2%)] |
| 03:12.59 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:13.19 | hardwire | I think Maynord sings like an angel. |
| 03:13.36 | hardwire | a beautiful, bald, bicurious angel. |
| 03:14.15 | jaytee | starts cleaning his rollerblades while viewing his gallery of Clay Aiken pics |
| 03:14.18 | jeev | duno who is that |
| 03:14.20 | jeev | LOL jaytee |
| 03:14.22 | jeev | hardwire, like the song? |
| 03:14.45 | hardwire | meh.. it's not for me |
| 03:14.50 | jeev | awesome song |
| 03:14.59 | jeev | hardwire, i'll look for "drill baby drill" for you |
| 03:15.03 | hardwire | naw |
| 03:15.05 | hardwire | I'm tooling out atm |
| 03:15.11 | hardwire | don't ruin my tool |
| 03:15.22 | jeev | best thing about tonight is that we're not fighting! |
| 03:15.29 | jeev | damn, this secondhand serenade shit is cool |
| 03:16.11 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 03:16.15 | jeev | [08:04pm] [N] SEND request from jaytee (n=jforde05@unaffiliated/jaytee) [0.0.0.199:0] of 10 - Nasty Letter.mp3 (8.08MB). |
| 03:16.15 | jeev | lol |
| 03:16.19 | pcrane | I've got a problem... |
| 03:16.41 | pcrane | serverA and serverB are ment to be connected via a SIP trunk |
| 03:16.50 | pcrane | I can send a call from serverA to serverB |
| 03:16.52 | jaytee | jeev, I got an abort message due to timeout |
| 03:17.06 | jeev | ignore it. i already gave you the link |
| 03:17.07 | pcrane | but when I try send the call back to serverA it doesn't work |
| 03:17.15 | pcrane | I get Failed to authenticate on INVITE |
| 03:17.19 | jaytee | jeev, I was trying to send you a tune |
| 03:17.20 | jeev | debug and pastebin |
| 03:17.25 | pcrane | heh |
| 03:17.28 | jeev | ahh, i dunno man, we have some fucked up proxy |
| 03:17.44 | jaytee | I can't do an http link at my end, sorry. |
| 03:18.10 | jaytee | it's the song from the end of Shooter. Good bluesy guitar and great lyrics |
| 03:18.16 | jeev | oh |
| 03:18.18 | hardwire | pcrane: like jeev said.. pastebin your sip peers for us to look at |
| 03:18.20 | jeev | youtube me |
| 03:18.32 | pcrane | just sorting it out ;) |
| 03:19.50 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 03:20.16 | jaytee | jeev, http://www.youtube.com/watch?v=kwISBR3tMtk |
| 03:20.27 | jaytee | that's not my post, it was already there |
| 03:21.47 | jaytee | I'm really liking "Leave no man behind". cool tune |
| 03:22.40 | pcrane | serverB: http://pastebin.com/m150d95d4 |
| 03:25.48 | pcrane | serverA:http://pastebin.com/m2160440c |
| 03:25.59 | jeev | red alert 3, oct 28 |
| 03:26.09 | pcrane | that has sip debug and the sip peers |
| 03:26.38 | *** join/#asterisk s0lid (n=s0lid@60.52.253.84) |
| 03:27.10 | pcrane | it's probably something really simple that I've forgotten to do |
| 03:28.07 | jeev | dunno\ |
| 03:28.09 | jeev | google it ? |
| 03:28.42 | pcrane | jeev: me? |
| 03:29.04 | pcrane | I've tried google... tried what it suggests, and I've not got anywhere |
| 03:29.38 | jeev | i'm sure fender will see it soon and tell you |
| 03:29.38 | jeev | lol |
| 03:29.39 | jeev | or yell at you |
| 03:29.44 | pcrane | yehah |
| 03:29.48 | pcrane | he'll yelll |
| 03:29.50 | pcrane | :p |
| 03:30.01 | jeev | you should put your config up too |
| 03:31.40 | pcrane | dialplan? |
| 03:32.15 | jeev | SO WHAT, i'm STILL A ROCKSTAR, I GOT MY ROCK MOVES |
| 03:32.15 | jeev | ! |
| 03:32.21 | jeev | i dunno, dial plan and sip |
| 03:32.38 | pcrane | sip is at the bottom of both pastebins |
| 03:32.56 | jeev | ah |
| 03:33.02 | pcrane | ;) |
| 03:33.25 | pcrane | there's probably an option like: justworkdammit=yes that I |
| 03:33.29 | pcrane | m missing |
| 03:33.33 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:33.41 | jeev | so you call serverB from serverA |
| 03:33.41 | jeev | it works |
| 03:33.45 | jeev | you call serverA from B, it doesn't ? |
| 03:34.02 | pcrane | serverA runs 1.2 (and no AMD) |
| 03:34.10 | pcrane | serverB runs 1.4 (and has AMD) |
| 03:34.18 | pcrane | I want to run AMD on the 1.2 stuff... |
| 03:34.45 | jeev | what the hell is AMD |
| 03:34.47 | pcrane | so, I initiate the calls on 1.2, send them to 1.4 (does AMD), and then sends them back to 1.2 |
| 03:34.53 | pcrane | answer machine detection |
| 03:35.07 | jeev | wow dunno |
| 03:35.26 | pcrane | that's why I'm working on this |
| 03:35.28 | hardwire | pcrane: what module does AMD? |
| 03:35.46 | pcrane | the 1.2 machine we're talking about is a fonality PBXtra system |
| 03:35.54 | pcrane | so, I can't touch the code or anything of that |
| 03:36.07 | pcrane | if I think what you're thinking... |
| 03:36.11 | jeev | jaytee, cool... i really dont like guitar shit though |
| 03:36.42 | jeev | too much guitar |
| 03:36.46 | jaytee | each to his own |
| 03:37.39 | Qwell | pcrane: then you're screwed. call them. |
| 03:37.56 | pcrane | how come? |
| 03:38.19 | Qwell | This is #asterisk, not #pbxtra |
| 03:39.30 | pcrane | yea |
| 03:39.31 | pcrane | I know |
| 03:39.38 | pcrane | I'm happy to modify the config files |
| 03:39.56 | pcrane | my original questions still stands though |
| 03:40.03 | pcrane | what am I doing wrong with this setup? |
| 03:40.29 | pcrane | I just elaborated on the reason why... |
| 03:40.33 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 03:40.46 | pcrane | I doubt that the #pbxtra people are going to be able to help me |
| 03:40.48 | Qwell | You can't do it. You already stated why you cna't. |
| 03:40.58 | Qwell | <pcrane> so, I can't touch the code or anything of that |
| 03:41.03 | Qwell | hence - you are screwed. |
| 03:41.07 | pcrane | c code |
| 03:41.11 | pcrane | not config files ;)_ |
| 03:41.13 | pcrane | sorry |
| 03:41.15 | pcrane | my bad |
| 03:41.33 | Qwell | config file isn't going to do much good if the code doesn't exist to use it. |
| 03:41.56 | pcrane | but it's based on asterisk, with a sip.conf |
| 03:42.02 | Qwell | <pcrane> so, I can't touch the code or anything of that |
| 03:42.09 | pcrane | so, why can't I create a sip trunk to somewhere else? |
| 03:42.25 | Qwell | <pcrane> I want to run AMD on the 1.2 stuff... |
| 03:42.34 | pcrane | yep |
| 03:42.36 | pcrane | like this |
| 03:42.38 | Qwell | you can't. |
| 03:43.01 | pcrane | call initiates on 1.2 -> send to 1.4 for amd -> sent back to 1.2 |
| 03:43.09 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 03:43.09 | pcrane | it's the -> that I need the sip trunk for |
| 03:43.14 | Qwell | okay, call them. |
| 03:43.37 | Qwell | Nobody here has any idea what modifications they've made to it. We can't help you. |
| 03:43.49 | pcrane | mmm... |
| 03:43.56 | pcrane | that'll be my problem later on |
| 03:44.13 | pcrane | for the moment though, in the test environment I've got setup is a standard 1.2 asterisk setup |
| 03:44.31 | pcrane | and I'm having problems with a sip trunk between it and a 1.4 setup |
| 03:44.54 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.37.238) |
| 03:45.44 | pcrane | so, the question I posted (what am I doing wrong) is still valid |
| 03:47.17 | pcrane | the error message I'm getting is "failed to authenticate on invite" when the call is sent back to the 1.2 machine |
| 03:47.46 | *** join/#asterisk marc7 (n=marc@S0106001c1024382d.gv.shawcable.net) |
| 03:50.36 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 03:51.52 | [T]ank | does anyone know of an ata that also has a 10/100 switched port on it? I do not want to have to set a switch with the ata and phone for this particular application, so one with just a single ethernet out port would be nice. |
| 03:56.36 | [TK]D-Fender | [T]ank: I've never heard of one... |
| 03:56.49 | [TK]D-Fender | [T]ank: I doubt there is a market to validate it. |
| 03:58.52 | [T]ank | [TK]D-Fender: looks like maybe one of the grandstream handy tones may do it. It has two ethernet ports. Just dont know if that is the intended use of the second port or not... looks like I may take a gamble and check it out. |
| 04:00.05 | jaytee | damn, Sox choked again! |
| 04:00.44 | [TK]D-Fender | [T]ank: No, every ata out there with 2 ports is because tis a router, never switched |
| 04:02.24 | *** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net) |
| 04:07.35 | [T]ank | [TK]D-Fender: would that still give me the same results i am after? just want to plut an ata in where i only have one ethernet jack but still want to plug in a laptop next to it as well. |
| 04:09.18 | [TK]D-Fender | [T]ank: If you feel like NAT-ing the Laptop behind it.... |
| 04:10.28 | pcrane | woot |
| 04:10.31 | pcrane | got it working ;) |
| 04:10.59 | pcrane | I was calling from a SIP phone with caller ID set, when passing the call back to the originating machine, it confused itself |
| 04:11.39 | *** join/#asterisk axisys (n=axisys@117.18.230.216) |
| 04:12.17 | hesco | is an IAX provider necessary to connect two * servers? Is a packet connection without the per minute charges possible? And how would that be wired? |
| 04:13.11 | hesco | ~book |
| 04:13.12 | jbot | somebody said book was 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://tfot.leifmadsen.com or see ~buybook |
| 04:16.39 | SQLDarkly | What is a good solution for hardware timing and that is its only purpose. Rest is all SIP |
| 04:18.17 | jaytee | SQLDarkly, if you're using 1.4 then zaptel's ztdummy driver will give you hardware timing without adding a card. |
| 04:18.28 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 04:18.35 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 04:19.04 | SQLDarkly | ztdummy is off slightly. This is high load so a more reliable solution is needed |
| 04:28.29 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 04:28.59 | *** join/#asterisk freakazoid0223 (n=mattc@68.238.189.158) |
| 04:30.18 | [TK]D-Fender | SQLDarkly: TDM400 with no modules should work |
| 04:30.31 | [TK]D-Fender | SQLDarkly: Actaully get the new base card, TDM410 |
| 04:34.36 | SQLDarkly | Ill check that out. Carried by Digium? |
| 04:34.43 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 04:35.31 | jaytee | www.telephonydepot.com |
| 04:36.49 | jaytee | time for me to get some sleep |
| 04:36.52 | jaytee | nite all |
| 04:40.41 | hesco | so if I configure reciprocal peers in the iax2.conf files of the * servers I want to connect, do I get a straight through connection, without paying an iax2 provider in between? |
| 04:48.34 | hesco | do include => lines work in all the configuration files or only in the extensions.conf file? |
| 04:48.48 | jameswf-home | #include |
| 04:48.52 | aiksa[LV] | and i am back. |
| 04:50.06 | [TK]D-Fender | hesco: Avoid in voicemail.conf |
| 04:51.34 | hesco | that's exactly the one I was eye'ing when I asked that question. Thanks for that warning. |
| 04:52.40 | hesco | does dialplan reload also reload the voicemail configuration as well, if not, how do I do that? |
| 04:55.15 | [TK]D-Fender | hesco: "module reload app_voicemail.so" |
| 04:57.03 | aiksa[LV] | [TK]D-Fender: still there? dont you ever sleep? |
| 05:03.20 | hesco | Thanks, [TK]D-Fender! |
| 05:17.20 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 05:22.14 | drmessano | If I join #include |
| 05:22.19 | drmessano | Do I see all the channels? |
| 05:23.01 | [TK]D-Fender | drmessano: No, HBO is always extra.... |
| 05:23.12 | [TK]D-Fender | aiksa[LV]: Speaking of which.... |
| 05:23.15 | [TK]D-Fender | bedtime.. later all |
| 05:23.46 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 05:29.58 | hesco | is there a way to easily broadcast a single message to multiple voicemail boxes? |
| 05:31.09 | *** join/#asterisk marc7 (n=marc@S0106001c1024382d.gv.shawcable.net) |
| 05:31.54 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 05:49.23 | *** join/#asterisk jameswf-home (n=james@ip72-200-94-120.tc.ph.cox.net) |
| 05:51.34 | *** join/#asterisk chendy (n=chatzill@58.60.178.47) |
| 06:05.16 | *** part/#asterisk Deeewayne (i=dwayne@76.29.245.9) |
| 06:11.35 | hesco | I had loaded sample configuration files to check out what is possible. But I would prefer to start with a clean slate. How can I rebuild skeleton config files without installing the samples, again? |
| 06:15.13 | aiksa[LV] | so any right here, right now is somebody with a knowledge of configuring Astribank ? |
| 06:16.57 | aiksa[LV] | tzafrir_laptop: ping, I was told that I could kindly refer to you for assistance. |
| 06:22.46 | *** join/#asterisk write_erase (n=Olivier@goodgw.m2m-fleet.com) |
| 06:23.21 | write_erase | Hi... Is Asterisk ready for production uses ? Can you tell me the URL for new 1.6 features . Thx |
| 06:23.42 | write_erase | Is Asterisk 1.6 ready for production... sorry . |
| 06:26.54 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:28.47 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 06:50.58 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 06:54.09 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 06:54.58 | joobie | hey guys.. im running "/usr/sbin/asterisk -rx 'sip show peers'" to see which active sip connections are setup to * .. it's showing some old ones in there and saying they are online.. is there a way i can refresh that information? |
| 06:55.13 | joobie | like one of the phones that is coming up online.. i have unplugged it entirely, but it still comes up as online |
| 06:56.21 | aiksa[LV] | joobie: you mean as qualify column? |
| 06:56.58 | *** join/#asterisk steliosk (n=Stelios@91.140.105.61) |
| 06:57.34 | joobie | aiksa[LV], not sure what you mean by qualify column.. I'm trying to view which handsets have an active sip connection to * |
| 06:57.59 | WimpMan | No, which ones have registered. |
| 06:58.01 | joobie | I thought the above command was the proces to do this, but it is showing a phone I have unplugged from the network as online |
| 06:58.19 | joobie | WimpMan, how can i update that though.. because one of the phones in the list is no longer registered |
| 06:58.47 | WimpMan | There is no such thing as an active connection. |
| 06:59.02 | WimpMan | Wait until the registration expires. |
| 06:59.21 | WimpMan | Or enable qualify. |
| 06:59.49 | WimpMan | That will check reacability regularly. |
| 07:01.18 | joobie | WimpMan, how frequently will it check the phone if enabled? |
| 07:01.45 | WimpMan | Configurable. |
| 07:02.11 | WimpMan | I think the default is 60s if ok, 10s if not ok. |
| 07:02.20 | joobie | ahh k |
| 07:02.37 | joobie | Wimp, sepereate to the above (btw thanks for that), is there a way i can force asterisk to refresh that register cache? |
| 07:02.41 | joobie | just curious.. |
| 07:02.51 | joobie | or is qualify / waiting for register to expire the only way? |
| 07:03.11 | WimpMan | What it "refresh" meant to do? |
| 07:03.27 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:03.42 | WimpMan | As I said: There is no active connection. The phone tells * where it can be found. |
| 07:03.44 | joobie | update the table that 'sip show peers' reads from |
| 07:04.13 | WimpMan | And unless the last time it did so exeeds a certain amount of time, * will try to connet it there in case of a call. |
| 07:04.49 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 07:05.06 | WimpMan | So you usually don't know if the Phones really are reachable. But that's where qualify goes. |
| 07:06.00 | yang | aiksa[LV]: while you are waiting for tzafrir's assistance you might want to check the Astribank guide - http://www.voip-info.org/wiki/view/Xorcom+Astribank |
| 07:06.26 | joobie | WimpMan, I see |
| 07:06.31 | joobie | thanks :) |
| 07:07.49 | WimpMan | Same concept as with gsm :-) |
| 07:07.50 | tengulre | I want to write a softphone on windows platform, server using asterisk ,anybody can give me some tips? |
| 07:11.28 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 07:15.41 | *** join/#asterisk chendy (n=chatzill@58.60.178.47) |
| 07:18.07 | *** join/#asterisk steliosk (n=Stelios@79.107.29.93) |
| 07:23.21 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 07:26.16 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 07:29.04 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.151.162) |
| 07:30.03 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 07:32.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:35.05 | aiksa[LV] | yang: I gave i tgalnce it looks like a stripped down version of xorcom online manual and troubleshoot guide, which I have reread for a number of times by now. |
| 07:35.36 | aiksa[LV] | but i`ll go through it again |
| 07:35.37 | *** join/#asterisk ant (n=ant@67-207-113-226.static.wiline.com) |
| 07:36.14 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 07:36.39 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 07:36.40 | ant | how do I get the caller id to follow once the call has been handed off from the receptionist? |
| 07:36.50 | ant | I'm stumped. |
| 07:36.55 | ant | (and it's late) |
| 07:38.53 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:43.30 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 07:43.45 | *** join/#asterisk C4away (n=DJpyro@66.185.107.193) |
| 07:44.43 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-a105b5308a020512) |
| 07:45.08 | C4away | any documentation on cdr_pgsql setup in 1.6? |
| 07:45.27 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 07:46.54 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 07:48.15 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:51.08 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 07:51.36 | hesco | C4away: mine has only the connection parameters. What do you need? the table creation statements? |
| 07:51.56 | C4away | well |
| 07:52.03 | C4away | in 1.4 you have to do something like ./configure --with-postgres=<dir where postgresql is installed> |
| 07:52.09 | C4away | is that required? |
| 07:52.15 | C4away | in 1.6? |
| 07:52.31 | C4away | do I need to specify the folder where psql is or where the server is installed? |
| 07:52.55 | C4away | (on my setup the postgresql server will be on another machine, however the postgresql client is installed locally) |
| 07:53.10 | C4away | does it use the socket/buffer method by default? |
| 07:53.12 | C4away | etc |
| 07:53.25 | C4away | lots of questions, not many answers |
| 07:53.32 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 07:54.06 | C4away | the difference between 1.4 and 1.6 is not very evident at this point... i'm sure as time goes on the documentation will flow forth, however I was hoping to have this implemented tonight |
| 07:54.35 | mvanbaak | C4away: ./configure should find the pgsql libs on your system. only when the pgsql headers are not in your ldconfig you have to specify where it is |
| 07:54.45 | C4away | ah good |
| 07:54.56 | C4away | ok, that's taken care of, I have postgresql-devel installed |
| 07:54.58 | mvanbaak | C4away: in the cdr_pgsql.conf you can specify how to connect to your pgsql server |
| 07:55.08 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:55.16 | mvanbaak | C4away: if you specify an ip and a port there it will use tcp |
| 07:55.23 | C4away | I am also upgrading from 1.6.0-beta9 to 1.6.0.1 in this process |
| 07:55.31 | *** join/#asterisk hi365_m (n=hi365@bzq-79-182-238-229.red.bezeqint.net) |
| 07:55.44 | C4away | so a recompile is needed (I didn't have the psql client on it before when I installed it) |
| 07:56.04 | C4away | so if I recompile/upgrade asterisk and set up the cdr_psql.conf file I should be good? |
| 07:56.15 | mvanbaak | probably |
| 07:56.20 | mvanbaak | what you can do to be sure is: |
| 07:56.30 | mvanbaak | ./configure && make menuselect |
| 07:56.47 | C4away | then enable the cdr_psql.so module? |
| 07:56.54 | C4away | make sure it's not greyed out with dependancies |
| 07:57.02 | mvanbaak | there you can check if the cdr_psql module is enabled |
| 07:57.02 | C4away | I can do that |
| 07:57.05 | mvanbaak | indeed |
| 07:57.08 | *** join/#asterisk X-Rob (n=rob-x@CPE-124-187-2-200.qld.bigpond.net.au) |
| 07:57.09 | C4away | thanks |
| 07:57.20 | mvanbaak | good luck |
| 07:57.57 | C4away | oh wow, menuselect on 1.6 is completely different |
| 07:58.18 | C4away | yep psql is enabled |
| 07:58.19 | C4away | =) |
| 07:58.27 | C4away | much appreciated |
| 08:08.35 | *** join/#asterisk X-Rob (n=rob-x@CPE-124-187-2-200.qld.bigpond.net.au) |
| 08:09.40 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
| 08:11.32 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:15.14 | phpboy | heya ll |
| 08:15.42 | mvanbaak | hey phpboy |
| 08:15.49 | phpboy | is it possible to supress the log in queue_log when using AddQueueMember? |
| 08:20.44 | *** join/#asterisk astrOdz (n=astrOdz@119.111.96.121) |
| 08:22.44 | C4away | hmm |
| 08:22.51 | *** join/#asterisk miloux (n=miloux@213.88.194.123) |
| 08:23.00 | C4away | so I should build the table the same way that I would for 1.4 then? |
| 08:23.07 | C4away | for 1.6 cdr_pgsql |
| 08:23.27 | *** join/#asterisk mandh (n=mandh@74.85.13.25) |
| 08:23.53 | *** join/#asterisk miloux (n=miloux@213.88.194.123) |
| 08:24.56 | *** join/#asterisk qdk (n=qdk@85.235.253.141) |
| 08:25.04 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 08:28.53 | joobie | guys anyone able to help debug a "No matching peer found" error? just added a user to sip.conf and it's not registering... |
| 08:30.15 | *** join/#asterisk zydoon (n=zydoon@41.225.156.138) |
| 08:30.32 | *** part/#asterisk zydoon (n=zydoon@41.225.156.138) |
| 08:33.02 | *** join/#asterisk X-Rob (n=rob-x@CPE-124-179-164-139.qld.bigpond.net.au) |
| 08:45.58 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 08:46.24 | yang | aiksa[LV]: I think that you can get some xorcom assistance now |
| 08:46.55 | tzafrir_laptop | yang, already doing so |
| 08:47.02 | yang | great |
| 08:47.56 | *** join/#asterisk RachedTN (n=virtex5@213.150.163.105) |
| 09:01.37 | *** join/#asterisk steliosk (n=Stelios@79.107.29.93) |
| 09:08.04 | *** join/#asterisk ganeshix (n=chatzill@cpe-24-29-46-121.nycap.res.rr.com) |
| 09:09.20 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 09:10.13 | joobie | guys im getting this error.. chan_sip.c:8507 check_auth: username mismatch, have <1232>, digest has <user1> |
| 09:10.36 | joobie | i've set [1232] to username=1232 .. also phone username is 1232 |
| 09:21.37 | mort_gib | joobie: Check it again |
| 09:23.04 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 09:24.50 | joobie | mort_gib, i have |
| 09:31.58 | astrOdz | hey does anyone know a good way to test call quality if you have many extensions |
| 09:34.25 | mort_gib | ast0dz: Ask your users :-) -You would be more interested in CPU/RAM/Network benchmarks. At least I am :-) |
| 09:34.45 | joobie | check_auth: username mismatch, have <1232>, digest has <user1> |
| 09:34.56 | joobie | does digest stand for the handset or the asterisk box? |
| 09:36.32 | mort_gib | handset is what you want to check.. I use Snoms and they are a bit odd at first |
| 09:37.47 | mort_gib | They have a setting "account" and "Authentication Username" that has to be the same (I think) |
| 09:38.12 | joobie | mort_gib, i have a polycom 330 |
| 09:38.17 | joobie | it just has one username setting... |
| 09:40.04 | mort_gib | Ok, I'm not an expert on Polycoms... I use the Soundstation 650 from time to time, and I have never had issues other than misspelling username/pwd |
| 09:40.49 | joobie | ahh |
| 09:40.52 | joobie | i just did a tcpdump |
| 09:40.53 | mort_gib | Apparently I'm demented since I have up to now used the Webinterface to configure them with....-sigh- |
| 09:41.57 | joobie | i can see in the hex the phone is sending Digest.username="user1" |
| 09:41.57 | mort_gib | Did you do this via a provisioning tool or via the webinterface?? |
| 09:42.11 | mort_gib | On the 650, under the "lines" you can set the account! |
| 09:42.20 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 09:44.17 | joobie | mort_gib, i set it via the phone itself |
| 09:44.26 | joobie | just jumped into web config and removed the proxy to see if thi helps |
| 09:44.28 | joobie | it's rebooting atm |
| 09:44.36 | joobie | ner didnt help |
| 09:45.04 | mort_gib | Ok, you have to try and find that username somewhere |
| 09:45.54 | *** join/#asterisk rrrobert (n=rrrobert@202.125.156.122) |
| 09:46.01 | rrrobert | hello folks. |
| 09:46.08 | joobie | ahh bingo |
| 09:46.11 | joobie | it was set in the web config |
| 09:46.14 | joobie | i see it :P |
| 09:46.14 | joobie | sec |
| 09:46.16 | joobie | removing |
| 09:46.20 | mort_gib | :-) thereyougo |
| 09:46.24 | rrrobert | I need a DIGIUM PRI card in pakistan |
| 09:46.33 | *** join/#asterisk Vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:46.44 | joobie | i think this will work |
| 09:46.49 | joobie | drum roll |
| 09:47.24 | joobie | bingo |
| 09:47.25 | joobie | it worked :) |
| 09:47.26 | joobie | cheers mort |
| 09:47.35 | joobie | i forgot i set a few settings in the webinterface |
| 09:48.01 | mort_gib | :-) No problem |
| 09:48.29 | mort_gib | These things are always simple and mostly occur because you are in a hurry to get the phone to work ;-) |
| 09:51.50 | *** join/#asterisk Karlitoo (n=serserve@213.137.110.67) |
| 09:53.01 | Karlitoo | morning all |
| 09:54.04 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:54.34 | aiksa[LV] | ~pb |
| 09:54.35 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 09:54.41 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.193) |
| 09:54.47 | C4colo | hmm |
| 09:54.59 | C4colo | is there a bug in the cdr_pgsql.so module? |
| 09:55.06 | C4colo | [Oct 20 03:52:17] ERROR[3687]: cdr_pgsql.c:321 pgsql_log: cdr_pgsql: Reason: ERROR: syntax error at or near ")" at character 17 |
| 09:55.23 | C4colo | why would there be a syntax error in the sql sent by the asterisk server to the postgresql database? |
| 09:55.58 | dominic1 | Hello guys, I have a beronet e1 card and want to migrate to dahdi. Can anybody tell me how I can use the misdn driver with dahdi? I heared that it's possible to use misdn cards with dahdi... I didn't find any further informations yet... |
| 10:02.27 | *** part/#asterisk salzh (n=Administ@58.247.195.160) |
| 10:07.36 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 10:09.29 | *** join/#asterisk tmjb (n=tane@212.200.239.230) |
| 10:16.01 | C4colo | ok |
| 10:16.08 | C4colo | done some pcap stuff and found this out |
| 10:16.13 | C4colo | the query it is trying to send is thus: |
| 10:16.31 | C4colo | INSERT INTO cdr ) VALUES ) |
| 10:16.36 | ana_micho | Hi all, I need to know if there is a way to force asterisk to send OK message to the peer gateway after receiving Session Progress from it |
| 10:20.39 | joobie | guys anyone from AU here? just wondering what a good ISP for broadband would be in AU to route voip and data through the same link? |
| 10:20.47 | joobie | it's for a small office.. so finding it hard to warrant a seperate line for voip |
| 10:21.06 | joobie | i was thinking maybe there would be a few AU isps that do QoS for voip traffic and that would be a good one to use? |
| 10:21.53 | JT | qos no idea, try doing it yourself? |
| 10:22.14 | *** join/#asterisk DarkRift (n=dark@65.92.171.88) |
| 10:22.14 | mort_gib | joobie: Not from Au, but consider checking jitter carefully and possibly use a good Firewall to do QoS, like OpenBSD's PF |
| 10:22.30 | mort_gib | Latency/Jitter |
| 10:22.36 | joobie | JT, i can do QOS myself in the LAN.. but once it leaves my LAN and goes to the ISP i think the ISP needs to do QoS |
| 10:22.47 | tmjb | is it possible to use Record(/var/lib/asterisk/sounds/recording:ulaw) in multiple formats say gsm,wav alaw? |
| 10:23.12 | mort_gib | IPV6, yes that would be nice, but I don't think you can get that... |
| 10:23.15 | JT | joobie: good luck with that, don't think anyone really does that |
| 10:23.15 | joobie | mort_gib, how to check latency/jitter? just ping tests? |
| 10:23.30 | mort_gib | Mostly that's called MPLS :-) |
| 10:23.31 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 10:23.50 | joobie | www.tpg.com.au do .. like they offer adsl connection with voip.. i presume that is QoS'ed |
| 10:24.02 | joobie | it's like a standard ADSL connection.. btu they brand that you can run voip over it |
| 10:24.07 | JT | tpg is junk |
| 10:24.11 | mort_gib | Yes, run a test that pings some 4-6 times every half hour for a few days |
| 10:24.13 | joobie | only difference i can see is the branding.. and the cost is a bit more |
| 10:24.17 | JT | it's worse than a decent isp with no qos |
| 10:24.37 | joobie | mort_gib, cheers :) |
| 10:24.42 | joobie | hehe JT |
| 10:25.12 | mort_gib | And even if you provider is great, you need a stable connection to your VOIP provider, which can be a challenge as traffic is bound to leave your ISP's network at some point in time |
| 10:25.30 | joobie | yea |
| 10:25.48 | joobie | what im not sure about is when downloading happens.. pings jump up |
| 10:26.01 | joobie | even with QoS i presume a download would increase latency |
| 10:26.41 | JT | unless you are getting a symmetrical connection, you should be much more concerned about uploads |
| 10:27.23 | joobie | how much b/w is needed for one call on sip? |
| 10:27.32 | joobie | JT, wasnt thinking to get symmetrical |
| 10:27.47 | joobie | i was thinking more so an adsl2+ conn |
| 10:27.56 | JT | joobie: using what codec? |
| 10:28.53 | joobie | ulaw |
| 10:29.02 | joobie | though that brings me to the next Q.. is ulaw the best codec to be using? |
| 10:29.09 | JT | no, alaw is |
| 10:29.13 | joobie | oh |
| 10:29.13 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 10:29.15 | JT | pointless to use ulaw in australia |
| 10:29.21 | joobie | how come JT |
| 10:29.31 | yang | ulaw = american ISDN standard |
| 10:29.40 | JT | alaw is the standard everywhere outside north america and japan |
| 10:30.06 | joobie | ahh |
| 10:30.29 | joobie | ill have to do some reading into the technical differences |
| 10:30.36 | joobie | i just went with ulaw because my sip provider suggested it... |
| 10:30.44 | JT | just different companding laws |
| 10:30.48 | JT | what provider? |
| 10:30.54 | joobie | comvergence.com.au |
| 10:31.05 | joobie | who do you use JT? |
| 10:31.31 | JT | for my own use, pennytel |
| 10:31.36 | JT | cheapest by far |
| 10:32.06 | joobie | good quality / reliability? |
| 10:32.15 | joobie | one of the guys in the office today suggested pennytel too |
| 10:32.15 | JT | generally |
| 10:32.54 | slingr | so |
| 10:33.10 | yang | hi Rico29 |
| 10:33.11 | joobie | neway JT.. how much b/w required for alaw if i switch over? |
| 10:33.20 | slingr | ulaw-audio = NTSC-video / alaw-audio = PAL-video? |
| 10:33.22 | yang | joobie: the same as for ulaw |
| 10:33.30 | JT | at 20ms packetisation, 85kbit/s over sip |
| 10:33.30 | slingr | kinda thing? |
| 10:33.31 | yang | joobie: 64kbit |
| 10:33.43 | JT | lol 64kbit/s is only the codec |
| 10:33.49 | JT | doesn't include voip overhead |
| 10:33.49 | yang | yes |
| 10:33.55 | joobie | 64kbit up and down each direction?? |
| 10:34.05 | JT | no. |
| 10:34.07 | JT | 85kbit/s |
| 10:34.08 | JT | total |
| 10:34.21 | JT | 85kbit/s up, 85kbit/s down |
| 10:34.28 | slingr | so a 512kbit upload should handle 4 channels of ulaw? |
| 10:34.38 | JT | right |
| 10:34.43 | joobie | damn |
| 10:34.47 | joobie | that could be a problem |
| 10:35.05 | JT | 5 or 6 channels |
| 10:35.27 | yang | joobie: you can try the free gsm codec or speex codec, or purchase g.729 from digium its 10 usd per channel |
| 10:35.29 | petererer | a-law is not a codec, is it |
| 10:35.44 | yang | gsm&speex use less bandwidth |
| 10:36.14 | JT | petererer: it is |
| 10:36.23 | JT | petererer: G.711-A-law |
| 10:36.38 | petererer | g.711 is the codec, a-law is signal compression and expending algorhythm |
| 10:36.45 | petererer | *expanding |
| 10:36.59 | *** join/#asterisk synchris (n=synchris@athedsl-220386.home.otenet.gr) |
| 10:37.00 | JT | either way, goes hand in hand |
| 10:37.26 | petererer | mmm, both combined makes sense |
| 10:37.40 | joobie | hmm |
| 10:37.40 | JT | g.711 specifies either companding law |
| 10:37.56 | joobie | is g729 the same as g729a? |
| 10:37.57 | petererer | algorithm :o |
| 10:38.01 | joobie | my sip proivder says they support the a version |
| 10:38.21 | petererer | http://en.wikipedia.org/wiki/G.729a |
| 10:38.22 | ana_micho | Hi all, Is there a way to force asterisk toi send OK message to an INVITE sent from a peer? |
| 10:39.21 | joobie | thanks petererer |
| 10:39.35 | tmjb | is it possible to use Record(/var/lib/asterisk/sounds/recording:ulaw) in multiple formats say at once gsm,wav alaw? |
| 10:39.55 | joobie | man-o-man.. that 85kbit/s speed really blows. |
| 10:40.09 | joobie | pretty much needa nother internet connection dedicated |
| 10:40.17 | JT | get out of the dial up relm :P |
| 10:40.32 | JT | if it's important, spend money |
| 10:40.38 | JT | ie. business |
| 10:40.42 | JT | otherwise be stigney |
| 10:41.05 | petererer | mmm, 85kbit/s is quite low by today's standards |
| 10:41.12 | yang | joobie: did you see what i wrote ? gsm is working at 13 kbit/s - http://www.voip-info.org/wiki-Codecs |
| 10:41.27 | JT | gsm sounds like shit compared to g.711 though |
| 10:41.28 | petererer | i suppose if you have multiple calls... |
| 10:41.39 | joobie | yang, but is gsm quality just as good? |
| 10:41.42 | joobie | ya.. |
| 10:41.46 | joobie | need good quality calls yang |
| 10:42.06 | yang | joobie: well its not as good as alaw/ulaw but it still works |
| 10:42.27 | yang | speex is probably better |
| 10:42.28 | joobie | petererer, in australia we have 256k uplink for standard adsl connections :/ |
| 10:42.39 | tzafrir_laptop | if you want good quality and rate, try speex |
| 10:42.49 | JT | there is no standard adsl connection, heh |
| 10:43.00 | petererer | so 85 kbit/s fits easily... for 2 calls :o |
| 10:43.03 | tzafrir_laptop | costs with more CPU time, but will probably be worth it |
| 10:43.07 | JT | problem with speex is finding a voip provider that supports it |
| 10:43.14 | joobie | lol petererer |
| 10:44.28 | joobie | how does the g729 pay-per-channel arrangement work? |
| 10:45.07 | joobie | ties into an asterisk module or sumthen? |
| 10:45.13 | JT | you pay for the patent licensing cost |
| 10:45.13 | yang | joobie: you go to digium site and order it, the rest i dont know |
| 10:45.25 | JT | it ties to your NIC mac address |
| 10:45.31 | joobie | but then i suppose your handsets will have issues with g729 if it's a paid for protocol |
| 10:45.37 | joobie | like i have polycom 330's.. |
| 10:45.48 | joobie | ahhhh |
| 10:46.20 | yang | joobie: most handsets support g729 |
| 10:46.23 | phpboy | hi |
| 10:46.32 | JT | all ip phone manufacturers pay for g.729 licenses otherwise no-one will buy their phones. |
| 10:46.38 | yang | joobie: but make sure you check about yours first |
| 10:46.42 | JT | often they buy g.723 licenses too |
| 10:47.20 | joobie | ahh |
| 10:47.33 | petererer | hmm, my phones use g.711a, but then that's a lan |
| 10:47.39 | joobie | that sucks that they charge for the codec :/ |
| 10:47.54 | JT | oh no, a whole 10 bucks, what will i ever do |
| 10:48.15 | joobie | is it a once off? |
| 10:48.49 | yang | tzafrir_laptop: can you confirm if FreePBX really has g729 allready in the package, or does it needs to be purchased extra (maybe the trixbox FreePBX distribution has one allready in) ? |
| 10:48.52 | JT | yes |
| 10:49.07 | JT | you need to pay for it |
| 10:49.18 | tzafrir_laptop | freepbx? not that I nkow of |
| 10:49.22 | tzafrir_laptop | maybe trixbox does |
| 10:49.23 | JT | why would a free product have a licensed product that costs money in it? |
| 10:49.28 | JT | doesn't make business sense |
| 10:49.38 | tzafrir_laptop | but only the codec itself, not the license |
| 10:49.59 | joobie | hmm |
| 10:50.03 | joobie | it looks like a once off license |
| 10:50.05 | joobie | which aint bad.. |
| 10:50.15 | yang | yes, a colleague said that he can use it on freepbx and that he hasn't paid for it, but he uses trixbox |
| 10:50.52 | yang | I am waiting for the g729 order to be processed today :) |
| 10:51.16 | joobie | yang.. it is a once-off license right? |
| 10:51.25 | yang | joobie: g729 is at 8kbps |
| 10:51.33 | yang | joobie: i think so yes |
| 10:51.41 | joobie | yaa |
| 10:51.43 | JT | not including overhead |
| 10:51.43 | joobie | it's worth it then |
| 10:51.46 | joobie | for once off |
| 10:51.51 | JT | it's closer to 25kbit/s with overhead |
| 10:51.55 | joobie | ahh |
| 10:51.57 | joobie | thanks JT |
| 10:52.00 | yang | JT: i always forget overhead |
| 10:53.19 | yang | joobie: its being said that the quality is similar to alaw/ulaw |
| 10:53.42 | JT | it's good considering how compressed it is |
| 10:53.49 | yang | joobie: so for adsl i think it must be very good |
| 10:53.50 | JT | but i can always tell if my call goes over g.729 |
| 10:54.07 | JT | and i think "stingey company" i hear them using g.729 |
| 10:54.17 | JT | if i hear a company i call using it... |
| 10:54.29 | yang | What kind of codec is being in use when the call leaves asterisk and travels over telecom lines? |
| 10:54.37 | JT | g.711 |
| 10:54.44 | joobie | JT, so it's noticable the quality ya? hmm.. |
| 10:54.49 | joobie | is it bad? |
| 10:54.54 | JT | it's not bad |
| 10:54.58 | JT | it's just not the same |
| 10:55.03 | joobie | i hear ya |
| 10:55.06 | joobie | hmm |
| 10:55.25 | yang | JT: but its probably the closest you can get with the compressed codecs? |
| 10:55.43 | JT | g.726 is much closer |
| 10:55.54 | JT | but it hates packet loss or jitter of any sort |
| 10:56.19 | yang | joobie: well I have tested GSM on one of my phones and it sounds kinda strange, I was also loosing many packets so i can't tell for sure how it goes |
| 10:56.48 | joobie | ahh |
| 10:56.54 | joobie | i think i will just have to test both |
| 10:57.01 | joobie | for 10$US it's reasonable |
| 10:57.10 | joobie | i will have to forgo nandos for a day |
| 10:57.11 | joobie | :P |
| 10:57.11 | JT | you don't need to pay to test |
| 10:57.19 | JT | connect ip phone direct to voip provider |
| 10:57.25 | joobie | ahh good idea |
| 10:57.30 | joobie | thanks JT |
| 10:57.36 | joobie | my voip provider supports it |
| 10:57.40 | joobie | will try with them |
| 10:57.50 | joobie | nandos will be happy |
| 10:58.51 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 10:59.06 | JT | anyway, if you don't use call recording, or have any IVRs/playback etc, just straight calls, asterisk does no transcoding, so no licensing requirement |
| 10:59.09 | joobie | JT, do you know much about DID or obtaining multiple indial numbers in AU ? .. that is, stable ways of getting pstn numbers forwarded to your * box? |
| 10:59.28 | JT | the most stable way and highest quality is always to get a PRI |
| 10:59.50 | aiksa[LV] | JT: and probably the most expensive |
| 10:59.54 | joobie | nod.. |
| 11:00.01 | joobie | PRI sorta defeats the cost saving of voip |
| 11:00.13 | JT | aiksa[LV]: not that expensive to receive calls |
| 11:00.20 | JT | PRIs are very cheap in australia |
| 11:00.24 | JT | much cheaper than in theUS |
| 11:00.37 | joobie | PRI is 10 lines right? |
| 11:00.41 | aiksa[LV] | JT ok, here we would have to go for at least 250$ a month for a line |
| 11:00.47 | aiksa[LV] | joobie: 31 channels |
| 11:00.49 | JT | 10, 20 or 30 in aust |
| 11:00.52 | yang | joobie: I can recommend voip.ms for DID numbers but I think that they only have USA&UK POPs would make a large latency to australia |
| 11:01.00 | JT | fullPRI E1 is 30 channels |
| 11:01.26 | JT | aiksa[LV]: only 30 voice channels. |
| 11:01.43 | joobie | yang, cheers.. yea probably would be some funky latency.. but i was after more so AU did providers |
| 11:01.55 | aiksa[LV] | well 1 is for signaling |
| 11:02.08 | JT | if you want to own the DID, you need a PRI |
| 11:02.18 | aiksa[LV] | and not always 30 voice chans. |
| 11:02.29 | joobie | JT, so PRI with a 100-indial? |
| 11:02.38 | JT | well, there might be a couple of business voip providers that give you proper DID ownership |
| 11:02.49 | JT | 30 or less chs, aiksa[LV] |
| 11:02.57 | aiksa[LV] | ok |
| 11:03.00 | JT | joobie: that would be my plan |
| 11:03.04 | aiksa[LV] | not more than 30 then |
| 11:03.33 | JT | 30 B channels, 1 D channel, 1 timeslot reserved for superframe sync |
| 11:04.04 | joobie | JT.. 30 PRI would be hundreds, if not over a grand a month |
| 11:04.15 | joobie | 100-indial is 30$ .. which is good:P |
| 11:04.22 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:04.24 | JT | joobie: in northern alaska maybe |
| 11:04.25 | joobie | but the PRI would cost an arm and a leg just for the 30 line rental |
| 11:04.34 | joobie | hmm |
| 11:04.37 | JT | 10 channels is $165 + $33 |
| 11:04.49 | JT | 33 for 100number range |
| 11:04.49 | joobie | eh.. with telstra? |
| 11:04.54 | JT | with Primus |
| 11:04.58 | joobie | ahh |
| 11:05.02 | JT | telstra, do you like pain? |
| 11:05.10 | joobie | that's what i was comparing prices with |
| 11:05.14 | joobie | we got a quote with telstra |
| 11:05.45 | JT | optus is $200 + $30 or so |
| 11:05.59 | JT | verizon is even cheaper but dunno if they do less than 30ch, and their install charges were a killer last i checked |
| 11:06.00 | joobie | damn that is cheap. |
| 11:06.56 | joobie | JT, can you recommend a card to use for a 30PRI ? |
| 11:07.17 | joobie | i havent played with isdn in asterisk.. not sure what a good card is for that purpose |
| 11:07.26 | *** join/#asterisk neuwald (n=neuwald@200.252.126.130) |
| 11:07.30 | JT | sangoma a101d |
| 11:07.33 | joobie | i'm seriously considering a 100 indial and 30PRI given those primus / optus costs you quoted.. I was turned off when i heard telstra originally |
| 11:07.45 | neuwald | hi folks. I have odbc running ok (odbc show it´s ok). now, how i get informations from database in dialplan? |
| 11:07.49 | JT | but if you plan to use freepbx/trix, it sometimes comes with shit sangoma drivers |
| 11:07.59 | *** join/#asterisk ElSonico (n=tav@MKCCCLXXXVIII.gprs.saunalahti.fi) |
| 11:08.07 | JT | joobie: primus do free install on a 24 month contract |
| 11:08.10 | JT | same for optus i think |
| 11:08.46 | joobie | JT, i have servers in the primus datacenter.. and two of the guys i work with used to work in the NOC there |
| 11:08.56 | joobie | good chance we can strike a decent deal with them |
| 11:08.58 | JT | 55king st? |
| 11:09.03 | joobie | nod |
| 11:09.09 | joobie | well one worked at the NOC |
| 11:09.11 | JT | pretty expensive datacentre |
| 11:09.17 | joobie | the other worked in level 2 500 collins |
| 11:09.24 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:09.31 | *** join/#asterisk oej (n=olle@193.136.47.146) |
| 11:09.44 | JT | if you want a decent datacentre, look at sydney or brisbane :P |
| 11:09.57 | joobie | my voip tails will all be melb based tho |
| 11:10.03 | joobie | wanted to try keep the * server local |
| 11:10.18 | JT | fair enough |
| 11:10.58 | JT | might want to take a look at micron21 |
| 11:11.48 | joobie | ew.. they run plesk |
| 11:11.57 | *** join/#asterisk MrNaz (n=mrnaz@114.128.61.218) |
| 11:12.06 | JT | who cares? |
| 11:12.21 | JT | plesk won't run on a co-loed server unless you install it |
| 11:13.36 | joobie | their prices are decent |
| 11:14.37 | JT | at the moment they have a pretty cosy little facility out in the east of melb |
| 11:14.48 | JT | but it has ridiculous levels of connectivity to 55king and 525 collins |
| 11:15.00 | JT | they build their own redundant fibre network |
| 11:15.06 | joobie | 525 collins is telstra ya? |
| 11:15.12 | joobie | nice |
| 11:15.22 | JT | securiton dc |
| 11:15.27 | JT | don't think that's telstra |
| 11:15.31 | joobie | ahh |
| 11:15.41 | joobie | 525 collins sounds around teh exchange building |
| 11:15.52 | JT | might be |
| 11:15.54 | yang | JT: what did you mean that for normal calls asterisk doesn't do transcoding so G.729 can be freely used ? How does it get enabled for the free use? |
| 11:16.16 | JT | yang: if the asterisk box has no reason to transcode, no licence instances are used |
| 11:16.20 | neuwald | Hi All. I have ODBC up and running, as you can see here: http://www.pastebin.ca/1231760 - now, how I get informations from ODBC/database in dialplan? |
| 11:16.40 | JT | g.729 > asterisk > g.729 == no reason to transcode |
| 11:16.48 | phpboy | :( |
| 11:17.00 | phpboy | I'm having a VERY hard time getting rid of chan_agent :( |
| 11:17.04 | yang | JT: oh, so just the uplink needs to stream via g729 then |
| 11:17.09 | JT | yes |
| 11:17.27 | phpboy | I cannot find/write a dial plan that works exactly the same as chan_agent does :( |
| 11:18.59 | phpboy | how do I go about deleting the following entry out of the asterisk DB:- /Queue/PersistentMembers/onecellControl : Local/2001}@agents/n;0;0;Shane James|Local/2001@agents/n;0;0;Local/2001@agents/n |
| 11:18.59 | phpboy | ? |
| 11:19.31 | *** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 11:19.43 | joobie | JT.. are the drivers decent for the sangoma a101d under * ? i |
| 11:19.54 | *** part/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 11:19.56 | joobie | i dont use trix/freepbx.. just straight * |
| 11:20.10 | joobie | and also do you happen to know a cheap place to order the sangoma a101d from? |
| 11:21.03 | mort_gib | joobie: Sangoma is generally ok, support a bit longwinded, but they are there for you |
| 11:22.31 | mort_gib | I use voipon.co.uk but then I'm in Spain :-) |
| 11:23.03 | joobie | heh :) |
| 11:23.10 | joobie | yes Spain could be a prob heh |
| 11:23.32 | mort_gib | No, I get my stuff from UK, they do almost day to day delivery |
| 11:23.34 | phpboy | how do I go about deleting the following entry out of the asterisk DB:- /Queue/PersistentMembers/onecellControl : Local/2001}@agents/n;0;0;Shane James|Local/2001@agents/n;0;0;Local/2001@agents/n |
| 11:24.09 | joobie | ahh |
| 11:24.59 | kaldemar | phpboy: database del <key> |
| 11:25.22 | kaldemar | sorry, database del <family> <key> |
| 11:25.24 | phpboy | what part of that is the key? |
| 11:25.35 | phpboy | what's the family and what's the key? |
| 11:25.50 | kaldemar | try it. |
| 11:25.55 | phpboy | :( |
| 11:26.03 | JT | joobie: yeah fine under asterisk |
| 11:26.06 | phpboy | I don't know which part is the key and which part is the family |
| 11:26.39 | yang | JT: If I change the codec preference on asterisk to use only "g729" for that sip account I get the error: [2008-10-20 13:25:36] WARNING[31396]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'ZAP' (cause 58 - Bearer capability not available) |
| 11:26.46 | kaldemar | phpboy: be creative. |
| 11:27.05 | kaldemar | or do some reading. |
| 11:27.06 | JT | yang: well of course if it's zaptel you need to transcode |
| 11:27.12 | yang | ah |
| 11:27.24 | mort_gib | yang: Did you register the G729 codes |
| 11:27.29 | kaldemar | Queue/PersistentMembers is the family and onecellControl is the key, most likely. |
| 11:28.05 | yang | mort_gib: no, i haven't purchased the license yet (thinking of doing so), but if those are Free to use with SIP connection then its no need to purchase |
| 11:28.15 | kaldemar | phpboy: / is the delimiter for families and keys. you should be able to figure out which is which. |
| 11:28.45 | mort_gib | If you want to transcode or use G729 you need a license |
| 11:28.53 | mort_gib | Change away from G729 and try again |
| 11:29.22 | mort_gib | I had a similar issue, with a reinstall of a system where the ONLY thing I forgot was to register G729 anew |
| 11:31.05 | yang | JT: if I try to call/receive over SIP I get [2008-10-20 13:29:31] WARNING[22943]: chan_sip.c:3024 sip_call: No audio format found to offer. Cancelling call to 222 |
| 11:31.22 | yang | JT: probably the uplink needs to enable g729 in this case? |
| 11:31.46 | JT | yeah |
| 11:36.59 | *** join/#asterisk Visin (n=david@ocean.nebaut.net) |
| 11:39.54 | *** join/#asterisk MrNaz (n=mrnaz@117.47.197.199) |
| 11:40.28 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 11:43.43 | phpboy | I really ned to replace chan_agent |
| 11:43.58 | phpboy | I am having a really difficult time getting this right :( |
| 11:49.47 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
| 11:53.21 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 11:53.41 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 11:58.18 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 12:03.08 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:23.25 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 12:26.53 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:33.34 | phpboy | [TK]D-Fender: do you by any chance know of a good chan_agent replacement in the dialplan? |
| 12:33.45 | [TK]D-Fender | phpboy: huh? |
| 12:34.23 | phpboy | I'm trying to replace chan_agent in my asterisk installation |
| 12:34.33 | phpboy | I'm having a LOT of trouble :( |
| 12:34.44 | [TK]D-Fender | phpboy: Your description is lacking.... |
| 12:34.59 | seanbright | agent callback login... $5 |
| 12:35.08 | seanbright | any takers? |
| 12:35.13 | phpboy | seanbright: that's what I'm trying to replace |
| 12:35.17 | seanbright | ding ding ding! |
| 12:35.29 | phpboy | agentcallbacklogin is troublesom |
| 12:35.37 | [TK]D-Fender | seanbright: I got mine free with *.... you'll have to peddle elsewhere ;) |
| 12:35.58 | seanbright | [TK]D-Fender: no no, i was betting $5 that he was referring to callback login |
| 12:36.13 | seanbright | phpboy: isn't there a sample extensions.conf in doc/ somewhere? |
| 12:38.40 | phpboy | seanbright: it doesn't have the functionality that I require |
| 12:38.52 | seanbright | i see |
| 12:38.54 | phpboy | the example pretty much binds an agent to a specific extension :/ |
| 12:39.04 | seanbright | i have never looked at it |
| 12:39.07 | seanbright | so i dunno. |
| 12:39.12 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-1c999b42eae101e3) |
| 12:40.01 | phpboy | seanbright: more importantly, queuemetrics is a problem |
| 12:40.07 | [TK]D-Fender | Yup... as ManxPower would say "Itdefinitely looks like a Monday" |
| 12:40.16 | phpboy | what's I've gone and done is a cheap hack |
| 12:40.22 | phpboy | :( |
| 12:40.28 | phpboy | (I think) |
| 12:40.36 | seanbright | most stuff is. |
| 12:40.44 | seanbright | is a cheap hack |
| 12:41.12 | seanbright | the only good thing about today is that the red sox lost last night |
| 12:41.20 | seanbright | that's a good start to my week |
| 12:48.18 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:50.14 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:51.47 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:51.55 | ana_micho | hi all,How to force asterisk to send an OK SIP message as a response to INVITE from an inbound peer? |
| 12:53.12 | synthetiq | wait() |
| 12:54.12 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:54.12 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:59.53 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 13:02.33 | synthetiq | anyone sell dids for egypt, indonesia, or india? |
| 13:05.18 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:07.02 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 13:12.37 | *** join/#asterisk nosbig (n=nosbig@cpe-75-185-59-24.columbus.res.rr.com) |
| 13:16.31 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 13:17.47 | *** join/#asterisk tvirus (i=TheVirus@c-69-243-46-240.hsd1.va.comcast.net) |
| 13:18.46 | tvirus | Is it possible to have a specific email subject if the same people are getting the same voicemail? If someone leaves a voicemail after hours, we all get it, is it possible to change the subject to say 'After Hours email'? |
| 13:19.19 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 13:19.59 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 13:21.17 | [TK]D-Fender | tvirus: Make a sendmail script front-end and have that pass it on to sendmail |
| 13:22.28 | tvirus | I'm using Postfix, so I'll see about doing that. |
| 13:22.59 | [TK]D-Fender | tvirus: * still uses a sendmail compatible script. |
| 13:23.16 | [TK]D-Fender | tvirus: jsut change the name of the one * uses and mod the parameters |
| 13:23.21 | ana_micho | Hi all, How to force asterisk Server to send OK after an INVITE message from an outbound peer? |
| 13:23.49 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 13:26.59 | *** join/#asterisk mog (n=mog@nat/digium/x-e9c6c4f59cedc5e6) |
| 13:26.59 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:32.54 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:35.47 | *** join/#asterisk MrNaz (n=mrnaz@222.123.117.70) |
| 13:35.54 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 13:35.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:38.14 | ana_micho | Hi all, How to force asterisk Server to send OK after an INVITE message from an outbound peer? |
| 13:38.28 | ana_micho | ANy body have any clue about how to do that? |
| 13:38.47 | creativx | ana_micho: obviously not many. |
| 13:39.27 | [TK]D-Fender | ana_micho: No more than 15 minutes and 11 lines (none actual text) ago |
| 13:39.46 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:40.24 | ana_micho | [TK]D-Fender, asterisk should not send OK SIP message after a INVITE sent from inbound peer? |
| 13:40.33 | creativx | lol |
| 13:41.00 | [TK]D-Fender | ana_micho: Might depend if its OK or not. |
| 13:42.03 | [TK]D-Fender | ana_micho: And you aren't doing yourself any favours by not showing use the SIP dialog for your inbound call along with its config. |
| 13:42.03 | [TK]D-Fender | us* |
| 13:42.11 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 13:42.15 | ana_micho | [TK]D-Fender, Sorry I'll do that |
| 13:42.49 | [TK]D-Fender | ana_micho: When you actually want help you should be more proactive and not make us ask for things like backup for your problem. |
| 13:43.11 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 13:43.38 | ana_micho | [TK]D-Fender, http://pastebin.com/d76ed39c3 |
| 13:44.22 | ana_micho | [TK]D-Fender, my Inbound carrier was telling me that they are not sending DTMF digits because they are not receiving OK message to their INVITE |
| 13:45.21 | [TK]D-Fender | ana_micho: [Oct 20 13:20:54] WARNING[10017]: res_agi.c:2129 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI |
| 13:46.33 | [TK]D-Fender | ana_micho: I do not see an explicit Answer in there. Do that as your first priority |
| 13:47.19 | ana_micho | [TK]D-Fender, Let me use AGI and see what will happen |
| 13:47.24 | cesar_CR | hello guys! |
| 13:47.30 | [TK]D-Fender | ana_micho: try the "answer" first |
| 13:47.51 | [TK]D-Fender | ana_micho: Just stated the other warning as a warning. Not sure about A2B's requirement foe deadagi which is possible |
| 13:50.15 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:51.22 | ana_micho | [TK]D-Fender, Let me add Answer and I'll let you know directly |
| 13:51.27 | ana_micho | [TK]D-Fender, thanks a lot |
| 13:54.29 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 13:54.54 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 13:55.09 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:56.15 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:56.55 | ana_micho | [TK]D-Fender, the problem occurs a lot when trying to call from a PSTN line |
| 13:57.13 | ana_micho | [TK]D-Fender, when calling from an endpoint registered on a SIP proxy everything looks fine |
| 13:57.27 | [TK]D-Fender | ana_micho: Please show us something useful |
| 13:57.42 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 13:58.02 | ana_micho | [TK]D-Fender, I'll show you now a successfull call with good DTMF sent |
| 13:58.02 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:58.06 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:59.08 | ana_micho | [TK]D-Fender, this is a working example http://pastebin.com/m528c860a |
| 13:59.23 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:00.40 | *** join/#asterisk MrNaz (n=mrnaz@222.123.109.7) |
| 14:01.58 | *** join/#asterisk MrNaz (n=mrnaz@222.123.109.7) |
| 14:08.15 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 14:11.56 | *** join/#asterisk zydoon (n=zydoon@41.225.156.138) |
| 14:13.58 | *** part/#asterisk zydoon (n=zydoon@41.225.156.138) |
| 14:15.23 | *** part/#asterisk RachedTN (n=virtex5@213.150.163.105) |
| 14:15.45 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-74744bb4b0499cb6) |
| 14:15.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:30.26 | *** join/#asterisk seanmh (n=seanmh@216.31.101.11) |
| 14:31.43 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-ee2d9f0415662912) |
| 14:31.43 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:32.20 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 14:33.12 | *** join/#asterisk queuetue (n=scott@MTRLPQ02-1279391519.sdsl.bell.ca) |
| 14:41.10 | *** join/#asterisk propellerhead (n=yogurt2u@host125.190-136-238.telecom.net.ar) |
| 14:41.21 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:42.21 | zchaos | anyone know of any good providers of 1-800 numbers? I thought i heard of a service with free incoming calls? |
| 14:42.37 | [TK]D-Fender | zchaos: Not for 800's |
| 14:43.52 | synthetiq | haha 1800 free incoming |
| 14:46.45 | mchou | would like to see the business model for that one |
| 14:47.21 | iCEBrkr | [TK]D-Fender: Yo |
| 14:47.29 | [TK]D-Fender | iCEBrkr: y0 |
| 14:47.46 | *** part/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 14:47.51 | iCEBrkr | [TK]D-Fender: Is [TK] a clan? I was playing TF2 last night and some [tk] dr.tom or something joined |
| 14:47.55 | iCEBrkr | :P |
| 14:48.07 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 14:48.07 | *** mode/#asterisk [+o angler] by ChanServ |
| 14:48.16 | hawk | mchou: Just some ads injected into the phonecalls... Ads seem to be the solution to all the other impossible business models... |
| 14:48.26 | [TK]D-Fender | mchou: I'd be made up of S&L vouchers papier mache'd together with a tag of "actual size" pinned on the bottom |
| 14:48.53 | [TK]D-Fender | it'd* |
| 14:51.59 | *** join/#asterisk asteriskpbx (n=asterisk@scalise.csata.tno.it) |
| 14:52.06 | asteriskpbx | hi to all |
| 14:52.19 | russellb | stares at asteriskpbx |
| 14:52.22 | asteriskpbx | anyone of you have experience with Bristuff and wants to share infos with me? |
| 14:52.24 | russellb | could you please change your nick? |
| 14:52.36 | asteriskpbx | humm it is registered with nickserv now |
| 14:52.43 | russellb | register a different one, then |
| 14:52.45 | asteriskpbx | if necessary I re-register with another name |
| 14:52.48 | asteriskpbx | ok |
| 14:52.54 | seanbright | ? |
| 14:53.15 | *** join/#asterisk astrOdz (n=astrOdz@119.92.213.100) |
| 14:53.21 | zchaos | <[TK]D-Fender> zchaos: Not for 800's |
| 14:53.25 | zchaos | suggestions on a good provider for 1800 then? |
| 14:53.35 | [TK]D-Fender | zchaos: Go show apround |
| 14:53.44 | [TK]D-Fender | zchaos: Go shop around |
| 14:54.23 | *** join/#asterisk ManxPower (n=manxpowe@217.sub-70-221-175.myvzw.com) |
| 14:57.13 | ast-thecode | I changed my nickname |
| 14:57.18 | ast-thecode | now... anyone of you have experience with Bristuff and wants to share infos with me? |
| 14:57.28 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:57.56 | tzafrir_laptop | ast-thecode, maybe if you'll ask some specific questions |
| 14:58.53 | jaytee | can someone help me? i've got this thingy that won't talk to the other thingy and it's beeping and booping alot. :-) |
| 14:59.28 | tzafrir_laptop | someone can |
| 14:59.40 | tzafrir_laptop | the problem is that someone just logged out |
| 15:00.04 | tzafrir_laptop | we're left with just anybody |
| 15:00.56 | ast-thecode | ok the question is this |
| 15:01.05 | ast-thecode | I have some installations with Bristuff in Italy |
| 15:01.35 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 15:01.43 | ast-thecode | about 2-3 installations have the problem that random (about 4-5 times a month) the zap channel of the BRI channel has a "InAlarm: 1" set |
| 15:02.01 | tzafrir_laptop | what device do you use? |
| 15:02.03 | ast-thecode | however I can stillr ecevive phone alls but I cannot call outgoing |
| 15:02.13 | ast-thecode | Junghanns 2-bri |
| 15:03.57 | ast-thecode | tzafrir_laptop you are Tzafrir Cohen? |
| 15:05.14 | [TK]D-Fender | Karlito's (A)Way! |
| 15:06.07 | ast-thecode | Tzafrir.. are you still there? |
| 15:06.16 | jameswf-home | ast-thecode: dont tell anyone, |
| 15:06.31 | tzafrir_laptop | ast-thecode, yes, and yes |
| 15:06.40 | jameswf-home | in here he takes off the tights and cape |
| 15:07.43 | tzafrir_laptop | What version of bristuff do you use? |
| 15:09.58 | ast-thecode | pleased to meet you |
| 15:10.03 | ast-thecode | RC3 |
| 15:13.54 | ast-thecode | that is the last |
| 15:14.40 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:14.52 | *** join/#asterisk ManxPower (n=manxpowe@217.sub-70-221-175.myvzw.com) |
| 15:15.13 | ast-thecode | I found that some other people in the web has the same problem but no-one have a solution |
| 15:18.55 | *** join/#asterisk metfan2007 (n=jc@201.103.3.167) |
| 15:18.59 | metfan2007 | hi all! |
| 15:20.33 | jameswf | not lately |
| 15:20.53 | metfan2007 | I'm recording audio from queues via queue.conf, but for some cases I'm getting a only UNIQUEID file name (like 1224515120.3394.gsm) and for other cases I'm getting something like agent-101-1224504026-3167.gsm, I need all the recording names like last one (agent-blabla) any idea? |
| 15:21.10 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 15:21.34 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:33.14 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 15:34.21 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:36.15 | jameswf | what is with AT&T and their screwed up release delays... |
| 15:36.31 | bkruse | jameswf: There is your problem, AT&T |
| 15:36.45 | jameswf | yeah not my choice |
| 15:37.00 | jeev | i installed asteriskNOW beta |
| 15:37.01 | jeev | to test :D |
| 15:37.04 | ast-thecode | tzafrir are you still there? |
| 15:37.08 | jameswf | i want the BB bold, need to steal one from canada |
| 15:37.52 | jeev | get me one too |
| 15:37.55 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:38.10 | jameswf | [TK]D-Fender: in in canada **HINT** |
| 15:38.12 | jameswf | :) |
| 15:38.54 | [TK]D-Fender | jameswf: You've got a PPC6700 right? |
| 15:39.12 | jameswf | nope |
| 15:39.51 | [TK]D-Fender | jameswf: Will probably take be a few mins to recall who that was I'm thinking of... |
| 15:40.19 | jeev | jameswf, buy me this http://cgi.ebay.com/New-Retail-Blackberry-Bold-9000-3G-Unlocked-Cell-Phone_W0QQitemZ290268477362QQcmdZViewItem?hash=item290268477362&_trkparms=72%3A570|39%3A1|66%3A2|65%3A12|240%3A1309&_trksid=p3286.c0.m14 |
| 15:40.39 | tzafrir_laptop | ast-thecode, there is RC3c |
| 15:43.09 | Ritzerisk | whats the best audio quality codec if having a remote Sip client |
| 15:43.23 | Ritzerisk | the audio i have now is soo slugish like a 2 sec delay |
| 15:46.00 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 15:46.04 | jameswf | g729 |
| 15:46.13 | *** join/#asterisk bmg505 (n=leon@196-209-76-252-tbnb-esr-2.dynamic.isadsl.co.za) |
| 15:46.28 | [TK]D-Fender | Ritzerisk: G.722 |
| 15:46.48 | joako | Ok.. I did something really stupid.... rm /etc/asterisk/sip.conf... but now one peer with the correct password gives error 401 on registration after rebuilding the config |
| 15:47.09 | joako | Any idea what can cause error 401 besides username or password? |
| 15:49.26 | Ritzerisk | sweet ill see how to change that from the codec i dont know doesnt the G.729a hog down the Cpu .... |
| 15:52.56 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 15:53.46 | *** join/#asterisk ManxPower (n=manxpowe@174.sub-75-201-161.myvzw.com) |
| 15:53.51 | mort_gib | TK -But G.729 does use less bandwidth, does is not?? |
| 15:54.04 | [TK]D-Fender | mort_gib: Less that...? |
| 15:54.06 | [TK]D-Fender | than* |
| 15:54.22 | ast-thecode | RC3c? |
| 15:54.25 | mort_gib | G.722 |
| 15:54.31 | [TK]D-Fender | mort_gib: Indeed it does. |
| 15:54.40 | mort_gib | But better quality right?? |
| 15:54.49 | [TK]D-Fender | mort_gib: G.722 is top quality |
| 15:55.14 | ast-thecode | sorry you are right.. the version I am using is: Asterisk 1.4.21.2-BRIstuffed-0.4.0-RC3c |
| 15:55.42 | ast-thecode | however with old versions I had same problems |
| 15:56.09 | mort_gib | I can't use that for external calls... |
| 15:58.08 | mort_gib | I have yet to install * for a client with a leased line :-( |
| 15:58.08 | joako | How do I diagnose error SIP/2.0 401 Unauthorized |
| 15:58.24 | joako | I checked the SIP UA and sip.conf and everything matches! |
| 15:59.11 | Ritzerisk | i think the g.729 uses 8kb but hits the cpu a bunch to but i could be wrong i just remember reading it somewhere |
| 15:59.21 | ManxPower | joako: Pastebin the CLI output of a failed call. |
| 15:59.41 | ManxPower | ~pb |
| 15:59.42 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:59.59 | tzafrir_laptop | ast-thecode, sorry, my mistake: so those channels are in alarm. because your provider dropped the line |
| 15:59.59 | mort_gib | Ritzerisk: True, but monitor your CPU before and after if you want |
| 16:00.08 | tzafrir_laptop | (common habbit on BRI) |
| 16:00.42 | mort_gib | tzafrir_laptop: Yeah?? Would that also drop the call?? |
| 16:00.54 | joako | @ManxPower http://pastebin.ca/1231950 |
| 16:01.02 | ManxPower | mort_gib: alarms ALWAYS drop calls. |
| 16:01.08 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 16:01.30 | mort_gib | Uhm, yes I have a client that gets dropped calls from time to time. |
| 16:01.45 | mort_gib | In * it looks like a normal hangup() |
| 16:01.52 | tzafrir_laptop | mort_gib, that only happens when there are no calls |
| 16:01.54 | ManxPower | joako: you will always have a 401 unauth. Then the device should re-connect with the password/username |
| 16:02.14 | tzafrir_laptop | To save them from providing power to ISDN phones |
| 16:02.18 | joako | Well thats all I have |
| 16:02.21 | ManxPower | joako: In the future when I say "pastebin the output of a failed call" I mean without debugging enabled, only verbost enabled. |
| 16:02.39 | joako | w/o sip debug there is nothing |
| 16:02.49 | ManxPower | joako: then tell me that before pasting. |
| 16:02.53 | mort_gib | I use Sangoma A500 so the "MediaGateway" handles calls |
| 16:03.00 | joako | i did rm /etc/asterisk/sip.conf by accident, rebuilt the sip.conf file and now it does not work |
| 16:03.01 | ManxPower | joako: what is your SIP device? |
| 16:03.12 | mort_gib | Speaking to Sangoma they claim that their cards works 100% |
| 16:03.16 | ManxPower | joako: so your problem is in sip.conf then. |
| 16:03.25 | tzafrir_laptop | mort_gib, it is generally better to use ptp and not ptmp and avoid that nonsense |
| 16:03.30 | joako | I am positive the username and p/w are correct.. its a grandstream GXP280 & I rebuilt the provisioning file |
| 16:03.44 | joako | Right.... so how do I determine what the issue is since it is not the username/pw? |
| 16:03.45 | ManxPower | mort_gib: since Digium redesigned their cards a few years ago their cards work in virtually all systems. |
| 16:04.01 | ManxPower | joako: how long is your sip.conf? |
| 16:04.18 | joako | 568 lines |
| 16:04.21 | ManxPower | ok, so you rebuilt BOTH sip.conf AND the provisioning file for the phone? |
| 16:04.37 | mort_gib | Ok, That client IS actually using ptmp |
| 16:04.38 | ast-thecode | tzafrir, sorry for the delay... It is not a problem of energy saving |
| 16:04.38 | ManxPower | joako: Well, I cannot help you with that. BEST of luck and next time make some backups. |
| 16:04.47 | [TK]D-Fender | joako: And the reason you haven't provided your sip.conf is...? |
| 16:04.48 | ast-thecode | infact this happens even during use of the lines |
| 16:05.15 | tzafrir_laptop | ast-thecode, can you pastebin some logs? |
| 16:05.25 | ManxPower | [TK]D-Fender: I've not asked for sip.conf. I don't wade thru 500+ lines of sip.conf for free. |
| 16:05.35 | joako | I think I fixed it /banghead |
| 16:05.45 | tzafrir_laptop | or better, send them to http://lists.three-dimensional.net/mailman/listinfo/bristuff-users |
| 16:05.46 | [TK]D-Fender | ManxPower: text search = 5 seconds. |
| 16:06.00 | joako | a diffrent SIP peer I put two entries for. I don't see how that would cause an issue but somehow it did |
| 16:06.02 | ManxPower | [TK]D-Fender: thank you for volunteering to help this guy. |
| 16:06.33 | [TK]D-Fender | ManxPower: He jsut fixed it himself... robably in reaction my my asking to look at the exact place he broke. |
| 16:06.38 | ManxPower | [TK]D-Fender: and yet, it never ends up being 5 seconds. |
| 16:06.43 | ast-thecode | the only error message is: == Primary D-Channel on span 2 down |
| 16:06.43 | ast-thecode | chan_zap.c: No D-channels available! Using Primary channel 3 as D-channel anyway! |
| 16:06.48 | [TK]D-Fender | ManxPower: Did jsut now :) |
| 16:07.00 | ManxPower | ast-thecode: that message indicates your LINE IS DOWN. |
| 16:07.10 | mort_gib | ManxPower: I'm using Sangoma cards |
| 16:07.37 | ManxPower | mort_gib: I switched to Sangoma 2 years before Digium redesigned their cards. |
| 16:07.54 | Ritzerisk | does asterisk have commands to monitor cpu usage or do i have to back out and use commands in linux |
| 16:08.12 | ManxPower | Ritzerisk: use the OS, Luke. |
| 16:08.24 | mort_gib | Ritzerisk: top |
| 16:08.27 | mort_gib | :-) |
| 16:08.28 | Ritzerisk | luke |
| 16:08.30 | *** join/#asterisk HumpBack (n=sdcdcssc@gentoo/developer/HumpBack) |
| 16:08.40 | HumpBack | Hello all |
| 16:08.56 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 16:09.03 | Ritzerisk | what conf you change your codec in again i cant remember which one |
| 16:09.15 | mort_gib | Whenever I finish off a system I dial in all handsets into meetme and watch top, like in trance :-) |
| 16:09.24 | ManxPower | Ritzerisk: ok. <obiwankenobi>Use the OS,Luke</obiwankenobi> |
| 16:10.16 | HumpBack | If I have a Asterisk box with a TE121B and a Ericson PBX with a E1 port what else do I need to connect one to the other? |
| 16:10.39 | ast-thecode | TZAFIR: which logs do you think that could be useful? |
| 16:10.42 | ManxPower | HumpBack: a crossover E-1 cable (I assume it is the same pinout as T-1 crossover) |
| 16:11.25 | ManxPower | HumpBack: If you know Asterisk it should take less than a few hours to do the Asterisk side. Good lick with the Ericson side. |
| 16:12.51 | Qwell | Please do not lick your PBXs. |
| 16:13.07 | denon | at least not in public |
| 16:13.52 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-89-142.w86-215.abo.wanadoo.fr) |
| 16:14.42 | WimpMan | Disconnect power before licking! |
| 16:15.10 | *** join/#asterisk n3hxs (n=HAMming@static-71-242-245-74.phlapa.east.verizon.net) |
| 16:15.43 | ast-thecode | exit |
| 16:16.34 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 16:17.12 | jameswf | was dissapointed fireworks show turned out to be a defective sparkler |
| 16:18.36 | ManxPower | jameswf: some idiot threw (what they thought was an empty) fireworks box on the campfire the other night. It was apparently not empty. |
| 16:19.13 | ManxPower | The result was.....exciting. |
| 16:20.42 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 16:25.12 | HumpBack | ManxPower: Thanks m8 |
| 16:27.09 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 16:30.08 | *** join/#asterisk talntid (n=eric@66.208.251.170) |
| 16:35.25 | Ritzerisk | in the sip conf the allow= do i put allow=ulaw then allow=g722 |
| 16:35.35 | Ritzerisk | cause it also has a disallow=all |
| 16:39.26 | [TK]D-Fender | Ritzerisk: Disallow first, then the allows |
| 16:40.43 | jameswf | I wonder if tim will pull those ugly green boxes from his site |
| 16:43.08 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:43.16 | *** join/#asterisk Defraz (n=T0tal@63.228.246.250) |
| 16:43.17 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:43.46 | *** join/#asterisk flohack (n=fhackenb@lancelot.acoveo.com) |
| 16:44.08 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:44.22 | diegows | asterisknow or trixbox? opinions? |
| 16:44.46 | Qwell | jameswf: |
| 16:44.57 | [TK]D-Fender | diegows: Both run GUI's that tell you how to run your system. You either like them, ro don't |
| 16:45.24 | tzafrir_laptop | any zaptel hardware setup for freepbx? |
| 16:45.34 | diegows | i know that, but may be one of them is more mature |
| 16:45.47 | diegows | or has some problemas |
| 16:45.49 | diegows | problems |
| 16:45.58 | [TK]D-Fender | diegows: FreePBX (included with each) is "mature" |
| 16:46.08 | [TK]D-Fender | diegows: *-GUI isn't so much |
| 16:46.15 | tzafrir_laptop | err, any way to make freepbx aware of zaptel^Wdahdi hardware? |
| 16:46.22 | lesouvage | diegows: you could take druid and scopserv into consideration. |
| 16:46.40 | tzafrir_laptop | [TK]D-Fender, the asterisk-gui includes at least that much |
| 16:47.06 | [TK]D-Fender | tzafrir_laptop: FreePBX is not 1.6 compliant IIRC |
| 16:47.10 | jameswf | diegows: probably the wrong place to ask, they have different audiences. if your an open source moralist probably asteriskNOW if you dont mind being a sheep like green andgreen and dont mind being taken out once and a while by a rouge update trixbox, if you want all that without the green piaf |
| 16:47.10 | diegows | lesouvage: thanks |
| 16:47.13 | Qwell | 2.5 is |
| 16:47.22 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 16:47.44 | [TK]D-Fender | s/not/sow |
| 16:47.50 | [TK]D-Fender | tzafrir_laptop: FreePBX is now 1.6 compliant IIRC |
| 16:47.57 | RypPn | ZAP2DAHDICOMPAT=true|false in amportal.conf does the necessary magic |
| 16:47.58 | [TK]D-Fender | Qwell: Typo... I did know it was |
| 16:48.32 | tzafrir_laptop | [TK]D-Fender, this is not what I asked about. It is not aware of the installed Z/D hardware |
| 16:48.46 | Qwell | tzafrir_laptop: no, I don't believe there's any autoconfiguration of hardware |
| 16:48.53 | tzafrir_laptop | You have to tell it what channels/groups to use explicitly |
| 16:48.55 | jameswf | if you want freedom DONT GUI buy the book |
| 16:48.58 | jameswf | ~buybook |
| 16:48.59 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 16:49.09 | Qwell | tzafrir_laptop: though I really can't imagine it being that difficult to add - especially with the scripts you've written |
| 16:49.37 | Qwell | if only I knew php.. |
| 16:49.54 | jameswf | Qwell: tzafrir_laptopdoes perl |
| 16:50.04 | Qwell | freepbx != perl |
| 16:50.34 | jameswf | freepbx is a mild blend of bash php and perl |
| 16:50.43 | jameswf | makes you feel kinda dirty |
| 16:50.47 | jameswf | :) |
| 16:51.00 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:51.02 | *** join/#asterisk Laureano (n=Laureano@190.245.108.2) |
| 16:51.08 | tvirus | So I got a new TDM410P with the EC module and dmesg is still saying 'missed interrupt'. Does this card not play well with Fedora? |
| 16:51.08 | aiksa[LV] | just wanted to make a major public thanks to tzafrir_laptop for helping with astribank setup! |
| 16:51.32 | jeev | why are wired routers cheaper than wireless |
| 16:51.33 | jeev | so gay |
| 16:51.53 | jameswf | heh Fonality wants to disrupt closed source VoIP hardware and software providers. |
| 16:51.59 | Nugget | calling things gay is gay. |
| 16:52.06 | jameswf | um dude |
| 16:52.08 | De_Mon | jeev ... |
| 16:52.10 | jeev | negative mcnugget |
| 16:52.19 | jeev | sorry |
| 16:52.21 | jeev | i meant more expensive |
| 16:52.24 | jeev | lol |
| 16:52.49 | De_Mon | Ah, that sounds better. The answer is still ... |
| 16:52.52 | coppice | jeev: ITS SOMETHING TO DO WITH THE VOLUME! |
| 16:53.02 | jeev | ............... damn you |
| 16:53.04 | De_Mon | ^^-- what he said |
| 16:53.14 | Qwell | coppice: I THINK YOU'RE CONFUSING WHAT THE WORD VOLUME MEANS |
| 16:53.15 | diegows | jameswf: I want an asterisk distribution with a good interface that allow custom configuration without broken it |
| 16:53.18 | jeev | asterisk is imposing a tax |
| 16:53.25 | jeev | to be in here, you must paypal me everything |
| 16:53.28 | jameswf | will use clialiases to replace all the depricated stuff |
| 16:53.45 | jameswf | diegows: ummmmmm |
| 16:54.17 | jeev | wishes he was fluent with php |
| 16:54.57 | jameswf | is pretty php fluent, not so mush freepbx way of php fluint |
| 16:55.14 | jeev | demands jameswf acts as my free, on demand, right hand man. |
| 16:55.19 | Ritzerisk | so disallow all then allow=722 |
| 16:55.20 | diegows | want an intertace coded in python :) |
| 16:55.49 | bpgoldsb | My volume goes up to 11. |
| 16:56.15 | jeev | sucks for you |
| 16:56.22 | De_Mon | aww ya. dexter's getting hitched |
| 16:57.03 | coppice | isn't fluency in PHP akin to being a potty mouth? |
| 16:57.50 | jeev | stop making it so difficult and give me the knowledge, i already know everything else. |
| 16:58.27 | tvirus | wctdm24xxp0: Missed interrupt. Increasing latency to 6 ms in order to compensate. <-- Anyway to debug that. 2 different cards and 2 different PCs running Fedora Core 9 in init3 |
| 16:59.53 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 17:01.48 | bpgoldsb | tvirus: Thats not actually a problem, according to Digium |
| 17:02.06 | bpgoldsb | That always happened, but until more recent drivers, the card wasn't smart enough to compensate. |
| 17:02.30 | bpgoldsb | I had the same thing when I went from a TE110 -> TE120 |
| 17:04.19 | tvirus | But the EC module isn't working. There's horrible echo that makes it impossible to talk to them. We have to call them back. |
| 17:05.40 | hesco | I'm sitting here recording a number of IRV prompts, dropping .call files in spool/.../outgoing to initiate the calls, and even after a server reload, *, which a moment ago was cooperating with me is now ignoring the .call file in its spool. Why might that be? |
| 17:07.05 | diegows | i need to configure asterisk in a way that when a user calls to some extension he keeps waiting with music until the other end press '1,2,3'. |
| 17:07.17 | diegows | i don't want the solution |
| 17:07.24 | diegows | just only a starting point |
| 17:07.36 | [TK]D-Fender | diegows: "core show application dial" |
| 17:08.01 | cj | anybody know if anybody builds a T2 (or greater) card? I heard there's a T3 de-MUX dongle, but it seems like it would better to plug it directly into the PCI-X bus as a single line. |
| 17:08.05 | [TK]D-Fender | hesco: datestamp |
| 17:08.58 | tvirus | It seems only channel 1 is affected with the EC shit. |
| 17:09.02 | tvirus | Channel 2 sounds fine. |
| 17:09.10 | cj | the argument I hear against it is that asterisk couldn't decode that many DS-0s... but if it's just bridging the channels to a different (set of) systems to do the hard work, it seems like it should be able to handle it... |
| 17:10.42 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:11.28 | Qwell | cj: That's only a little over a quad T1 |
| 17:11.54 | cj | Qwell: isn't a T3 4 T2s, and a T2 7 T1s? |
| 17:12.22 | Qwell | a "T2" is 96 channels. DS3 is 672 |
| 17:12.34 | cj | comes out to 28 T1s if that's right... |
| 17:12.48 | Qwell | correct |
| 17:13.11 | cj | so how is a T3 only a little over a quad T1? sounds like it's 7 quad T1s...? |
| 17:13.24 | Qwell | You asked for a "T2" card. |
| 17:13.31 | Qwell | That is just a 4 T1s |
| 17:13.40 | cj | oh, sorry... I guess I was talking about the (or greater) part :) |
| 17:14.11 | coppice | does anyone actually use T2? |
| 17:14.15 | cj | yeah, T2 = quad T1... couldn't it be delivered over a single line, so it didn't have to be split out prior to plugging into the PCI-X bus? |
| 17:14.32 | cj | coppice: I've only ever read about it in theory... never seen one for lease :) |
| 17:14.42 | Qwell | coppice: not that I've seen |
| 17:14.49 | hesco | [TK]D-Fender: yes, datestamp was the first thing I checked. But it was in the past. I did a halt now, removed the recalcitrant .call file, restarted the server and dropped a new .call file in the mix and its working again. No idea what that was about. |
| 17:15.04 | coppice | E2 used to be common, but things seem to jump from E1 to E3 these days |
| 17:15.15 | hesco | but thanks for the idea. |
| 17:15.51 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 17:16.08 | [TK]D-Fender | coppice: Becomes less efficient / cost effective to collect too many intermediate steps I guess.... |
| 17:17.37 | coppice | we developed steam powered E2 junction laser cards in the 80s |
| 17:25.51 | cvnet | im trying to register a sip phone on my asterisk box, but it doesnt register, and on sip set debug i dont see anything at all, what should i try? |
| 17:26.25 | Ritzerisk | anyone know which file with centos i can use to nano the eth0 i cant remember where it hides from me |
| 17:27.01 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:27.05 | Qwell | Ritzerisk: somewhere in /etc/udev/ |
| 17:27.41 | Qwell | maybe |
| 17:27.48 | Ritzerisk | haha i tried dev |
| 17:28.50 | jameswf | 1,076,549 calls weeee |
| 17:28.52 | jeev | qwell, where is russell |
| 17:29.37 | *** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 17:29.40 | devilsoulblack | hi |
| 17:31.01 | cvnet | help plz |
| 17:31.05 | cvnet | im trying to register a sip phone on my asterisk box, but it doesnt register, and on sip set debug i dont see anything at all, what should i try? |
| 17:31.12 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 17:31.34 | Qwell | cvnet: check your firewall |
| 17:31.44 | Qwell | and/or nat settings |
| 17:31.53 | cvnet | firewall is disabled |
| 17:31.56 | devilsoulblack | i recieve soo many message " == Primary D-Channel on span 1 up" |
| 17:32.03 | cvnet | and im ssl'ing to the server |
| 17:32.16 | cvnet | and i can ping |
| 17:32.37 | cvnet | touch the sip file and just put [general] |
| 17:33.00 | Ritzerisk | haha dam its etc/sysconfig/network-scripts/ifcfg-eth0 darn |
| 17:34.30 | *** join/#asterisk Greek-Boy (n=email@41.222.89.114) |
| 17:36.09 | cvnet | any suggestions? |
| 17:37.20 | *** part/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 17:42.39 | joako | What would cause the error "rejected because extension not found" I do sip show peer and then show dialplan and I am dialing a correct extension!! |
| 17:43.01 | caio1982 | correct extension, wrong context? |
| 17:50.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:55.10 | joako | No, I have the correct context! |
| 18:02.20 | jameswf | goal is to generate 600 calls per second... |
| 18:03.05 | jameswf | real calls none of that sip crap |
| 18:05.51 | *** join/#asterisk steliosk (n=Stelios@79.107.29.93) |
| 18:06.12 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 18:07.51 | hesco | Since some changes I made last night, I've been getting the following cdr error: pgsql_log: cdr_pgsql: Reason: ERROR: column "calldate" specified more than once. Where is it please that the caldate might be specified more than once? |
| 18:08.36 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
| 18:08.43 | hesco | the mapping in cdr_custom.conf is still commented out. |
| 18:10.33 | jeev | jameswf, you know you love SIP |
| 18:10.53 | joako | What would cause the error "rejected because extension not found" I do sip show peer and then show dialplan and I am dialing a correct extension, context!! |
| 18:15.38 | joako | can someone try to call sip:14193016245@sip02.netjdn.com ? |
| 18:15.45 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 18:17.17 | Ritzerisk | would anyone know if asterisk GUI 2.0 come with asterisknow by default or do i have to apt get it |
| 18:18.50 | codefreeze-lap | jameswf: if not sip, what tech, then? |
| 18:19.40 | jameswf | zap/dah |
| 18:20.13 | codefreeze-lap | jameswf: how many simul calls will there be at 600 cpi? |
| 18:20.48 | jeev | http://www.dreamhost.com/images/facetime-mikes.jpg LOL, that's their tech support? he looks 11. |
| 18:21.16 | jeev | oh, it's their childhood pics |
| 18:23.00 | Katty | jbot: ode to joy? |
| 18:23.21 | Katty | Qwell: teach jbot http://www.youtube.com/watch?v=xpcUxwpOQ_A |
| 18:23.42 | jameswf | I need 600 calls per second probably 3 seecond calls maybe 1800 |
| 18:23.49 | joako | Well I resolved my issue by disabling domain support |
| 18:26.07 | Katty | [TK]D-Fender: isn't it the cutest? http://www.flickr.com/photos/izaah/2956268549/in/set-72157608195837483/ |
| 18:26.15 | Katty | [TK]D-Fender: his left ears has come up today. |
| 18:26.20 | Katty | [TK]D-Fender: the right one still flops, tho. |
| 18:27.04 | cvnet | katty very cute, i love dogs, but rather BIG dogs |
| 18:27.28 | Katty | cvnet: riddick is only 11 weeks (= |
| 18:27.38 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 18:27.39 | Katty | cvnet: he's gonna be big :> |
| 18:27.49 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-89-142.w86-215.abo.wanadoo.fr) |
| 18:30.34 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:30.45 | hesco | using: cdr_pgsql, where would "'calldate' (be) specified more than once." ???? |
| 18:34.24 | hesco | damn: I just watched ode to joy from the youtube link above and epiphany grabbed my dsp and now kiax won't play sound. |
| 18:34.54 | hesco | one of these days I'm going to learn how to restart my alsa w/o having to shut down my browser |
| 18:36.58 | jameswf | wow I just dumped a years worth of images of my blackerry... strange... |
| 18:46.35 | hesco | I'm building an IRV menu. I have my incoming menu in place and the dispatch code is outlined and working. I'm using single digit extensions for this menu. If I start adding additional extensions with multi-digit extensions, but in a different context, will this break my one digit opening menu extensions? |
| 18:47.45 | [TK]D-Fender | hesco: No, it'll just mean that where things overlap you will have to wait at the end of your 1-digit entries to see if more digits come |
| 18:50.33 | hesco | if I set the timeout for those digits on a short leash for the IRV menu, will the context of that change be limited to that s extension or at least to that context? |
| 18:51.56 | Greek-Boy | Which ODBC driver is better for connecting to MS SQL Server? freetds or tdsodbc? |
| 18:53.14 | [TK]D-Fender | hesco: Any timeouts you set are the scope of the IVR, not of the items themselves. |
| 18:53.35 | [TK]D-Fender | hesco: IVR can see "X" number of different extens. timeouts apply to what you dial VS them. |
| 18:54.37 | hesco | 'scope of the IRV' does that mean the context the IRV is defined in? Or the entire extensions.conf? |
| 18:56.36 | *** join/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 18:56.40 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:57.02 | [TK]D-Fender | hesco: No, timeouts are set in your very dialplan and can vary at the point of exectution. You could have 500 extens in a context that are each technically "ivrs" |
| 18:57.23 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 18:57.56 | [TK]D-Fender | hesco: You can set timeout in 1 pattern match and then do 10 goto's and those timeouts are what's used unless overridden. |
| 18:58.30 | *** join/#asterisk Nate187 (n=Nate187@gw.bigrivertel.net) |
| 18:59.11 | hardwire | anybody auto-emailing from the asterisk dialplan? |
| 18:59.28 | jeev | hi hardwire. |
| 18:59.29 | [TK]D-Fender | hardwire: of what? |
| 18:59.31 | *** join/#asterisk rcy (n=rcy@S010600003981572c.vc.shawcable.net) |
| 18:59.40 | mog | hardwire, i have via system |
| 18:59.57 | hardwire | [TK]D-Fender: I have a fax server that sets itself to off-hook whenever it needs admin attention |
| 19:00.25 | hardwire | so I need to email via sendmail from the dialplan if the channel is available but just off-hook |
| 19:00.45 | hardwire | meh.. I'll just call a script that does it all |
| 19:00.53 | Nate187 | I use the system command and a script to email |
| 19:01.01 | hardwire | it needs to log back into the CLI anyways and do some zap parsing |
| 19:01.35 | jeev | murder on the government |
| 19:02.16 | Nate187 | is there anyway to prevent music on hold from restarting everytime? so it will cycle thru all of the files instead of callers hearing the first few seconds of the same file |
| 19:02.32 | hardwire | Nate187: yup |
| 19:02.35 | hardwire | you "tune" into it. |
| 19:02.36 | hardwire | :) |
| 19:03.05 | Nate187 | figured i'd have to go that route, was hoping asterisk alone would do it |
| 19:03.15 | hardwire | check out asterisk-addons |
| 19:03.15 | jeev | ok, serious business question here. i have an amazing idea that has to do with someone's capability and already set up business. what is the best way to protect myself from him stealing the idea and doing the plan with me? a NDA? |
| 19:03.20 | hardwire | compile rawplayer |
| 19:03.48 | hardwire | http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it |
| 19:04.28 | Nate187 | nice |
| 19:04.30 | hardwire | atleast I thought that worked |
| 19:04.33 | Nate187 | will try that out |
| 19:04.43 | Nate187 | i've already got everything as raw files anyway |
| 19:04.46 | hardwire | last time I had to deal with that issue (no clue how my MOH reacts atm) I used ESD |
| 19:06.00 | *** join/#asterisk hfb (n=hfb@pool-98-112-232-238.lsanca.dsl-w.verizon.net) |
| 19:06.47 | jeev | damnit |
| 19:06.57 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:07.28 | hardwire | jeev: what. |
| 19:09.52 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:11.16 | hardwire | pokes jeev.. where are you.. in the world? |
| 19:14.46 | Peaceful | Can anyone point me towards debugging problems with music-on-hold quality? |
| 19:15.31 | *** join/#asterisk tkbeat (n=tk@p54B96995.dip.t-dialin.net) |
| 19:16.05 | Peaceful | I've already simplified the music-on-hold by reducing it to just one song. |
| 19:16.09 | Katty | hai guys |
| 19:16.31 | Peaceful | I also converted the one song from mp3 to ulaw and wav, and deleted the mp3. |
| 19:16.57 | Peaceful | But the moh continues to stutter when lots of people are on hold |
| 19:17.13 | hardwire | Peaceful: what kind of asterisk system? |
| 19:17.17 | hardwire | got any zap hardware? |
| 19:17.48 | Peaceful | 1.2.30.2, two TE420b's, on a 8-core Xeon with 32GB RAM |
| 19:18.10 | Peaceful | SIP on the office side, all Zap to the outside world |
| 19:18.22 | hardwire | crap.. bbl.. meeting I forgot about |
| 19:18.28 | Peaceful | np |
| 19:18.33 | hardwire | Peaceful: you may have to send me your hardware for extra diagnosis. |
| 19:18.39 | hardwire | I may keep it a while. |
| 19:18.45 | hardwire | like.. a long while. |
| 19:18.48 | Peaceful | hardwire: lol :-P fat chance |
| 19:18.51 | hardwire | bbl |
| 19:18.54 | petererer | takes a while to diagnose :) |
| 19:19.27 | Peaceful | Anyone can get that setup, it's just under $10k! Seriously though... |
| 19:19.28 | petererer | what would an asterisk box actually do with all that ram... |
| 19:19.38 | Peaceful | ram disk for recordings |
| 19:20.00 | petererer | that's a lot of recordings |
| 19:20.35 | Peaceful | ya |
| 19:20.42 | Peaceful | 10GB/day in mp3's currently |
| 19:20.51 | Peaceful | hmm |
| 19:21.19 | Peaceful | so the stuttery call I'm listening to is not the _one_ song I reduced hold music to... Looks like I mixed up my classes... maybe it's an mp3 problem after all... |
| 19:24.07 | aiks[LV]_off | hardwire: and what ar you going to do with that box - analyze SETi signal? |
| 19:26.43 | Peaceful | hmm...so far, so good |
| 19:27.02 | dioedu | hello, in queues.conf, the parameter timeoutrestart doesn't treat NOANSWER event ? Just CONGESTION and BUSY ? |
| 19:27.08 | Peaceful | the mp3 files had reappeared thanks to subversion updates. Now that they're really gone, my one non-mp3 song seems non-stuttery |
| 19:28.00 | dioedu | if i have two agents with different priority and one of then doesn't answer, the call will ring forever... |
| 19:28.47 | [TK]D-Fender | dioedu: It'll only ring forever if you have no inter-agent dial timeout and/or no queue timeout |
| 19:29.03 | dioedu | in the agent with bigger priority |
| 19:29.37 | dioedu | i have a timeout configured in the queues.conf |
| 19:30.24 | dioedu | when i have this timeout in the agent with bigger priority, the call play again in the same agent. |
| 19:30.55 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 19:31.24 | dioedu | [TK]D-Fender, if i have same priority, this problem doesn't happen |
| 19:32.22 | dioedu | i think that priority can be used just with local channel's |
| 19:33.18 | dioedu | because in the local channel's i can treat each action, like, remove the agent from queue if i don't have answer |
| 19:33.24 | *** join/#asterisk zydoon (n=zydoon@41.225.158.64) |
| 19:33.40 | *** part/#asterisk zydoon (n=zydoon@41.225.158.64) |
| 19:33.56 | Katty | jbot: ode to joy |
| 19:34.00 | Katty | Qwell: :< |
| 19:35.07 | *** part/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 19:38.47 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
| 19:38.48 | Katty | jbot: are you broken? |
| 19:38.52 | Katty | pokes jbot |
| 19:40.13 | [TK]D-Fender | ~areyouadog ? |
| 19:40.13 | jbot | Bark! Bark! |
| 19:40.17 | [TK]D-Fender | yup! |
| 19:41.40 | Katty | :< |
| 19:41.48 | Katty | jbot needs to be more cultured. |
| 19:41.53 | jdnWEST | Anyone have an asterisk box at a colo where it interfaces with PRI's dirrectly? |
| 19:42.30 | [TK]D-Fender | jdnWEST: And how does "colo" PRI differ from the PRI smartjack 20ft behind me? |
| 19:45.07 | jdnWEST | D-Fender: none, exept that my PRI's ONLY come in on my fiber line here, and that didn't stay up durring IKE |
| 19:45.32 | jdnWEST | D=Fender: I want 1 PRI coming in over the internet, and routeing down any of the 4 different data pipes i have using BGP :) |
| 19:45.49 | [TK]D-Fender | jdnWEST: PRI has nothing to do with the internet... |
| 19:46.52 | jdnWEST | My trunk group has to come from a TWTC PRI to maintain caller ID, and If i'm going to have some redundency on my inc. calls I need my calls to route over the internet so... |
| 19:47.24 | jdnWEST | Sip trunk, then convert back to PRI (my switch dosn't support SIP.. well not a standard implementation of it) |
| 19:47.48 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 19:48.00 | [TK]D-Fender | jdnWEST: What switch? You are giving a very shattered description of your topography... |
| 19:48.12 | jdnWEST | something like this PRI --> Sip box -> Internet (2 fiber lines, wi-max, and data-T1) ---> SIP box --- PRI interface |
| 19:48.19 | jdnWEST | Is what I want |
| 19:48.25 | jdnWEST | I have a millinium EON phone switch |
| 19:48.35 | jdnWEST | with 4 PRI's currently coming off a private fiber ring |
| 19:49.01 | jdnWEST | We lost our ring durring the hurricane IKE, so I want to be able to pull 1 PRI over the internet (we kept our wi-max connection and our T1) |
| 19:49.02 | thomas | good afternoon. |
| 19:49.10 | [TK]D-Fender | jdnWEST: ou're connecting non SIP PBX's together with gateways basically? |
| 19:49.23 | thomas | is it normal if i shuting down my asterisk, all the subscriptions is gone? |
| 19:49.23 | jdnWEST | basically |
| 19:49.56 | [TK]D-Fender | jdnWEST: Ok, where does * factor into this? |
| 19:50.30 | jdnWEST | Can I use it to sip bridge, or is that more of a freeswitch thing? |
| 19:51.19 | jdnWEST | I need a PRI to come outa the wall in a safe place (far from the texas coast), and bridge it over the internet back into a PRI here in houston and keep the Caller ID's the same. |
| 19:51.38 | [TK]D-Fender | jdnWEST: As in use an * server per site to do the PRI>SIP conversion? |
| 19:51.56 | jdnWEST | yes |
| 19:52.17 | [TK]D-Fender | jdnWEST: Sure. |
| 19:52.32 | jdnWEST | or if anyone has any other sugestions, I figured the * channel is where i'd find the largest amount of VOIP people in the know :) |
| 19:53.25 | jdnWEST | D-Fender: i'm not really scared of CL, but is this going to be a pretty non-standard usage and require a bit of digging to pull off? |
| 19:54.54 | [TK]D-Fender | jdnWEST: If you're new to * then yes this will be a learning curve and your needs definitely best suited to non-GUI custom configs. |
| 19:54.57 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 19:54.57 | *** mode/#asterisk [+o file] by ChanServ |
| 19:58.46 | jdnWEST | D-Fender: I'd be setting this up as a backup for an emergency call center. Realistically is this going to give me the reliability, as well as how does it handle fax's and alpha pager handshakes asuming a VPN with QOS on the routers to manage bandwidth? |
| 20:00.50 | [TK]D-Fender | jdnWEST: Pretty much toss faxing out the window. |
| 20:00.51 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:01.13 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 20:02.02 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 20:04.20 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 20:07.29 | jdnWEST | D-Fender: hmmm I guess I can have fixed channels on the local PRI's for faxing, alpha pagers though have got to be able to work.. |
| 20:07.30 | jeev | anyone see anything wrong with this? "<digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-3]xx" dialplan.digitmap.timeOut="3"/>" |
| 20:07.55 | diegows | after read "core show application dial" I don't undertand how can I wait for some digits on the called side before connect the caller |
| 20:07.58 | diegows | :( |
| 20:12.56 | aiks[LV]_off | diegows: i guess you dont do it with dial |
| 20:13.01 | Greek-Boy | Which ODBC driver is better for connecting to MS SQL Server? freetds or tdsodbc? |
| 20:13.04 | aiks[LV]_off | at least I never do |
| 20:13.04 | [TK]D-Fender | diegows: ont he CALLED side? Your previous weak description implied the calling side. |
| 20:14.05 | diegows | yeah |
| 20:14.10 | aiks[LV]_off | diegows: you mean like entering the code for accepting the incomming call? |
| 20:14.18 | [TK]D-Fender | diegows: M() <-- read up |
| 20:14.33 | [TK]D-Fender | diegows: And don't even dream of implementing it as an IVR in there, use "Read" |
| 20:15.04 | diegows | the called side must press '1', then on the caller side the music stop and then both sides are linked |
| 20:15.07 | aiks[LV]_off | [TK]D-Fender:"as an IVR" - you mean through dialplan? |
| 20:15.20 | aiks[LV]_off | through extensions that is |
| 20:15.47 | aiks[LV]_off | diegows: if i am not very mistaken that could be done by queue configuration |
| 20:15.52 | diegows | [TK]D-Fender: i tried with M(), but the music stops when the macro is executed |
| 20:15.59 | aiks[LV]_off | but i might be off the target |
| 20:16.09 | diegows | aiks[LV]_off: the queue stops the music then the extensions is answered |
| 20:16.13 | diegows | and I don't want that |
| 20:16.34 | [TK]D-Fender | diegows: Wait this is during a QUEUE now? |
| 20:16.45 | diegows | [TK]D-Fender: no, no |
| 20:16.47 | [TK]D-Fender | diegows: use the "context=" for your queue to set the exit context |
| 20:17.01 | diegows | [TK]D-Fender: aiks[LV]_off suggested a queue |
| 20:17.22 | [TK]D-Fender | diegows: Make up your mind, I'm getting tired of running in circles |
| 20:18.09 | diegows | i want a simple extension |
| 20:18.12 | diegows | no queues |
| 20:18.39 | [TK]D-Fender | diegows: to do what? |
| 20:18.43 | diegows | this extension is an gsm gateway |
| 20:19.03 | diegows | when you call it, during the call there is silence for about 20-30 seconds |
| 20:19.37 | diegows | so, I wait to keep the caller with music |
| 20:19.52 | diegows | after the destination on the mobile phone press '1' |
| 20:19.59 | aiks[LV]_off | [TK]D-Fender: - it looked like queue scenario to me |
| 20:20.07 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:20.10 | aiks[LV]_off | but i was jumping to conclusions |
| 20:20.25 | [TK]D-Fender | diegows: that would be the CALLEE. |
| 20:20.36 | diegows | ok, sorry |
| 20:20.45 | [TK]D-Fender | diegows: not the CALLER. Please get your head straight about this. |
| 20:20.50 | diegows | ok ok |
| 20:20.59 | [TK]D-Fender | diegows: And that would be the "M()" option for dial. |
| 20:21.27 | [TK]D-Fender | diegows: And there is another option if you want MoH instead of ringing. Go read the instructions for it |
| 20:21.43 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 20:21.51 | aiks[LV]_off | off for a smoke and then staright to dreamland |
| 20:22.02 | aiks[LV]_off | its been a long, long, long day |
| 20:22.20 | aiks[LV]_off | tmr wont be any better. |
| 20:23.13 | *** join/#asterisk Kate-o (n=vajda1ka@unaffiliated/kateo/x-334924) |
| 20:23.23 | aiks[LV]_off | cu everyone. |
| 20:24.48 | [TK]D-Fender | checkout time here as well. BBL |
| 20:25.27 | *** join/#asterisk n3hxs (n=HAMming@static-71-242-245-74.phlapa.east.verizon.net) |
| 20:27.09 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 20:27.24 | *** part/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 20:29.44 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 20:33.09 | *** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 20:33.30 | devilsoulblack | hi |
| 20:34.01 | Kate-o | hi |
| 20:34.10 | devilsoulblack | any one get this msg "Sending Set Asynchronous Balanced Mode Extended" |
| 20:35.51 | seanbright | yes |
| 20:35.57 | seanbright | PRI related, right? |
| 20:36.00 | jeev | hey guys, i'm having an issue with my dial plan. [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-3]xx , extensions are in the 200's and 300's, but if i call anything locally with 7 digits, it dials the extension |
| 20:37.13 | devilsoulblack | yes ist a PRI |
| 20:37.20 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:38.43 | Nate187 | try [2-3]xxT |
| 20:38.57 | jeev | hm |
| 20:39.19 | Nate187 | i'm guessing whatever you dial is matching that first |
| 20:39.21 | jeev | hm |
| 20:39.28 | jeev | so would i do |
| 20:39.42 | jeev | [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxxxx|[2-3]xxT |
| 20:40.14 | Nate187 | you'll have a similar problem later: |
| 20:40.27 | Nate187 | anything that matches [2-9]xxxxxxxxx |
| 20:40.33 | Nate187 | will first match [2-9]xxxxxx |
| 20:40.55 | jeev | hmm |
| 20:41.02 | jeev | how shall i do it? just tell them to dial 1818 ? |
| 20:41.19 | Nate187 | is this for a pbx? |
| 20:41.43 | jeev | whta do you mean? i'm gonna drop it in sip.cfg and make the phones update. it's a bunch of polycom's. |
| 20:42.25 | Nate187 | could set them up to dial 9 first for outside calls |
| 20:44.29 | *** join/#asterisk jjshoe (i=jjshoe@cpe-76-175-157-237.socal.res.rr.com) |
| 20:44.57 | jeev | that'l annoy them |
| 20:45.26 | devilsoulblack | any one get this msg "Sending Set Asynchronous Balanced Mode Extended" over PRI ? |
| 20:46.01 | Nate187 | you're kinda stuck between dialing 9 first or having to wait for the inter digit timeout to expire |
| 20:46.18 | lesouvage | I just set up a meetme() room. Sound quality of the moh and the voices is ok but the announcement (you are the first etc......) is crap. What could be the reason for this bad soudquality of the announcement? |
| 20:46.19 | Nate187 | but if they don't mind the wait, use [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxxxxT|[2-3]xxT |
| 20:46.43 | jeev | i think i'll just let them dial 1818 if they want to call anything in the same area code that starts with 2 or 3, they'll have to dial 1818 first i guess. or i can ask them if 9 wont bother them |
| 20:47.56 | jeev | they prefer 9 i guess |
| 20:48.01 | Nate187 | the best way i've found to implement changes like that is to not give them the option |
| 20:48.09 | lesouvage | jeev: use xxxxxxxxxxxxx.T and see if that acts like you want. I have been messing around with this and changed it in something like this. I don't have access to the phones right now. |
| 20:48.14 | Nate187 | say "we have upgraded the phones to comply with a national standard adn you have to dial 9 for outside calls now" |
| 20:48.20 | jeev | no, i asked |
| 20:48.50 | jeev | they said it's ok! |
| 20:49.09 | lesouvage | jeev: this way you can set your dialpatterns in Asterisk. |
| 20:49.46 | jeev | my extensions.conf file needs a make over like russellb needs one asthetically |
| 20:50.20 | jeev | hey, i wonder if i can do PA with these polycom 330's and asterisk |
| 20:50.35 | lesouvage | You mean russelb himself?? |
| 20:50.57 | Nate187 | i just set up a feature so that when i get a sales call or a surve i say hang on, hit #9, and rickroll them |
| 20:51.01 | Nate187 | asterisk is great |
| 20:51.16 | jeev | yes! |
| 20:51.30 | jeev | rickroll is wack! |
| 20:52.00 | jeev | my stolen cable modem's firmware was like like the html file and it took arguments, this one guy tinyurl'd me a link that made it download firmware from somewhere and it erased my flash. now that's rick rolling! |
| 20:53.36 | Nate187 | that sucks |
| 20:53.54 | jeev | :> |
| 20:54.18 | Katty | so bored. |
| 20:54.36 | jeev | live sex show. |
| 20:54.37 | jeev | now? |
| 20:54.51 | Katty | gosh. |
| 20:54.59 | Katty | simmer down now. |
| 20:55.08 | jeev | why |
| 20:56.37 | Katty | because it's not nice. |
| 20:56.37 | jeev | bah |
| 20:56.44 | jeev | it's not nice to admit you have boobies and not show them. |
| 20:56.56 | Katty | frowns |
| 20:57.01 | Katty | <bkw> NEXT!!!! |
| 20:57.03 | hardwire | whats going on here?! |
| 20:57.23 | hardwire | looks outside and notes it's not spring break.. |
| 20:57.26 | hardwire | NOT BY A LONG SHOT |
| 20:57.31 | Katty | no, no it's not. |
| 20:57.38 | Katty | And the girls gone wild van is nowhere in sight. |
| 20:57.38 | russellb | jeev: stfu, or i will ban you again |
| 20:57.46 | hardwire | jeev: again? |
| 20:57.51 | jeev | damnit |
| 20:57.57 | hardwire | thinks jeev should behave :) |
| 20:58.05 | hardwire | GOOD BOYS AND GIRLS GO TO HEAVEN |
| 20:58.06 | Katty | things russellb needs a hug (= |
| 20:58.10 | Katty | applies hugging to russellb |
| 20:58.20 | Katty | file: you can haz hug too? |
| 20:58.26 | russellb | Katty: thanks :) |
| 20:58.27 | *** join/#asterisk RachedTN (n=rached@41.224.183.112) |
| 20:58.31 | Qwell | :( |
| 20:58.32 | hardwire | err. |
| 20:58.34 | Qwell | I see how it is |
| 20:58.36 | Katty | :< |
| 20:58.37 | hardwire | Qwell next |
| 20:58.38 | hardwire | then me |
| 20:58.40 | Katty | hugs Qwell |
| 20:58.58 | hardwire | ahem.. I DEFENDED YOUR HONOR! |
| 20:59.13 | Katty | that does not mean you get unsocliciated hugs. |
| 20:59.17 | hardwire | yes. |
| 20:59.22 | hardwire | I'm cute as a bugs nose.. |
| 20:59.28 | Qwell | his was solicited |
| 20:59.31 | hardwire | people hug me and they don't even know they are doing it. |
| 20:59.35 | Katty | Qwell: oh. |
| 20:59.38 | Katty | Qwell: right. |
| 20:59.46 | Katty | Qwell: ;) |
| 20:59.53 | Qwell | so was mine though :p |
| 20:59.56 | hardwire | no hug solicitors?! |
| 20:59.58 | Katty | pats Qwell |
| 21:00.03 | Katty | Qwell: that's because i love you, dear. |
| 21:00.57 | hardwire | http://www.cultureandrecreation.gov.au/articles/weather/ |
| 21:01.09 | hardwire | I still think that is some sort of conspiracy by the tourist agencies. |
| 21:01.14 | hardwire | stupid tilt. |
| 21:01.40 | Katty | searches for dahdi compiling instructions |
| 21:02.29 | Katty | jbot: dahdi? |
| 21:02.30 | jbot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 21:03.27 | Katty | russellb: http://svn.digium.com/view/asterisk/tags/1.6.0-rc4/Zaptel-to-DAHDI.txt?view=co 404 :< |
| 21:03.29 | hardwire | is at the grocery store |
| 21:03.43 | russellb | Katty: sorry, view is disabled right now |
| 21:03.47 | hardwire | Katty: something wonky happened to svn |
| 21:03.48 | russellb | Katty: s/view/svn/ |
| 21:03.54 | hardwire | I tried to view the audiobridge stuff.. for the fail |
| 21:04.00 | hardwire | just downloads it |
| 21:04.05 | Katty | russellb: kay. |
| 21:06.21 | jeev | wtf, the dialplan wont update in sip.cfg |
| 21:06.22 | jeev | what is happening |
| 21:06.46 | Katty | oh look! |
| 21:06.50 | Katty | directions in the readme! |
| 21:06.51 | Katty | ;) |
| 21:07.33 | *** join/#asterisk Deeewayn1 (n=Deeewayn@nat/digium/x-a81aa89a5aad4bcd) |
| 21:07.33 | *** mode/#asterisk [+o Deeewayn1] by ChanServ |
| 21:08.21 | Katty | eyes Deeewayn1 |
| 21:08.46 | *** join/#asterisk PeterFA (n=Peter@unaffiliated/peterfa) |
| 21:08.55 | Deeewayn1 | waves to Katty |
| 21:09.14 | Katty | so the compile order is libpri, dahdi-linux, dahdi tools, asterisk, addons, wanpipe |
| 21:09.27 | Katty | or some other order? |
| 21:09.33 | Katty | Deeewayn1: herro, stranger. |
| 21:09.36 | PeterFA | How long will it take to configure an asterisk server to support 11 lines? |
| 21:09.39 | *** join/#asterisk mohawk (n=mohawk@host217-40-110-154.in-addr.btopenworld.com) |
| 21:09.40 | PeterFA | And how do I connect out? |
| 21:09.46 | Deeewayn1 | my nick is sad :-( |
| 21:09.53 | Katty | Deeewayn1: do you have another one? |
| 21:09.57 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:10.17 | Katty | PeterFA: considering how vaguely your question was... i'd say it take you a few days. |
| 21:10.19 | Deeewayn1 | I am also d3wayne |
| 21:10.25 | Katty | hmm. |
| 21:10.29 | Deeewayn1 | but he is asleep |
| 21:10.35 | Katty | oh, i see. |
| 21:10.59 | Katty | Qwell: order: libpri, dahdi-linux, dahdi tools, asterisk, addons, wanpipe? |
| 21:11.16 | *** join/#asterisk write__erase (n=write@royale.aixmarseille.com) |
| 21:11.17 | Deeewayn1 | Katty: I do the first 4 in that order |
| 21:11.23 | Katty | mkay. |
| 21:11.27 | Katty | here i go then! |
| 21:11.29 | Katty | puts on helmet |
| 21:11.32 | Deeewayn1 | good luck |
| 21:11.45 | Deeewayn1 | puts an Asterisk sticker on Katty's helmet |
| 21:12.03 | Katty | oh thanks ;) |
| 21:15.12 | hardwire | PeterFA: there's lots of books out there that answer that |
| 21:15.20 | hardwire | and lots of people here that stand to profit if you buy one. |
| 21:15.25 | hardwire | well.. two or so. |
| 21:15.41 | hardwire | ~book |
| 21:15.42 | jbot | it has been said that book 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://tfot.leifmadsen.com or see ~buybook |
| 21:15.50 | write__erase | Hi .... At the moment I'm only experienced with Asterisk/SIP. One of my customer want an Asterisk deployement with SCCP cisco . Is it working properly, in a production environement ? |
| 21:15.54 | PeterFA | hardwire, well, supporting Asterisk helps out a lot of people. |
| 21:16.17 | hardwire | PeterFA: I support it! |
| 21:16.18 | hardwire | :) |
| 21:16.35 | hardwire | I'll support you too, my biggest support method is G.I. Joe style RTFM'ing.. |
| 21:16.38 | hardwire | Knowledge is power! |
| 21:16.47 | hardwire | Can you dig it?! |
| 21:16.53 | hardwire | wait.. did I just go slim jim? |
| 21:17.35 | Katty | dangit. i forgot my headers. |
| 21:17.40 | Katty | why do i always forget my headers. |
| 21:17.49 | Katty | makes postit |
| 21:18.02 | hardwire | helmet was in the way |
| 21:18.31 | Katty | hardwire: Dark Helmet |
| 21:20.05 | Katty | oh. it gets stuff. |
| 21:20.06 | Katty | how fancy |
| 21:20.18 | Katty | claps |
| 21:20.26 | hardwire | Katty: you spaceballzing me? |
| 21:20.38 | *** join/#asterisk X-Rob (n=rob-x@CPE-124-187-88-57.qld.bigpond.net.au) |
| 21:21.22 | Katty | well it was funny ;) |
| 21:22.44 | Katty | hardwire: you've been JAMMED |
| 21:23.27 | hardwire | raspberry! |
| 21:23.38 | hardwire | Only ONE PERSON in this universe would dare jam me with raspberry! |
| 21:23.45 | Katty | hehe! |
| 21:23.58 | Katty | Dahdi didn't asplode! |
| 21:24.09 | Katty | asterisk is taking forever, tho :< |
| 21:24.18 | hardwire | -j ? |
| 21:24.26 | Katty | make |
| 21:24.31 | hardwire | Katty: how many leenux boxes can you use for compiling? |
| 21:24.36 | hardwire | distcc is awesome |
| 21:24.43 | Katty | we are a very small operation |
| 21:24.49 | Katty | i hand compile every machine (= |
| 21:24.59 | hardwire | where is we? |
| 21:25.01 | Katty | they come out better... like homemade cookies. |
| 21:25.12 | Katty | missouri (= |
| 21:25.14 | hardwire | do you put little buttons on them? |
| 21:25.25 | Katty | no, but i put stickers on them |
| 21:25.31 | Katty | does that count? |
| 21:25.31 | hardwire | stickers on the cookies? |
| 21:25.33 | hardwire | wtf. |
| 21:25.36 | Katty | on the case!!! |
| 21:25.37 | Katty | geeze |
| 21:25.40 | Katty | sighs |
| 21:25.43 | hardwire | sighs |
| 21:25.47 | Katty | teehee |
| 21:25.59 | Katty | why would i put buttons on a cookie? |
| 21:26.33 | PeterFA | How do I connect an Asterisk box to where it can call out? |
| 21:27.06 | Katty | wow. |
| 21:27.09 | Katty | Nate187 |
| 21:27.11 | Katty | i think i know him |
| 21:27.12 | PeterFA | Well, I know I can connect to a carrier online. |
| 21:27.18 | Katty | they're just down the street at the telco |
| 21:27.19 | PeterFA | But how would I connect to a local phone company. |
| 21:27.35 | Katty | PeterFA: you do not parse. |
| 21:28.03 | Katty | russellb: WHAT, asterisk now has phoneprovision samples?! |
| 21:28.06 | Katty | russellb: this is madness! |
| 21:28.12 | hardwire | PeterFA: private message me |
| 21:28.17 | russellb | Katty: why yes it does |
| 21:28.21 | hardwire | I'll save this channel from your awesome newbieness :) |
| 21:28.23 | Katty | russellb: most excellent. |
| 21:28.27 | Katty | russellb: you get a gold star! |
| 21:28.39 | russellb | heh, i didn't do that |
| 21:28.44 | russellb | but i'll take the star anyway |
| 21:28.58 | Katty | well who did it? |
| 21:29.40 | russellb | terry wilson did most of the work for res_phoneprov (otherwiseguy on IRC) |
| 21:30.53 | Katty | russellb: i'll give him a gold star next time i see him. |
| 21:35.37 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:35.37 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 21:36.24 | Katty | anyone here work for sangoma? |
| 21:36.38 | lmadsen | if you were going to compare the asterisk dialplan syntax to a language (sytnax wise) which would you compare it to? The reason I ask is because I've just installed GeSHi which is a syntax highligher, and wonder if anyone might know which one looks the best with dialplan syntax? |
| 21:37.01 | jeev | my phone wont update it's digitmap from sip.cfg, wtf |
| 21:37.17 | jeev | lmadsen, a new syntax should be called 'jeev' |
| 21:37.41 | Katty | lmadsen: can't say i've ever used that before. |
| 21:37.42 | lmadsen | jeev: easy to make as all you need to highlight is the question mark |
| 21:37.53 | lmadsen | Katty: ya, using a plugin for wordpress |
| 21:38.12 | Katty | that's neat. |
| 21:38.26 | Katty | about as neat as this wanpipe tgz 404ing |
| 21:38.27 | Katty | pouts |
| 21:39.10 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 21:39.10 | thomas | how i can with snom pickup? |
| 21:39.23 | thomas | have the extention: <sip:11@192.168.100.1;user=phone>|*8 |
| 21:39.24 | thomas | wrong? |
| 21:41.08 | Katty | check your features |
| 21:41.12 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:41.13 | Katty | i think *8 is default tho |
| 21:41.46 | thomas | Katty: I Would |
| 21:55.38 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 21:56.03 | hardwire | your welcome. |
| 21:57.39 | jdnWEST | Anyone have a sip client to recomend on the iphone? |
| 21:59.14 | russellb | fring .. |
| 22:01.51 | hardwire | that should be part of the topic |
| 22:01.58 | *** join/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 22:02.11 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:02.28 | jeev | kicks russellb |
| 22:02.51 | RachedTN | hello, I have installed the wrong version of zaptel and libpri, now i wanna to remove them, but i really don't know how to do it correctly ! |
| 22:03.01 | hardwire | jeev: you're angsty.. I've found. |
| 22:03.06 | jtodd | hardwire: I think sipgate.de made one. |
| 22:03.42 | hardwire | I don't even have an iPhone |
| 22:03.52 | hardwire | but I played with sipgate for wince.. nyeh. |
| 22:04.02 | hardwire | maybe it was the phone I was using.. but.. nyeh |
| 22:04.49 | jtodd | oh, sorry - I meant "jdnWEST" - didn't read the lines closely enough. |
| 22:05.27 | hardwire | squint harder |
| 22:06.12 | RachedTN | ping hardwire, I have installed the wrong version of zaptel and libpri, now i wanna to remove them, but i really don't know how to do it correctly, i tried apt-get remove --purge libpri (i am on debian etch), but it says : Couldn't find package libpri |
| 22:06.31 | *** join/#asterisk ManxPower (n=manxpowe@107.sub-70-223-177.myvzw.com) |
| 22:06.48 | hardwire | followed with "ask hardwire on #asterisk" ? |
| 22:07.21 | hardwire | RachedTN: `dpkg -l | grep libpri` |
| 22:07.31 | hardwire | I think it's libpri1 or 2 or eleventy7. |
| 22:08.01 | hardwire | or just `apt-get --purge remove ^zaptel.* ^libpri.*` |
| 22:08.21 | hardwire | I forget if it uses * or + |
| 22:08.30 | RachedTN | OK, thanks, sorry to interrupt your conversation ;) |
| 22:08.35 | hardwire | you didn't |
| 22:08.43 | hardwire | jtodd is just some freak. |
| 22:08.47 | hardwire | but you caught me off guard |
| 22:08.55 | hardwire | next time my ninja senses will find you first. |
| 22:09.23 | RachedTN | lol |
| 22:09.45 | hardwire | you figure it out? |
| 22:10.09 | hardwire | cause I'm still trying to figure out what the wrong versions of zaptel and libpri are :) |
| 22:10.19 | hardwire | are you compiling asterisk on etch? |
| 22:10.58 | StephenF[W] | anyone using Polycom phones, how do you normally program a receptionist phone (im using 650) for incoming calls? Do you just have all incoming calls goto the reception extension and display like 6 line keys for that extension... or maybe create another extension for incoming calls and register the phone to both the incoming extension and the local extension... |
| 22:11.11 | StephenF[W] | I dunno, im looking for best practices. What works for you |
| 22:11.30 | hardwire | usually you have the reception in a queue |
| 22:11.34 | *** join/#asterisk stevej678 (n=steve@202.7.249.15.dynamic.rev.aanet.com.au) |
| 22:11.48 | hardwire | that way other phones can act as reception very easily |
| 22:12.06 | *** join/#asterisk LINKER3000L (n=chatzill@78.32.25.201) |
| 22:12.07 | StephenF[W] | ahh, so if there isnt a receptionist for some reason, the other phones can pick up |
| 22:12.07 | hardwire | thent he queue handles the waittime and average hold time announcements if you want those.. or just music on hold |
| 22:12.09 | RachedTN | yes i am compiling asterisk on etch |
| 22:12.14 | hardwire | once the receptionist hangs up she gets another call |
| 22:12.37 | hardwire | StephenF[W]: sometimes, just to be evil, I set up four sip lines on a phone like a reception phone |
| 22:12.41 | hardwire | then join each sip line to a queue |
| 22:12.42 | RachedTN | i have compile asterisk on fedora 7, and it works nice |
| 22:12.55 | StephenF[W] | haha, 4 simultaneous calls |
| 22:12.58 | hardwire | that way she can juggle them if she wants, and always get the call on the top |
| 22:13.01 | RachedTN | now i am trying to do the same thing in debian etch |
| 22:13.02 | hardwire | you dig? |
| 22:13.13 | lesouvage | SephanF[W]: you could ad a queue to the system, make the receptionist member of the queue and let the calls roll in. If it is very crowded you just promote some of the staff to be the second receptionist so they can together handle the incoming calls. |
| 22:13.13 | StephenF[W] | right |
| 22:13.25 | hardwire | any queue member that answers a queue, even if it's the same phone, always gets the next actual person that called in. |
| 22:13.35 | StephenF[W] | ok so sounds like a queue is the way to go |
| 22:13.43 | hardwire | you bet your sweet ass pants it is. |
| 22:13.54 | lesouvage | yes |
| 22:13.56 | StephenF[W] | so what would you display on the reception phone? Just 4 virtual line keys? |
| 22:13.59 | StephenF[W] | lol |
| 22:14.12 | StephenF[W] | i will never bet my sweet ass pants, I cant risk that |
| 22:14.44 | hardwire | then set up an extension for other phones to call if they want to dynamically add their extension to the queue, temporarily. |
| 22:14.53 | StephenF[W] | then on the phone's sidcar Im thinking so BLF for key staff members, so the receptionist can easily see their status and route calls to them |
| 22:15.04 | StephenF[W] | hardwire nice, i like it |
| 22:15.29 | hardwire | StephenF[W]: other people here might know, but I think you need 4 seperate sip accounts if you want to do that |
| 22:15.46 | *** part/#asterisk RachedTN (n=rached@41.224.183.112) |
| 22:15.55 | hardwire | I'm pretty sure adding the same sip line in the queues.conf 4 times won't work |
| 22:15.57 | hardwire | but I could be wrong |
| 22:16.01 | lesouvage | SephanF[W]: BLf is fine and nice and blinking lights are really something but my advice is to go one thing at a time. You started with a qeueu. I would start with blinking lights if the queue is up and running. |
| 22:16.10 | hardwire | it may use that as a "is channel in use > count in queues.conf" logic. |
| 22:16.23 | StephenF[W] | hardwire, they are just virtual lines on the phone itself. four virtual buttons for the same extension |
| 22:16.38 | hardwire | you may need to change that to 4 actual sip lines |
| 22:16.39 | StephenF[W] | lesouvage I agree, im just trying to get the big picture so I know where to start and where im going |
| 22:16.50 | hardwire | SIP/00101 through SIP/00104 |
| 22:17.03 | StephenF[W] | hardwire oh you mean for it to work in the queue |
| 22:17.05 | hardwire | but that may suck for other reasons |
| 22:17.09 | hardwire | StephenF[W]: yeh |
| 22:17.12 | StephenF[W] | gotcha |
| 22:17.23 | hardwire | but it would suck if somebody called sip/00101 internally and it went nowhere. |
| 22:17.28 | hardwire | I guess you're gonna have to play |
| 22:17.37 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 22:17.42 | lesouvage | StephenF[W]: Asterisk is taking care of the waiting of the incoming lines and routing them to free agents. Agents should have 1 line. It doesn't make sense to have in incoming line moved from the queue to an unanswered line on a phone. |
| 22:17.56 | hardwire | Rached left. |
| 22:18.19 | hardwire | lesouvage: except if the agent is supposed to answer multiple lines :) |
| 22:18.23 | StephenF[W] | lesouvage, yeah but its not really a "call center" with agents. Its a receptionist, so she should see all the incoming calls even if she is already on one |
| 22:18.25 | hardwire | and be part of a larger queue |
| 22:18.37 | lesouvage | hardwire: you can just answer one line at a time |
| 22:18.49 | StephenF[W] | lesouvage but there is Hold |
| 22:18.52 | hardwire | lesouvage: but not if she is a queue member |
| 22:18.53 | hardwire | afaik |
| 22:19.19 | StephenF[W] | i dunno |
| 22:19.22 | hardwire | lesouvage: maybe you can clear this up.. if a queue member has 4 lines of "availabilty" over one sip account, will their phone ring for each new queue memeber? |
| 22:19.36 | hardwire | each new incoming call |
| 22:19.37 | hardwire | sorry |
| 22:19.43 | lesouvage | StephenF[W]: that will drive the receptionist crazy |
| 22:19.46 | hardwire | somebody smart please answer. |
| 22:19.48 | StephenF[W] | hehe |
| 22:19.51 | hardwire | lesouvage: our reception does it fine. |
| 22:19.58 | hardwire | blah blah blah, please hold |
| 22:19.58 | ManxPower | hardwire: I imagine they would. That would not be the case if each line appearance was a different sip account |
| 22:19.59 | hardwire | blah blah blah, please hold |
| 22:20.16 | StephenF[W] | lesouvage so you think, all calls should get MOH until the receptionist finishes with her current call? |
| 22:20.17 | lesouvage | hardwire: maybe it does, but it doesn't make sense. |
| 22:20.25 | lmadsen | hardwire: it will unless you have hints and call-limit enabled which will allow asterisk to know the device status, and as long as the channel status updated to InUse, then Queue() won't |
| 22:20.32 | hardwire | ManxPower: so your thinking one sip account would ring multiple times if it was part of a queue? |
| 22:20.48 | hardwire | errr multiple concurrent times |
| 22:21.03 | hardwire | lmadsen: danke danke |
| 22:21.13 | hardwire | StephenF[W]: so.. you don't need the extra fancy lines like I was thinking |
| 22:21.13 | StephenF[W] | lesouvage, what if she/he is on the line with a vendor and a customer calls in. She should be able to put the vendor on hold to tend to the customer, right? |
| 22:21.19 | lesouvage | Setphen: if her only job is to transfer calls, she has to that one by one. |
| 22:21.26 | ManxPower | hardwire: try it an see. I don't know the exact method queue uses to determine if a device is in use. |
| 22:21.33 | ManxPower | but it IS the case for normal Dial |
| 22:21.38 | StephenF[W] | yeah, i forsee some trial and error here ;) |
| 22:21.46 | lesouvage | StephenF[W]: it sound like your reception is understaffed ;-) |
| 22:22.02 | hardwire | lesouvage: if my reception was only handling one call at a time, nothing would get done. |
| 22:22.07 | StephenF[W] | lesouvage, um its a small office of 5 people. her job does not consist of only transfering calls |
| 22:22.11 | ManxPower | one of the many reasons I do NOT recommend having the same SIP account for all the call appearances |
| 22:22.27 | hardwire | maybe in europe you guys just sit around and smoke your fancy cigars and let the phones ring.. |
| 22:22.31 | StephenF[W] | ManxPower so you would just have the phone register multiple sip accounts? |
| 22:22.32 | hardwire | :P |
| 22:22.52 | ManxPower | hardwire: I have never needed Queues, I tried them, found them overly complex and confusing and wrote my own "queueing" system in the dialplan |
| 22:23.00 | StephenF[W] | lesouvage I could see that in a big office where there are multiple "operators" but not in a small office |
| 22:23.05 | lmadsen | ManxPower: I've also done that :) |
| 22:23.05 | hardwire | ManxPower: I agree with you 100% on having seperate extensions for agent/queue members |
| 22:23.05 | ManxPower | StephenF[W]: one account per line appearance. |
| 22:23.08 | lesouvage | hardwire: What does the receptionist say in the phone. All in queue just wait a moment. I guess not, see is talking to them one by one. |
| 22:23.16 | hardwire | ManxPower: that lets you put them in a context where they can't dial out |
| 22:23.19 | hardwire | aka forward |
| 22:23.31 | hardwire | I freaking hate when agents forward to their cellies. |
| 22:24.04 | StephenF[W] | ManxPower, right so for example. We have 4 sip trunks. So in theory we could have 4 incoming calls. I might want to register 4 line appearences on the reception phone so she could handle all four trunks |
| 22:24.06 | hardwire | StephenF[W]: aren't you glad you asked? |
| 22:24.26 | StephenF[W] | no its good, i like the different opinions |
| 22:24.41 | hardwire | StephenF[W]: queues are the scalable solution, or using an IVR |
| 22:24.42 | lesouvage | StephenF[W]: if you ust have a small office with sometimes 2,3 and once a month 4 calls at the same time just have 4 or 6 lines on the phone and press the line that you want to answer. |
| 22:24.56 | hardwire | but queues make it simple to handle 8 calls going to one person. |
| 22:24.58 | StephenF[W] | lesouvage thats what im thinking |
| 22:24.59 | lesouvage | No need for a queue |
| 22:25.20 | StephenF[W] | lesouvage, unless we want other uses to answer the phones when she goes to lunch |
| 22:25.22 | lesouvage | It all depends on what is actually happening. |
| 22:25.23 | hardwire | lesouvage: for 1:1, probably not |
| 22:25.37 | hardwire | StephenF[W]: she could also just forward the phone to somebody |
| 22:25.43 | StephenF[W] | hmm, good point |
| 22:25.46 | hardwire | if she had multiple lines to her phone she'd have to do it for each one. |
| 22:25.49 | hardwire | so yeh. |
| 22:25.52 | StephenF[W] | yuck |
| 22:25.52 | hardwire | have phun! |
| 22:25.54 | StephenF[W] | lol |
| 22:25.57 | hardwire | blog on your results. |
| 22:26.05 | StephenF[W] | seriously |
| 22:26.13 | hardwire | I'm hanging out at a grocery store today |
| 22:26.17 | hardwire | it's .. very awesome |
| 22:26.29 | hardwire | sitting here next to a coffee place, sipping my joe.. taking in all the hotties |
| 22:26.41 | hardwire | I'm engaged.. but it's nice to know Anchorage still has hotties running around |
| 22:26.57 | lesouvage | harwire: she ould just have one sip number or hotdesk on her phone, no need to forward every line available (line is a non voip idea concept) |
| 22:27.03 | hardwire | saw a few cougars earlier.. broke eye contact. |
| 22:27.24 | hardwire | lesouvage: lines are kinda quirky in voiplandia, your right |
| 22:27.31 | hardwire | but they are very handy for managing people |
| 22:27.56 | hardwire | having a DID routed to her first line which allows her call out access is perfect for forwarding her non-agent direct calls. |
| 22:28.11 | hardwire | having another line set up to be part of a queue which has a different dialplan context will stop her from forwarding calls. |
| 22:28.19 | hardwire | her being the receptionist.. it could be a he.. but I like hers more. |
| 22:28.47 | *** join/#asterisk legis (n=wadsack@unaffiliated/legis) |
| 22:28.58 | hardwire | I've noticed that sip phones don't set a required variable which counts redirects and deflections |
| 22:29.22 | hardwire | which should be used for denying forward access from the dialplans logic. |
| 22:29.25 | lesouvage | hardwire: funny, I heard today " the receptionist (lady) is bsy please wait, and I got a man on the line. (the line was in Dutch) |
| 22:29.41 | hardwire | lesouvage: it may have still been a lady. |
| 22:29.59 | hardwire | they are heavy smokers over there eh.. :) |
| 22:30.02 | lesouvage | hardwire: it was definitly not a lady. |
| 22:30.34 | hardwire | lesouvage: lets refer to them as receptionistas and receptionistos |
| 22:30.43 | lesouvage | OK |
| 22:30.47 | hardwire | excellent |
| 22:30.53 | hardwire | that's all. |
| 22:31.04 | lesouvage | And what if we have more then one with diffent genders? |
| 22:31.11 | lesouvage | different |
| 22:31.21 | hardwire | receptionistes |
| 22:31.58 | legis | Hi, I have a SIP to trunk to my provider with *, the trunk is a route inside their network(no internet), I need to used that SIP from the Internet, how can I acomplish this? |
| 22:32.26 | legis | was thinking of puting a second asterisk box, but how would I interconnect them? |
| 22:32.30 | hardwire | legis: fiber over the toilet. |
| 22:32.42 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:32.45 | hardwire | or possibly some sort of ESP channel. |
| 22:32.45 | lesouvage | StephenF[W]: Do you got the info you were looking for? |
| 22:32.51 | StephenF[W] | hardwire did you see that Google Aprils Fools joke? |
| 22:32.56 | hardwire | yessir |
| 22:33.00 | cvnet | im trying to register a sip phone on my asterisk box, but it doesnt register, and on sip set debug i dont see anything at all, what should i try? |
| 22:33.02 | StephenF[W] | lesouvage yeah I got some ideas, time to test them out |
| 22:33.12 | StephenF[W] | hardwire ok wondering if thats what you were referencing |
| 22:33.24 | StephenF[W] | lesouvage thanks :) |
| 22:33.25 | legis | hardwire: you mean with crosconnect with E1 cards? |
| 22:33.37 | hardwire | cvnet: ouch |
| 22:33.46 | hardwire | legis: that would be silly |
| 22:33.50 | lesouvage | StephenF[W]: there isn't one answer, it all depends on what is happening and what is expected . |
| 22:33.54 | StephenF[W] | right |
| 22:33.56 | hardwire | so they have no internet at their business? |
| 22:34.09 | [TK]D-Fender | cvnet: If you see nothing with SIP debug enabled, then packets aren't even getting to *. Check your routing & firewalls |
| 22:34.24 | legis | hardwire: yes they have but they won't allow it |
| 22:34.36 | legis | hardwire: is just a route inside their network |
| 22:34.36 | hardwire | legis: they won't allow inbound? they won't allow outbound? |
| 22:34.50 | cvnet | [TK]D-Fender firewall is disabled, and i can ping the server, and i can ping outside network from the server |
| 22:34.59 | hardwire | their system, if connected to the internet network and the default gateway, could register to YOU. then you'd use that path |
| 22:35.07 | [TK]D-Fender | cvnet: Its one or the other. |
| 22:35.09 | cvnet | does * has a builtin firewall? |
| 22:35.17 | hardwire | cvnet: linux does, sorta |
| 22:35.28 | cvnet | i turn off the linux firewall |
| 22:35.31 | [TK]D-Fender | cvnet: * no, the OS you're running it ok, yes |
| 22:35.32 | hardwire | cvnet: sudo iptables-save | less -S |
| 22:35.36 | hardwire | anything? |
| 22:35.49 | *** join/#asterisk Firass-VC22 (n=firass@rza.vikcomm.wwu.edu) |
| 22:35.59 | legis | hardwire: yes, the route is inside their WAN with no internet, I was think putting another * box with INternet a crossconnect them |
| 22:36.12 | hardwire | legis: I'm not sure I understand, but does this company want to connect to you with Voice over the Internet Protocol without using the Internet ? |
| 22:36.42 | hardwire | legis: I'd buy their phone system a DSL and call it good. |
| 22:36.49 | legis | hardwire: They don't have to give me Internet, its over IP on their own network |
| 22:37.08 | hardwire | legis: and your job is to? |
| 22:37.47 | hardwire | your in europe eh.. can you use a metro ethernet provider or transparent lan service through a local ISP or MAN provider? |
| 22:38.04 | legis | hardwire: to make calls from the Internet through that trunk, so I was thinkg of putting another * box(this one with INternet) but how to crossconect them? |
| 22:38.07 | hardwire | just put that on the ethernet segment the PBX is on.. |
| 22:38.19 | hardwire | legis: is it a VoIP PBX? |
| 22:38.21 | cvnet | http://pastebin.com/m112f1601 |
| 22:38.52 | cvnet | but i disabled firewall all togehter |
| 22:38.54 | hardwire | legis: if you placed a * box on their site with one network card on the PBX network and one network card with internet access, that would work fine |
| 22:38.56 | hardwire | just use SIP |
| 22:38.57 | legis | hardwire: on the provider side is softswitch |
| 22:39.19 | legis | hardwire: I can't placed nothing on their side |
| 22:39.23 | hardwire | I'm lost.. |
| 22:39.57 | [TK]D-Fender | cvnet: that does not look disabled |
| 22:40.02 | legis | hardwire: yeah is hard for to me explain this |
| 22:40.20 | cvnet | from fedora gui i disabled it |
| 22:40.26 | hardwire | legis: are you joining their network as an extension to their network? |
| 22:40.27 | [TK]D-Fender | cvnet: that as hug REJECTS written on it |
| 22:40.35 | *** join/#asterisk rcy` (n=rcy@S010600003981572c.vc.shawcable.net) |
| 22:40.37 | [TK]D-Fender | cvnet: trash it from CLI |
| 22:40.47 | [TK]D-Fender | cvnet: And back it it |
| 22:40.49 | [TK]D-Fender | up* |
| 22:41.14 | hardwire | legis: so who is they.. and who are you? |
| 22:41.27 | hardwire | what roles do you have with each other? |
| 22:41.33 | legis | hardwire: they gave me a /30 right, that /30 has a route to their SS, that route is only a data channel, no internet |
| 22:42.00 | legis | hardwire: they are a VTP, I'm just buying them a trunk |
| 22:42.25 | legis | hardwire: a 60ch SIP trunk |
| 22:42.26 | hardwire | the /30 goes over what to get to your site? |
| 22:42.33 | legis | hardwire: fiber |
| 22:42.45 | hardwire | so you have a fiber connection to them. |
| 22:42.54 | legis | hardwire: yeah |
| 22:43.03 | hardwire | and they gave you one side of a /30 |
| 22:43.12 | legis | hardwire: correct |
| 22:43.19 | hardwire | and their softswitch is on the other side of the /30 |
| 22:43.34 | legis | hardwire: yeah, |
| 22:43.44 | hardwire | and your asterisk box, on your side, speaks sip |
| 22:43.50 | hardwire | and their softswitch on their side speaks sip |
| 22:43.55 | hardwire | or is that wrong. |
| 22:44.01 | legis | hardwire: yes, I can make calls fine, inbound,outbond |
| 22:44.19 | hardwire | but the biggest problem is? |
| 22:44.28 | legis | hardwire: the problem I would like to make calls from the Internet using that trunk |
| 22:44.37 | hardwire | that's simple |
| 22:44.45 | hardwire | your side needs to have internet. |
| 22:44.55 | hardwire | and route calls via DID to their softswitch |
| 22:45.12 | legis | hardwire: I have Internet |
| 22:45.23 | hardwire | now you need numbers coming into you, right? |
| 22:45.57 | legis | hardwire: but the Internet is in another vlan no the route they gave me |
| 22:46.04 | hardwire | your asterisk box is part of the /30 as well as part of your local network with access to the internet |
| 22:46.09 | hardwire | or direct connection to the internet. |
| 22:46.12 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:46.20 | legis | hardwire: the asterisk box i use to make calls with them has no Internet |
| 22:46.25 | hardwire | ok |
| 22:46.29 | legis | hardwire: direct connection to them |
| 22:46.34 | hardwire | and it's in the same office you have internet access at, correct? |
| 22:46.40 | legis | hardwire: yeah |
| 22:46.50 | hardwire | throw in another network card? |
| 22:46.56 | hardwire | hook it up to the internet |
| 22:47.15 | hardwire | maybe call the fiber provider and have another tagged vlan shuffled in as well for IP service |
| 22:47.52 | hardwire | legis: you don't need a default route on your asterisk box to talk to the other side of the /30 |
| 22:48.09 | hardwire | but you will need one to talk to the internet. |
| 22:48.24 | hardwire | you can somewhat safely put the /30 network on the same network as your internet traffic, without them having issues |
| 22:48.28 | hardwire | as long as their side never sends DHCP |
| 22:48.54 | hardwire | then you would have one ethernet card with an internet or internal IP with a default route to the internet |
| 22:49.01 | hardwire | and an "alias" on top of that with your side of the /30 |
| 22:49.13 | *** join/#asterisk djbecas (i=Asterisk@79.68.108.93.rev.vodafone.pt) |
| 22:49.32 | hardwire | all calls coming in would be proxied from the internet through the asterisk or sip gateway software of your choice, to the other soft switch |
| 22:50.18 | legis | hardwire: I need their the default route to talk to the SS, the default route IP != from the SS IP |
| 22:50.34 | hardwire | no you don't |
| 22:50.45 | djbecas | any one has a digium b410p? Sometimes inbound calls fail can someone help me? |
| 22:50.47 | hardwire | whats their SS IP? |
| 22:50.57 | hardwire | lets call it 10.50.60.1 |
| 22:51.06 | hardwire | and their side of the /30 is 209.112.112.3 |
| 22:51.12 | legis | is on a different subnet from the /30 they gave me |
| 22:51.17 | hardwire | ip route add 10.50.60.1 via 209.112.112.3 |
| 22:51.27 | hardwire | tada? |
| 22:51.32 | hardwire | ip route add 10.50.60.0/24 via 209.112.112.3 |
| 22:51.41 | hardwire | ip route add default via 192.168.100.1 |
| 22:51.51 | hardwire | two gateways, one cup. |
| 22:52.26 | hardwire | remove your default route sir. |
| 22:52.30 | hardwire | then add that direct route |
| 22:52.41 | hardwire | you'll see you don't need a default route to talk to their network |
| 22:53.11 | hardwire | don't make me call the local geek squad and have them ambush you. |
| 22:53.15 | legis | I don't get it |
| 22:53.16 | legis | heh |
| 22:53.27 | hardwire | legis: it's all good |
| 22:53.38 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 22:53.39 | hardwire | I don't know what kind of work you do, but routes on ip networks take mad skillz |
| 22:54.02 | hardwire | you can get said mad skillz by dicking around a lot with random mishmoshed hardware, in the dark, away from the wife. |
| 22:54.16 | legis | lets suppose they gave this /30: IP: 192.168.1.2 netmask: /30 GW: 192.168.1.1 |
| 22:54.26 | hardwire | sure |
| 22:54.31 | [TK]D-Fender | ew |
| 22:54.34 | hardwire | and whats your office networks internal subnet. |
| 22:55.06 | legis | their SS IP is 10.10.10.10 |
| 22:55.10 | hardwire | ok |
| 22:55.20 | murdock_ut | Hey guys, and gals if any, I remember seeing a bug/feature request for making asterisk send caller id information on a call that was picked up from the parking lot. Does anyone know the issue #. I can't seem to find it. |
| 22:55.25 | legis | I need to setup the /30 in the asterisk box to connect to their trunk |
| 22:55.26 | hardwire | ip route add 10.10.10.0/24 via 192.168.1.1 |
| 22:55.42 | hardwire | I think it's via gw 192.168.1.1 |
| 22:55.44 | hardwire | but you get the idea |
| 22:56.28 | legis | you want to put another NIC with the connection |
| 22:56.44 | hardwire | no.. but you can |
| 22:56.46 | hardwire | haha. |
| 22:56.49 | legis | but I need to set GW for the net connect too |
| 22:57.00 | hardwire | legis: what asterisk/what linux distro? |
| 22:57.05 | legis | I mean another NIC with net connection |
| 22:57.13 | legis | hardwire: debian/1.4 |
| 22:57.22 | hardwire | legis: used pastebin before? |
| 22:57.30 | legis | hardwire: sure |
| 22:57.41 | legis | hardwire: what do you need? |
| 22:57.44 | hardwire | are you comfy setting up new network connections in /etc/network/interfaces under debian? |
| 22:57.58 | legis | hardwire: yeah |
| 22:58.13 | hardwire | remove the default gateway for the fiber trunk |
| 22:59.25 | hardwire | now you should just have the /30 configured, correct? |
| 23:00.00 | hardwire | add 'up /sbin/iptables route add 10.10.10.0/24 via 192.168.1.1' |
| 23:00.03 | legis | hardwire: yes, just the IP and the netmask, no GW |
| 23:00.14 | hardwire | then ifdown it |
| 23:00.15 | hardwire | then ifup it |
| 23:00.26 | hardwire | you'll get an error on the ifdown if you didn't already bring it down |
| 23:00.41 | hardwire | once you bring it up, check your route table ala 'ip route show' |
| 23:00.58 | hardwire | I'm guessing you already have two network cards in this box.. |
| 23:01.09 | hardwire | CAUSE I GUESS AWESOME |
| 23:01.19 | *** join/#asterisk mosty (n=mosty@59.167.191.34) |
| 23:01.32 | legis | hardwire: yes |
| 23:01.50 | hardwire | legis: did you restart the network interface? |
| 23:03.36 | legis | hardwire: one min |
| 23:05.25 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 23:08.42 | hardwire | I think the grocery store wants me to stop using their bandwidth |
| 23:09.53 | legis | heh |
| 23:12.00 | hardwire | legis: well.. I probably have to go. but a few other people can help you set up static routes on debian here too.. or in the #debian channel on irc.oftc.net |
| 23:12.14 | legis | hardwire: so is just a routing issue? |
| 23:12.25 | hardwire | actually.. this is a routing normality. |
| 23:12.40 | hardwire | the issue is that you weren't hip to it yet. |
| 23:12.57 | mosty | i have a weird problem with sip callerid on asterisk 1.4.22, the NOTIFY packets have the correct From header, but the INVITE packets put the receiver number in the From header (but with the correct callerid name), so in the call logs it looks like the receiver is called themselves. the call itself works. i have a sip debug if anyone is willing to take a look |
| 23:13.04 | hardwire | they remote side did their jorb just fine |
| 23:13.09 | legis | hardwire: ok, so what do I put on the other nick? |
| 23:13.20 | hardwire | legis: that depends |
| 23:13.28 | hardwire | do you have any public IP addresses to give your asterisk box? |
| 23:13.36 | legis | hardwire: yes, its all public |
| 23:13.40 | hardwire | if so, put it on a public network and give it a public ip/subnet/gateway |
| 23:13.42 | hardwire | and tada |
| 23:13.43 | hardwire | yer done |
| 23:14.09 | hardwire | you should have eth0 with your /30 and a static route and eth1 with your public subnet and a route to your public gateway. |
| 23:14.12 | hardwire | toots. I have to run.. |
| 23:14.29 | legis | hardwire: thanks alot, I will try, |
| 23:14.32 | legis | hardwire: that |
| 23:14.33 | hardwire | word |
| 23:14.33 | jeev | hmm, hardwire, you use 9 to dial out ? |
| 23:14.45 | hardwire | jeev: I always seem to end up using 8 |
| 23:14.54 | hardwire | and 7 for forced dialing through other trunks |
| 23:14.55 | hardwire | 6 |
| 23:14.55 | hardwire | 5 |
| 23:14.55 | hardwire | 4 |
| 23:14.56 | hardwire | etc.. |
| 23:14.57 | hardwire | toots |
| 23:16.38 | hardwire | I'd like to just not use a dial out number |
| 23:16.45 | hardwire | but nobody likes that for some reason |
| 23:16.59 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:20.44 | *** join/#asterisk LiNeTuX_Home (n=LiNeTuX@253.238.95.24.cfl.res.rr.com) |
| 23:21.55 | hesco | the book on page 134 says: "and you shouldn't use single-character extensions for your own use, as they are reserved." My welcome menu is using single-digit extensions for the menu options. I haven't run into issues with this yet. Will I soon? Can anyone explain this line from the book, please? |
| 23:23.08 | *** part/#asterisk djbecas (i=Asterisk@79.68.108.93.rev.vodafone.pt) |
| 23:23.34 | [TK]D-Fender | hesco: ignore that. |
| 23:25.43 | nikko | i think single digits are reserved for future statuses; i,t,s,h, etc. |
| 23:26.01 | nikko | sorry, single characters, not digits |
| 23:28.20 | hesco | thanks folks. Ignore that I can do. |
| 23:28.26 | hesco | pressing on. |
| 23:28.54 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 23:30.47 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 23:31.18 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 23:32.47 | hesco | the programmer in me is reluctant to create a global variable when one of more limited scope might work as well. Is it possible to create a variable without making it global? |
| 23:33.16 | hesco | Is this an issue I should even be concerned about in this context? |
| 23:36.24 | jaytee | hesco, have you read the channelvariables.txt file in the /docs folder of the source? there's alot of good info there and some of it is pertinent to more than just channel variables. |
| 23:42.23 | hesco | jaytee: thanks, will do. |
| 23:42.59 | *** part/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 23:46.57 | *** part/#asterisk Deeewayn1 (n=Deeewayn@nat/digium/x-a81aa89a5aad4bcd) |
| 23:50.38 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
